Alsa-devel April 2018

alsa-devel@alsa-project.org

144 participants
268 discussions

[alsa-devel] [PATCH] soundwire: Update email address for Vinod
by Vinod Koul 06 May '18

06 May '18

Update the email address for SoundWire maintainer Signed-off-by: Vinod Koul <vkoul(a)kernel.org> --- MAINTAINERS | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 715d054531af..5caaad0d465d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -12687,7 +12687,7 @@ F: include/uapi/sound/ F: sound/ SOUND - COMPRESSED AUDIO -M: Vinod Koul <vinod.koul(a)intel.com> +M: Vinod Koul <vkoul(a)kernel.org> L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Supported -- 2.7.4

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[alsa-devel] [PATCH v3] media: i2c: tda1997: replace codec to component
by Kuninori Morimoto 05 May '18

05 May '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Tim, Mauro This replace patch is needed for ALSA SoC, otherwise it can't probe anymore. Mark, I think it needs your acked-by ? v2 -> v3 - fixup driver name (= tda1997) drivers/media/i2c/tda1997x.c | 25 +++++++++++++------------ 1 file changed, 13 insertions(+), 12 deletions(-) diff --git a/drivers/media/i2c/tda1997x.c b/drivers/media/i2c/tda1997x.c index 3021913..33d7fcf 100644 --- a/drivers/media/i2c/tda1997x.c +++ b/drivers/media/i2c/tda1997x.c @@ -2444,7 +2444,7 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct tda1997x_state *state = snd_soc_dai_get_drvdata(dai); - struct snd_soc_codec *codec = dai->codec; + struct snd_soc_component *component = dai->component; struct snd_pcm_runtime *rtd = substream->runtime; int rate, err; @@ -2452,11 +2452,11 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, err = snd_pcm_hw_constraint_minmax(rtd, SNDRV_PCM_HW_PARAM_RATE, rate, rate); if (err < 0) { - dev_err(codec->dev, "failed to constrain samplerate to %dHz\n", + dev_err(component->dev, "failed to constrain samplerate to %dHz\n", rate); return err; } - dev_info(codec->dev, "set samplerate constraint to %dHz\n", rate); + dev_info(component->dev, "set samplerate constraint to %dHz\n", rate); return 0; } @@ -2479,20 +2479,22 @@ static int tda1997x_pcm_startup(struct snd_pcm_substream *substream, .ops = &tda1997x_dai_ops, }; -static int tda1997x_codec_probe(struct snd_soc_codec *codec) +static int tda1997x_codec_probe(struct snd_soc_component *component) { return 0; } -static int tda1997x_codec_remove(struct snd_soc_codec *codec) +static void tda1997x_codec_remove(struct snd_soc_component *component) { - return 0; } -static struct snd_soc_codec_driver tda1997x_codec_driver = { - .probe = tda1997x_codec_probe, - .remove = tda1997x_codec_remove, - .reg_word_size = sizeof(u16), +static struct snd_soc_component_driver tda1997x_codec_driver = { + .probe = tda1997x_codec_probe, + .remove = tda1997x_codec_remove, + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static int tda1997x_probe(struct i2c_client *client, @@ -2737,7 +2739,7 @@ static int tda1997x_probe(struct i2c_client *client, else formats = SNDRV_PCM_FMTBIT_S16_LE; tda1997x_audio_dai.capture.formats = formats; - ret = snd_soc_register_codec(&state->client->dev, + ret = devm_snd_soc_register_component(&state->client->dev, &tda1997x_codec_driver, &tda1997x_audio_dai, 1); if (ret) { @@ -2782,7 +2784,6 @@ static int tda1997x_remove(struct i2c_client *client) struct tda1997x_platform_data *pdata = &state->pdata; if (pdata->audout_format) { - snd_soc_unregister_codec(&client->dev); mutex_destroy(&state->audio_lock); } -- 1.9.1

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[alsa-devel] [PATCH v3 0/7] ASoC: omap: Deprecate omap-pcm for sdma-pcm
by Peter Ujfalusi 05 May '18

05 May '18

Hi, Changes since v2: - Return -ENODEV from sdma_pcm_platform_register() in case !CONFIG_SND_SDMA_SOC Changes since v1: - Rebased on the n810 patches (Kconfig conflict) - Delete the omap-pcm as the last step instead of keeping the .c/.h file. If there is a need it can be resurrected with a simple revert. - I left out the Acked-by from Jarkko for the last patch since I have changed it omap-pcm served us well over the years, but there is no reason to keep it around as we can rely on generic dmaengine_pcm code without the need to maintain custom code. The series has been tested on: omap2 n810, omap3 beagle-xm, omap4 PandaES, omap4 Blaze (SDP), am57xx beagle-x15, dra7 evm. McBSP, McASP, McPDM, DMIC and HDMI audio was tested and I have found no issue with them. I was not able to test on OMAP1 for various reasons, but mainly because the Nokia 770 audio is not supported in mainline... Regards, Peter --- Peter Ujfalusi (7): ASoC: omap: Introduce the generic_dmaengine_pcm based sdma-pcm ASoC: davinci-mcasp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-hdmi-audio: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-dmic: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcpdm: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap-mcbsp: Convert to use the sdma-pcm instead of omap-pcm ASoC: omap: Delete the obsolete omap-pcm include/sound/omap-pcm.h | 30 ---- sound/soc/davinci/Kconfig | 2 +- sound/soc/davinci/davinci-mcasp.c | 8 +- sound/soc/omap/Kconfig | 33 ++-- sound/soc/omap/Makefile | 4 +- sound/soc/omap/omap-dmic.c | 4 +- sound/soc/omap/omap-hdmi-audio.c | 5 +- sound/soc/omap/omap-mcbsp.c | 4 +- sound/soc/omap/omap-mcpdm.c | 4 +- sound/soc/omap/omap-pcm.c | 262 ------------------------------ sound/soc/omap/sdma-pcm.c | 68 ++++++++ sound/soc/omap/sdma-pcm.h | 21 +++ 12 files changed, 124 insertions(+), 321 deletions(-) delete mode 100644 include/sound/omap-pcm.h delete mode 100644 sound/soc/omap/omap-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.c create mode 100644 sound/soc/omap/sdma-pcm.h -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH v2] ASoC: da7219: read fmw property to get mclk for non-dts systems
by Akshu Agrawal 03 May '18

03 May '18

Non-dts based systems can use ACPI DSDT to pass on the mclk to da7219. This enables da7219 mclk to be linked to system clock. Enable/Disable of the mclk is already handled in the codec so platform drivers don't have to explicitly do handling of mclk. Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- v2: Fixed kbuild error include/sound/da7219.h | 2 ++ sound/soc/codecs/da7219.c | 7 ++++++- 2 files changed, 8 insertions(+), 1 deletion(-) diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 1bfcb16..df7ddf4 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -38,6 +38,8 @@ struct da7219_pdata { const char *dai_clks_name; + const char *mclk_name; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 980a6a8..aed68a4 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1624,6 +1624,8 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone dev_warn(dev, "Using default clk name: %s\n", pdata->dai_clks_name); + device_property_read_string(dev, "dlg,mclk-name", &pdata->mclk_name); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -1905,7 +1907,10 @@ static int da7219_probe(struct snd_soc_component *component) da7219_handle_pdata(component); /* Check if MCLK provided */ - da7219->mclk = devm_clk_get(component->dev, "mclk"); + if (da7219->pdata->mclk_name) + da7219->mclk = clk_get(NULL, da7219->pdata->mclk_name); + if (!da7219->mclk) + da7219->mclk = devm_clk_get(component->dev, "mclk"); if (IS_ERR(da7219->mclk)) { if (PTR_ERR(da7219->mclk) != -ENOENT) { ret = PTR_ERR(da7219->mclk); -- 1.9.1

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[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

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[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

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[alsa-devel] [PATCH v2 0/5] ALSA: xen-front: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 02 May '18

02 May '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Please note: this patch series depends on [3]. This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Corresponding backend, implemented as a user-space application, can be found at [2]. Thank you, Oleksandr Changes since v1: ***************** 1. Moved driver from sound/drivers to sound/xen 2. Coding style changes to better meet Linux Kernel 3. Added explicit back and front synchronization In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events 4. Added explicit back and front parameter negotiation In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameters given: request passes desired parameter's intervals/masks and the response to this request returns allowed min/max intervals/masks to be used. [1] https://xenproject.org/ [2] https://github.com/xen-troops/snd_be [3] https://lkml.org/lkml/2018/4/12/522 Oleksandr Andrushchenko (5): ALSA: xen-front: Introduce Xen para-virtualized sound frontend driver ALSA: xen-front: Read sound driver configuration from Xen store ALSA: xen-front: Implement Xen event channel handling ALSA: xen-front: Implement handling of shared buffers ALSA: xen-front: Implement ALSA virtual sound driver sound/Kconfig | 2 + sound/Makefile | 2 +- sound/xen/Kconfig | 10 + sound/xen/Makefile | 9 + sound/xen/xen_snd_front.c | 410 +++++++++++++++++++ sound/xen/xen_snd_front.h | 57 +++ sound/xen/xen_snd_front_alsa.c | 830 ++++++++++++++++++++++++++++++++++++++ sound/xen/xen_snd_front_alsa.h | 23 ++ sound/xen/xen_snd_front_cfg.c | 517 ++++++++++++++++++++++++ sound/xen/xen_snd_front_cfg.h | 46 +++ sound/xen/xen_snd_front_evtchnl.c | 478 ++++++++++++++++++++++ sound/xen/xen_snd_front_evtchnl.h | 92 +++++ sound/xen/xen_snd_front_shbuf.c | 193 +++++++++ sound/xen/xen_snd_front_shbuf.h | 36 ++ 14 files changed, 2704 insertions(+), 1 deletion(-) create mode 100644 sound/xen/Kconfig create mode 100644 sound/xen/Makefile create mode 100644 sound/xen/xen_snd_front.c create mode 100644 sound/xen/xen_snd_front.h create mode 100644 sound/xen/xen_snd_front_alsa.c create mode 100644 sound/xen/xen_snd_front_alsa.h create mode 100644 sound/xen/xen_snd_front_cfg.c create mode 100644 sound/xen/xen_snd_front_cfg.h create mode 100644 sound/xen/xen_snd_front_evtchnl.c create mode 100644 sound/xen/xen_snd_front_evtchnl.h create mode 100644 sound/xen/xen_snd_front_shbuf.c create mode 100644 sound/xen/xen_snd_front_shbuf.h -- 2.16.2

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[alsa-devel] [PATCH v4 1/3] dell-led: Change dell-led.h to dell-common.h
by Kai-Heng Feng 02 May '18

02 May '18

This header will be used for more than just led. Change it to a more generic name. Cc: Mario Limonciello <mario.limonciello(a)dell.com> Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- v4: Change the commit message to clarify there's no more runtime pm warning. Also skip the check for thunderbolt attached devices. v3: Simplify dell_switchable_gfx_is_enabled() by returning bool instead of error code. Use DMI_DEV_TYPE_OEM_STRING to match Dell System. v2: Mario suggested to squash the HDA part into the same series. drivers/platform/x86/dell-laptop.c | 2 +- include/linux/{dell-led.h => dell-common.h} | 4 ++-- sound/pci/hda/dell_wmi_helper.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) rename include/linux/{dell-led.h => dell-common.h} (61%) diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index c52c6723374b..8ba820e6c3d0 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -29,7 +29,7 @@ #include <linux/mm.h> #include <linux/i8042.h> #include <linux/debugfs.h> -#include <linux/dell-led.h> +#include <linux/dell-common.h> #include <linux/seq_file.h> #include <acpi/video.h> #include "dell-rbtn.h" diff --git a/include/linux/dell-led.h b/include/linux/dell-common.h similarity index 61% rename from include/linux/dell-led.h rename to include/linux/dell-common.h index 92521471517f..37e4b614dd74 100644 --- a/include/linux/dell-led.h +++ b/include/linux/dell-common.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ -#ifndef __DELL_LED_H__ -#define __DELL_LED_H__ +#ifndef __DELL_COMMON_H__ +#define __DELL_COMMON_H__ int dell_micmute_led_set(int on); diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..56050cc3c0ee 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -4,7 +4,7 @@ */ #if IS_ENABLED(CONFIG_DELL_LAPTOP) -#include <linux/dell-led.h> +#include <linux/dell-common.h> enum { MICMUTE_LED_ON, -- 2.17.0

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[alsa-devel] [PATCH 0/5] conf/ucm: chtrt5645: Various updates
by Hans de Goede 02 May '18

02 May '18

Hi All, Here is a series of chtrt5645 UCM updates, the first 2 patches bring in some fixes for the generic config from: https://github.com/plbossart/UCM The 3th patch adds some analog mic input fixes, together these 3 patches bring the generic chtrt5645 config in sync with the newer chtrt5645-mono-speaker-analog-mic config, but then for stereo speakers. The 4th and the 5th patch add long-name using board/model specific configs for the Asus T100HA and Lenovo Miix 320, these use a digital mic connected to the DMIC1 resp. DMIC2. input. The generic chtrt5645 config has been tested on a Lenovo Miix 310 and a Kazam Vision tablet, both of which feature stereo speakers and an analog microphone. The Asus T100HA and Lenovo Miix 320 configs have been tested on the resp. models they apply to. Regards, Hans

2 6

[alsa-devel] [PATCH] ASoC: mediatek: avoid using snd_soc_platform
by KaiChieh Chuang 01 May '18

01 May '18

avoid using snd_soc_platform, which is removed after 4.18 Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index d046ea8..8966ee1 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -62,7 +62,7 @@ static int mtk_afe_pcm_new(struct snd_soc_pcm_runtime *rtd) size = afe->mtk_afe_hardware->buffer_bytes_max; return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - rtd->platform->dev, + afe->dev, size, size); } -- 1.7.9.5

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Alsa-devel April 2018