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

[alsa-devel] [PATCH v3 00/13] Enable HDA Codec support on Intel Platforms
by Pierre-Louis Bossart 04 Jul '18

04 Jul '18

Many Intel platforms (SKL, KBL) etc. in the market supports enhanced audio capabilities which also includes DSP processing. The default HDaudio legacy driver does not allow for the use of the DSP, this patch set makes it possible while reusing existing code for HDAudio codecs and without significant changes to the legacy driver. This v3 is not split into two batches as done for v1 and v2, but keeps the same logical progression. The first three patches are mostly data structure changes, the DSP support capability is added then with an ASoC HDA driver and the last patches are fixes required for Skylake+. The changes to the HDAudio legacy driver are minimal. Tests were run successfully on multiple platforms (Dell XPS13, KBL NUC, APL NUC and LeafHill reference board). Credits: all the initial code was written by Rakesh Ughreja, the rebase to broonie/for-next, cleanups and additional tests were done by Pierre Bossart. Changes v3: - port to component model - additional tests on ApolloLake and KabyLake NUC devices - cleanups (alignment, typos, etc) Changes v2: - Resolved review comments and rebased to latest kernel. - added module load support for codec drivers. Rakesh Ughreja (13): ALSA: hdac: Remove usage of struct hdac_ext_device and use hdac_device instead ALSA: hdac: Remove usage of struct hdac_ext_bus and use hdac_bus instead ALSA: hdac: Remove usage of struct hdac_ext_driver, use hdac_driver instead ASoC: Intel: Boards: Machine driver for SKL+ w/ HDAudio codecs ASoC: Intel: Skylake: Add entry in sst_acpi_mach for HDA codecs ASoC: Intel: Skylake: add HDA BE DAIs ASoC: Intel: Skylake: use hda_bus instead of hdac_bus ALSA: hda: split snd_hda_codec_new function ALSA: hdac: remove memory allocation from snd_hdac_ext_bus_device_init ALSA: hdac: add extended ops in the hdac_bus ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers ALSA: hdac: ext: add wait for codec to respond after link reset ASoC: Intel: Skylake: fix widget handling include/sound/hdaudio.h | 29 ++ include/sound/hdaudio_ext.h | 123 ++----- sound/hda/ext/hdac_ext_bus.c | 80 ++--- sound/hda/ext/hdac_ext_controller.c | 64 ++-- sound/hda/ext/hdac_ext_stream.c | 104 +++--- sound/pci/hda/hda_bind.c | 12 + sound/pci/hda/hda_codec.c | 68 +++- sound/pci/hda/hda_codec.h | 2 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/hdac_hda.c | 469 +++++++++++++++++++++++++++ sound/soc/codecs/hdac_hda.h | 24 ++ sound/soc/codecs/hdac_hdmi.c | 416 +++++++++++------------- sound/soc/intel/boards/Kconfig | 9 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_hda_dsp_common.c | 133 ++++++++ sound/soc/intel/boards/skl_hda_dsp_common.h | 37 +++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 174 ++++++++++ sound/soc/intel/skylake/skl-messages.c | 50 ++- sound/soc/intel/skylake/skl-nhlt.c | 8 +- sound/soc/intel/skylake/skl-pcm.c | 182 +++++++---- sound/soc/intel/skylake/skl-topology.c | 23 +- sound/soc/intel/skylake/skl-topology.h | 6 +- sound/soc/intel/skylake/skl.c | 296 +++++++++++------ sound/soc/intel/skylake/skl.h | 13 +- 25 files changed, 1640 insertions(+), 691 deletions(-) create mode 100644 sound/soc/codecs/hdac_hda.c create mode 100644 sound/soc/codecs/hdac_hda.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.c create mode 100644 sound/soc/intel/boards/skl_hda_dsp_common.h create mode 100644 sound/soc/intel/boards/skl_hda_dsp_generic.c -- 2.14.1

3 22

[alsa-devel] [PATCH v4 0/7] soundwire: Add multi link support
by Shreyas NC 04 Jul '18

04 Jul '18

Currently, in the SoundWire subsystem, the concept of stream is limited to a Master and one or more Slaves(Codecs). This series aims to add support for multiple Master(s) sharing the same reference clock and synchronized in the hardware. This patch series adds: - Helpers to lock bus instances part of the stream - Boiler plate conversion of code to support a list of Master runtime - Support multi link bank switch to support synchronization between multiple masters - Add Intel platform ops for pre/post bank switch changes in v4: - Added changes in sdw_stream_runtime structure to track Masters added to stream - Add changes to support ml_bankswitch only if both the hardware supports and the stream is handled by multiple masters changes in v3: - Added comments to explain the trigger for multi link bank switch - Added comments in the intel_post_bank_switch ops to explain how synchronous trigger of bank switch is handled. Sanyog Kale (2): Documentation: soundwire: Add documentation for multi link soundwire: Add support to lock across bus instances Shreyas NC (4): soundwire: Initialize completion for defer messages soundwire: keep track of Masters in a stream soundwire: Add support for multi link bank switch soundwire: intel: Add pre/post bank switch ops Vinod Koul (1): soundwire: Handle multiple master instances in a stream Documentation/driver-api/soundwire/stream.rst | 28 ++ drivers/soundwire/bus.c | 6 + drivers/soundwire/bus.h | 4 + drivers/soundwire/intel.c | 65 ++++ drivers/soundwire/stream.c | 483 +++++++++++++++++++------- include/linux/soundwire/sdw.h | 12 +- 6 files changed, 467 insertions(+), 131 deletions(-) -- 2.7.4

5 23

[alsa-devel] [PATCH v3] ASoC: Intel: Boards: Add GLK Realtek Maxim I2S machine driver
by Naveen Manohar 03 Jul '18

03 Jul '18

Patch adds Geminilake I2S machine driver which uses following codecs: RT5682 and MAX98357A. Signed-off-by: Naveen Manohar <naveen.m(a)intel.com> Signed-off-by: Harsha Priya <harshapriya.n(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v1->v2: reverse xmas tree alignment of var declaration, Add Acked-by. v2->v3: corrected: platform-name for echo-ref dai & change-log position. sound/soc/intel/boards/Kconfig | 14 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/glk_rt5682_max98357a.c | 644 ++++++++++++++++++++++++++ 3 files changed, 660 insertions(+) create mode 100644 sound/soc/intel/boards/glk_rt5682_max98357a.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 2479748..cccda87 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,20 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH + tristate "GLK with RT5682 and MAX98357A in I2S Mode" + depends on MFD_INTEL_LPSS && I2C && ACPI + select SND_SOC_RT5682 + select SND_SOC_MAX98357A + select SND_SOC_DMIC + select SND_SOC_HDAC_HDMI + select SND_HDA_DSP_LOADER + help + This adds support for ASoC machine driver for Geminilake platforms + with RT5682 + MAX98357A I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507..87ef8b4 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bdw-rt5677-mach-objs := bdw-rt5677.o snd-soc-sst-broadwell-objs := broadwell.o snd-soc-sst-bxt-da7219_max98357a-objs := bxt_da7219_max98357a.o snd-soc-sst-bxt-rt298-objs := bxt_rt298.o +snd-soc-sst-glk-rt5682_max98357a-objs := glk_rt5682_max98357a.o snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-bytcr-rt5651-objs := bytcr_rt5651.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o @@ -27,6 +28,7 @@ obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH) += snd-soc-sst-bxt-da7219_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BXT_RT298_MACH) += snd-soc-sst-bxt-rt298.o +obj-$(CONFIG_SND_SOC_INTEL_GLK_RT5682_MAX98357A_MACH) += snd-soc-sst-glk-rt5682_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o obj-$(CONFIG_SND_SOC_INTEL_BDW_RT5677_MACH) += snd-soc-sst-bdw-rt5677-mach.o obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c new file mode 100644 index 0000000..2c36ff4 --- /dev/null +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -0,0 +1,644 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2018 Intel Corporation. + +/* + * Intel Geminilake I2S Machine Driver with MAX98357A & RT5682 Codecs + * + * Modified from: + * Intel Apollolake I2S Machine driver + */ + +#include <linux/input.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/jack.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../skylake/skl.h" +#include "../../codecs/rt5682.h" +#include "../../codecs/hdac_hdmi.h" + +/* The platform clock outputs 19.2Mhz clock to codec as I2S MCLK */ +#define GLK_PLAT_CLK_FREQ 19200000 +#define RT5682_PLL_FREQ (48000 * 512) +#define GLK_REALTEK_CODEC_DAI "rt5682-aif1" +#define GLK_MAXIM_CODEC_DAI "HiFi" +#define MAXIM_DEV0_NAME "MX98357A:00" +#define DUAL_CHANNEL 2 +#define QUAD_CHANNEL 4 +#define NAME_SIZE 32 + +static struct snd_soc_jack geminilake_hdmi[3]; + +struct glk_hdmi_pcm { + struct list_head head; + struct snd_soc_dai *codec_dai; + int device; +}; + +struct glk_card_private { + struct snd_soc_jack geminilake_headset; + struct list_head hdmi_pcm_list; +}; + +enum { + GLK_DPCM_AUDIO_PB = 0, + GLK_DPCM_AUDIO_CP, + GLK_DPCM_AUDIO_HS_PB, + GLK_DPCM_AUDIO_ECHO_REF_CP, + GLK_DPCM_AUDIO_REF_CP, + GLK_DPCM_AUDIO_DMIC_CP, + GLK_DPCM_AUDIO_HDMI1_PB, + GLK_DPCM_AUDIO_HDMI2_PB, + GLK_DPCM_AUDIO_HDMI3_PB, +}; + +static int platform_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai; + int ret = 0; + + codec_dai = snd_soc_card_get_codec_dai(card, GLK_REALTEK_CODEC_DAI); + if (!codec_dai) { + dev_err(card->dev, "Codec dai not found; Unable to set/unset codec pll\n"); + return -EIO; + } + + if (SND_SOC_DAPM_EVENT_OFF(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, + RT5682_PLL1_S_MCLK, 0, 0); + if (ret) + dev_err(card->dev, "failed to stop PLL: %d\n", ret); + } else if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5682_PLL1_S_MCLK, + GLK_PLAT_CLK_FREQ, RT5682_PLL_FREQ); + if (ret < 0) { + dev_err(card->dev, "can't set codec pll: %d\n", ret); + return ret; + } + } + + if (ret) + dev_err(card->dev, "failed to start internal clk: %d\n", ret); + + return ret; +} + +static const struct snd_kcontrol_new geminilake_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Spk"), +}; + +static const struct snd_soc_dapm_widget geminilake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_SPK("Spk", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SPK("HDMI1", NULL), + SND_SOC_DAPM_SPK("HDMI2", NULL), + SND_SOC_DAPM_SPK("HDMI3", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + platform_clock_control, SND_SOC_DAPM_PRE_PMU | + SND_SOC_DAPM_POST_PMD), +}; + +static const struct snd_soc_dapm_route geminilake_map[] = { + /* HP jack connectors - unknown if we have jack detection */ + { "Headphone Jack", NULL, "Platform Clock" }, + { "Headphone Jack", NULL, "HPOL" }, + { "Headphone Jack", NULL, "HPOR" }, + + /* speaker */ + { "Spk", NULL, "Speaker" }, + + /* other jacks */ + { "Headset Mic", NULL, "Platform Clock" }, + { "IN1P", NULL, "Headset Mic" }, + + /* digital mics */ + { "DMic", NULL, "SoC DMIC" }, + + /* CODEC BE connections */ + { "HiFi Playback", NULL, "ssp1 Tx" }, + { "ssp1 Tx", NULL, "codec0_out" }, + + { "AIF1 Playback", NULL, "ssp2 Tx" }, + { "ssp2 Tx", NULL, "codec1_out" }, + + { "codec0_in", NULL, "ssp2 Rx" }, + { "ssp2 Rx", NULL, "AIF1 Capture" }, + + { "HDMI1", NULL, "hif5-0 Output" }, + { "HDMI2", NULL, "hif6-0 Output" }, + { "HDMI2", NULL, "hif7-0 Output" }, + + { "hifi3", NULL, "iDisp3 Tx" }, + { "iDisp3 Tx", NULL, "iDisp3_out" }, + { "hifi2", NULL, "iDisp2 Tx" }, + { "iDisp2 Tx", NULL, "iDisp2_out" }, + { "hifi1", NULL, "iDisp1 Tx" }, + { "iDisp1 Tx", NULL, "iDisp1_out" }, + + /* DMIC */ + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "DMIC AIF" }, +}; + +static int geminilake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); + + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set(fmt, SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static int geminilake_rt5682_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_component *component = rtd->codec_dai->component; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct snd_soc_jack *jack; + int ret; + + /* Configure sysclk for codec */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT5682_SCLK_S_PLL1, + RT5682_PLL_FREQ, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "snd_soc_dai_set_sysclk err = %d\n", ret); + + /* + * Headset buttons map to the google Reference headset. + * These can be configured by userspace. + */ + ret = snd_soc_card_jack_new(rtd->card, "Headset Jack", + SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3 | SND_JACK_LINEOUT, + &ctx->geminilake_headset, NULL, 0); + if (ret) { + dev_err(rtd->dev, "Headset Jack creation failed: %d\n", ret); + return ret; + } + + jack = &ctx->geminilake_headset; + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_PLAYPAUSE); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEDOWN); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOICECOMMAND); + ret = snd_soc_component_set_jack(component, jack, NULL); + + if (ret) { + dev_err(rtd->dev, "Headset Jack call-back failed: %d\n", ret); + return ret; + } + + return ret; +}; + +static int geminilake_rt5682_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; + int ret; + + /* Set valid bitmask & configuration for I2S in 24 bit */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x0, 0x0, 2, 24); + if (ret < 0) { + dev_err(rtd->dev, "set TDM slot err:%d\n", ret); + return ret; + } + + return ret; +} + +static struct snd_soc_ops geminilake_rt5682_ops = { + .hw_params = geminilake_rt5682_hw_params, +}; + +static int geminilake_hdmi_init(struct snd_soc_pcm_runtime *rtd) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(rtd->card); + struct snd_soc_dai *dai = rtd->codec_dai; + struct glk_hdmi_pcm *pcm; + + pcm = devm_kzalloc(rtd->card->dev, sizeof(*pcm), GFP_KERNEL); + if (!pcm) + return -ENOMEM; + + pcm->device = GLK_DPCM_AUDIO_HDMI1_PB + dai->id; + pcm->codec_dai = dai; + + list_add_tail(&pcm->head, &ctx->hdmi_pcm_list); + + return 0; +} + +static int geminilake_rt5682_fe_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_component *component = rtd->cpu_dai->component; + struct snd_soc_dapm_context *dapm; + int ret; + + dapm = snd_soc_component_get_dapm(component); + ret = snd_soc_dapm_ignore_suspend(dapm, "Reference Capture"); + if (ret) { + dev_err(rtd->dev, "Ref Cap ignore suspend failed %d\n", ret); + return ret; + } + + return ret; +} + +static const unsigned int rates[] = { + 48000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_rates = { + .count = ARRAY_SIZE(rates), + .list = rates, + .mask = 0, +}; + +static const unsigned int channels[] = { + DUAL_CHANNEL, +}; + +static const struct snd_pcm_hw_constraint_list constraints_channels = { + .count = ARRAY_SIZE(channels), + .list = channels, + .mask = 0, +}; + +static unsigned int channels_quad[] = { + QUAD_CHANNEL, +}; + +static struct snd_pcm_hw_constraint_list constraints_channels_quad = { + .count = ARRAY_SIZE(channels_quad), + .list = channels_quad, + .mask = 0, +}; + +static int geminilake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = hw_param_interval(params, + SNDRV_PCM_HW_PARAM_CHANNELS); + + /* + * set BE channel constraint as user FE channels + */ + channels->min = channels->max = 4; + + return 0; +} + +static int geminilake_dmic_startup(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + + runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; + snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &constraints_channels_quad); + + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); +} + +static const struct snd_soc_ops geminilake_dmic_ops = { + .startup = geminilake_dmic_startup, +}; + +static const unsigned int rates_16000[] = { + 16000, +}; + +static const struct snd_pcm_hw_constraint_list constraints_16000 = { + .count = ARRAY_SIZE(rates_16000), + .list = rates_16000, +}; + +static int geminilake_refcap_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_16000); +}; + +static const struct snd_soc_ops geminilake_refcap_ops = { + .startup = geminilake_refcap_startup, +}; + +/* geminilake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link geminilake_dais[] = { + /* Front End DAI links */ + [GLK_DPCM_AUDIO_PB] = { + .name = "Glk Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .init = geminilake_rt5682_fe_init, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + }, + [GLK_DPCM_AUDIO_CP] = { + .name = "Glk Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:0e.0", + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .nonatomic = 1, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_capture = 1, + }, + [GLK_DPCM_AUDIO_HS_PB] = { + .name = "Glk Audio Headset Playback", + .stream_name = "Headset Audio", + .cpu_dai_name = "System Pin2", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .dpcm_playback = 1, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_ECHO_REF_CP] = { + .name = "Glk Audio Echo Reference cap", + .stream_name = "Echoreference Capture", + .cpu_dai_name = "Echoref Pin", + .platform_name = "0000:00:0e.0", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .init = NULL, + .capture_only = 1, + .nonatomic = 1, + }, + [GLK_DPCM_AUDIO_REF_CP] = { + .name = "Glk Audio Reference cap", + .stream_name = "Refcap", + .cpu_dai_name = "Reference Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_refcap_ops, + }, + [GLK_DPCM_AUDIO_DMIC_CP] = { + .name = "Glk Audio DMIC cap", + .stream_name = "dmiccap", + .cpu_dai_name = "DMIC Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .init = NULL, + .dpcm_capture = 1, + .nonatomic = 1, + .dynamic = 1, + .ops = &geminilake_dmic_ops, + }, + [GLK_DPCM_AUDIO_HDMI1_PB] = { + .name = "Glk HDMI Port1", + .stream_name = "Hdmi1", + .cpu_dai_name = "HDMI1 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI2_PB] = { + .name = "Glk HDMI Port2", + .stream_name = "Hdmi2", + .cpu_dai_name = "HDMI2 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .dpcm_playback = 1, + .init = NULL, + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .nonatomic = 1, + .dynamic = 1, + }, + [GLK_DPCM_AUDIO_HDMI3_PB] = { + .name = "Glk HDMI Port3", + .stream_name = "Hdmi3", + .cpu_dai_name = "HDMI3 Pin", + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .platform_name = "0000:00:0e.0", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, SND_SOC_DPCM_TRIGGER_POST}, + .dpcm_playback = 1, + .init = NULL, + .nonatomic = 1, + .dynamic = 1, + }, + /* Back End DAI links */ + { + /* SSP1 - Codec */ + .name = "SSP1-Codec", + .id = 0, + .cpu_dai_name = "SSP1 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = MAXIM_DEV0_NAME, + .codec_dai_name = GLK_MAXIM_CODEC_DAI, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .dpcm_playback = 1, + }, + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .id = 1, + .cpu_dai_name = "SSP2 Pin", + .platform_name = "0000:00:0e.0", + .no_pcm = 1, + .codec_name = "i2c-10EC5682:00", + .codec_dai_name = GLK_REALTEK_CODEC_DAI, + .init = geminilake_rt5682_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = geminilake_ssp_fixup, + .ops = &geminilake_rt5682_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .id = 2, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:0e.0", + .ignore_suspend = 1, + .be_hw_params_fixup = geminilake_dmic_fixup, + .dpcm_capture = 1, + .no_pcm = 1, + }, + { + .name = "iDisp1", + .id = 3, + .cpu_dai_name = "iDisp1 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi1", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp2", + .id = 4, + .cpu_dai_name = "iDisp2 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi2", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, + { + .name = "iDisp3", + .id = 5, + .cpu_dai_name = "iDisp3 Pin", + .codec_name = "ehdaudio0D2", + .codec_dai_name = "intel-hdmi-hifi3", + .platform_name = "0000:00:0e.0", + .init = geminilake_hdmi_init, + .dpcm_playback = 1, + .no_pcm = 1, + }, +}; + +static int glk_card_late_probe(struct snd_soc_card *card) +{ + struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); + struct snd_soc_component *component = NULL; + char jack_name[NAME_SIZE]; + struct glk_hdmi_pcm *pcm; + int err = 0; + int i = 0; + + list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { + component = pcm->codec_dai->component; + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pcm=%d Jack", pcm->device); + err = snd_soc_card_jack_new(card, jack_name, + SND_JACK_AVOUT, &geminilake_hdmi[i], + NULL, 0); + + if (err) + return err; + + err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, + &geminilake_hdmi[i]); + if (err < 0) + return err; + + i++; + } + + if (!component) + return -EINVAL; + + return hdac_hdmi_jack_port_init(component, &card->dapm); +} + +/* geminilake audio machine driver for SPT + RT5682 */ +static struct snd_soc_card glk_audio_card_rt5682_m98357a = { + .name = "glkrt5682max", + .owner = THIS_MODULE, + .dai_link = geminilake_dais, + .num_links = ARRAY_SIZE(geminilake_dais), + .controls = geminilake_controls, + .num_controls = ARRAY_SIZE(geminilake_controls), + .dapm_widgets = geminilake_widgets, + .num_dapm_widgets = ARRAY_SIZE(geminilake_widgets), + .dapm_routes = geminilake_map, + .num_dapm_routes = ARRAY_SIZE(geminilake_map), + .fully_routed = true, + .late_probe = glk_card_late_probe, +}; + +static int geminilake_audio_probe(struct platform_device *pdev) +{ + struct glk_card_private *ctx; + + ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_ATOMIC); + if (!ctx) + return -ENOMEM; + + INIT_LIST_HEAD(&ctx->hdmi_pcm_list); + + glk_audio_card_rt5682_m98357a.dev = &pdev->dev; + snd_soc_card_set_drvdata(&glk_audio_card_rt5682_m98357a, ctx); + + return devm_snd_soc_register_card(&pdev->dev, + &glk_audio_card_rt5682_m98357a); +} + +static const struct platform_device_id glk_board_ids[] = { + { + .name = "glk_rt5682_max98357a", + .driver_data = + (kernel_ulong_t)&glk_audio_card_rt5682_m98357a, + }, + { } +}; + +static struct platform_driver geminilake_audio = { + .probe = geminilake_audio_probe, + .driver = { + .name = "glk_rt5682_max98357a", + .pm = &snd_soc_pm_ops, + }, + .id_table = glk_board_ids, +}; +module_platform_driver(geminilake_audio) + +/* Module information */ +MODULE_DESCRIPTION("Geminilake Audio Machine driver-RT5682 & MAX98357A in I2S mode"); +MODULE_AUTHOR("Naveen Manohar <naveen.m(a)intel.com>"); +MODULE_AUTHOR("Harsha Priya <harshapriya.n(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:glk_rt5682_max98357a"); -- 2.7.4

2 1

[alsa-devel] [PATCH v3 0/2] ASoC: add es7241 support
by Jerome Brunet 03 Jul '18

03 Jul '18

Add support for the es7241 which is a simple 2 channels analog to digital converter. Changes since v1: [0] * Add missing break as reported by smatch [1] My device is hardwired in slave mode, which probably why this slipped through unnoticed. Changes since v2: [2] * Supply are now required dt properties * Add a dt property to state the serial interface configuration [0]: https://lkml.kernel.org/r/20180629152444.21525-1-jbrunet@baylibre.com [1]: https://lkml.kernel.org/r/20180630082829.qwnhkk4f76n32omm@mwanda [2]: https://lkml.kernel.org/r/20180630172241.10806-1-jbrunet@baylibre.com Jerome Brunet (2): ASoC: es7241: add dt-bindings documentation for the es7241 adc ASoC: es7241: add es7241 codec support .../devicetree/bindings/sound/everest,es7241.txt | 28 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es7241.c | 322 +++++++++++++++++++++ 4 files changed, 356 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.txt create mode 100644 sound/soc/codecs/es7241.c -- 2.14.4

2 8

[alsa-devel] [PATCH 1/2] jack: Allow build with older alsa-lib
by Takashi Iwai 03 Jul '18

03 Jul '18

The new API snd_pcm_ioplug_avail() doesn't appear prior to alsa-lib 1.1.7. Add an own code for allowing the build with alsa-lib 1.1.6. It has a very slight overhead than the original snd_pcm_ioplug_avail() but should be OK as a workaround. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- jack/pcm_jack.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index 98a6f7ebb82e..fbc9dafc594b 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -58,6 +58,15 @@ typedef struct { static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); +/* snd_pcm_ioplug_avail() was introduced after alsa-lib 1.1.6 */ +#if SND_LIB_VERSION < 0x10107 +static snd_pcm_uframes_t snd_pcm_ioplug_avail(const snd_pcm_ioplug_t *io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + return io->buffer_size - snd_pcm_ioplug_hw_avail(io, hw_ptr, appl_ptr); +} +#endif static int pcm_poll_block_check(snd_pcm_ioplug_t *io) { -- 2.18.0

1 2

[alsa-devel] [PATCH] ASoC: rt5682: add button detection mode control
by Bard Liao 03 Jul '18

03 Jul '18

We are currently using power saving mode for button detection. However, it will impact the headset recording performance. This patch will switch button detection to normal mode in capture and switch to power saving mode in the end of capture. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5682.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index baad177..640d400 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -857,6 +857,8 @@ static int rt5682_button_detect(struct snd_soc_component *component) btn_type = val & 0xfff0; snd_soc_component_write(component, RT5682_4BTN_IL_CMD_1, val); pr_debug("%s btn_type=%x\n", __func__, btn_type); + snd_soc_component_update_bits(component, + RT5682_SAR_IL_CMD_2, 0x10, 0x10); return btn_type; } @@ -1645,6 +1647,8 @@ static const struct snd_soc_dapm_widget rt5682_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5682_STO1_ADC_DIG_VOL, RT5682_R_MUTE_SFT, 1, rt5682_sto1_adc_r_mix, ARRAY_SIZE(rt5682_sto1_adc_r_mix)), + SND_SOC_DAPM_SUPPLY("BTN Detection Mode", RT5682_SAR_IL_CMD_1, + 14, 1, NULL, 0), /* ADC PGA */ SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1807,6 +1811,8 @@ static const struct snd_soc_dapm_route rt5682_dapm_routes[] = { {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + {"ADC Stereo1 Filter", NULL, "BTN Detection Mode"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: pxa-ssp: mark expected switch fall-through" to the asoc tree
by Mark Brown 03 Jul '18

03 Jul '18

The patch ASoC: pxa-ssp: mark expected switch fall-through has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e0431de301cbb8e3915261dfff4d0b072738de69 Mon Sep 17 00:00:00 2001 From: "Gustavo A. R. Silva" <gustavo(a)embeddedor.com> Date: Mon, 2 Jul 2018 07:17:07 -0500 Subject: [PATCH] ASoC: pxa-ssp: mark expected switch fall-through In preparation to enabling -Wimplicit-fallthrough, mark switch cases where we are expecting to fall through. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/pxa/pxa-ssp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index f8339bb01251..ff1e0bd8d407 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -470,6 +470,7 @@ static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; + /* fall through */ case SND_SOC_DAIFMT_DSP_B: sscr0 |= SSCR0_MOD | SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; -- 2.18.0.rc2

1 0

[alsa-devel] [PATCH 1/6] ASoC: AMD: Always stop ch2 first
by Daniel Kurtz 03 Jul '18

03 Jul '18

Commit 6b116dfb4633a ("ASoC: AMD: make channel 1 dma as circular") made both channels circular, so this comment and logic no longer applies. Always stop ch2 (the channel closest to the output) before ch1. This ensures that the downstream circular DMA channel does not continue to play/capture repeated samples after the upstream circular DMA channel has already stopped. Signed-off-by: Daniel Kurtz <djkurtz(a)chromium.org> --- sound/soc/amd/acp-pcm-dma.c | 17 ++--------------- 1 file changed, 2 insertions(+), 15 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 3c3d398d0d0b1..4665ae12e74e4 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1067,21 +1067,8 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: case SNDRV_PCM_TRIGGER_SUSPEND: - /* For playback, non circular dma should be stopped first - * i.e Sysram to acp dma transfer channel(rtd->ch1) should be - * stopped before stopping cirular dma which is acp sram to i2s - * fifo dma transfer channel(rtd->ch2). Where as in Capture - * scenario, i2s fifo to acp sram dma channel(rtd->ch2) stopped - * first before stopping acp sram to sysram which is circular - * dma(rtd->ch1). - */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - acp_dma_stop(rtd->acp_mmio, rtd->ch1); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch2); - } else { - acp_dma_stop(rtd->acp_mmio, rtd->ch2); - ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); - } + acp_dma_stop(rtd->acp_mmio, rtd->ch2); + ret = acp_dma_stop(rtd->acp_mmio, rtd->ch1); rtd->bytescount = 0; break; default: -- 2.18.0.399.gad0ab374a1-goog

2 11

[alsa-devel] [PATCH v2 0/3] ASoC: es7134: driver updates
by Jerome Brunet 03 Jul '18

03 Jul '18

This patchset fixes issues of v1 [0] around power supply requirement and add es7154 support. AVDD is renamed to PVDD to better match the component datasheet. v1 assumed optional power supplies, which was a wrong thing do. Now VDD, is required on all chip variant and PVDD is required on the es7154. [0]: https://lkml.kernel.org/r/20180629150924.18197-1-jbrunet@baylibre.com Jerome Brunet (3): ASoC: es7134: update DT binding with new compatible and supplies ASoC: es7134: correct required power supplies ASoC: es7134: add support for the es7154 .../devicetree/bindings/sound/everest,es7134.txt | 7 +- sound/soc/codecs/es7134.c | 104 ++++++++++++++++++++- 2 files changed, 106 insertions(+), 5 deletions(-) -- 2.14.4

2 6

[alsa-devel] [PATCH 0/2] ASoC: atmel-i2s: Remove unnecessary audio PLL clock
by Codrin Ciubotariu 03 Jul '18

03 Jul '18

Thanks to Quentin Schulz's patch: commit 1a1a36d72e3d ("clk: at91: clk-generated: make gclk determine audio_pll rate") the generated clock (gclk) itself is able to configure the audio PLL clock (aclk). For this reason, the I2S driver no longer needs to set aclk, but only gclk. This patchet removes all references to the audio PLL clock from the driver and updates the driver's bindings. Codrin Ciubotariu (2): ASoC: atmel-i2s: Remove unnecessary audio PLL clock (aclk) ASoC: atmel-i2s: dt-bindings: Remove unnecessary phandle to aclk .../devicetree/bindings/sound/atmel-i2s.txt | 11 ++--- sound/soc/atmel/atmel-i2s.c | 46 +++---------------- 2 files changed, 12 insertions(+), 45 deletions(-) -- 2.17.1

2 4

[alsa-devel] [PATCH - JACK IO plug 1/2 v2] jack: Avoid call to snd_pcm_avail_update() from JACK thread
by twischer＠de.adit-jv.com 03 Jul '18

03 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> snd_pcm_avail_update() can call snd_pcm_jack_stop() but snd_pcm_jack_stop() should not be called by the JACK thread. It should only be called by the thread how has called snd_pcm_jack_start(). In addition the execution of snd_pcm_avail_update() can take a while. Therefore it should not be called by the JACK thread to not block this thread. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- v2 uses snd_pcm_uframes_t as type for avail and does not check for avail < 0 becasue snd_pcm_ioplug_avail() will never retrun a value < 0 diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index e3df4d2..5a57e13 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -62,13 +62,13 @@ static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); static int pcm_poll_block_check(snd_pcm_ioplug_t *io) { static char buf[32]; - snd_pcm_sframes_t avail; + snd_pcm_uframes_t avail; snd_pcm_jack_t *jack = io->private_data; if (io->state == SND_PCM_STATE_RUNNING || (io->state == SND_PCM_STATE_PREPARED && io->stream == SND_PCM_STREAM_CAPTURE)) { - avail = snd_pcm_avail_update(io->pcm); - if (avail >= 0 && avail < jack->min_avail) { + avail = snd_pcm_ioplug_avail(io, jack->hw_ptr, io->appl_ptr); + if (avail < jack->min_avail) { while (read(io->poll_fd, &buf, sizeof(buf)) == sizeof(buf)) ; return 1; @@ -81,11 +81,11 @@ static int pcm_poll_block_check(snd_pcm_ioplug_t *io) static int pcm_poll_unblock_check(snd_pcm_ioplug_t *io) { static char buf[1]; - snd_pcm_sframes_t avail; + snd_pcm_uframes_t avail; snd_pcm_jack_t *jack = io->private_data; - avail = snd_pcm_avail_update(io->pcm); - if (avail < 0 || avail >= jack->min_avail) { + avail = snd_pcm_ioplug_avail(io, jack->hw_ptr, io->appl_ptr); + if (avail >= jack->min_avail) { write(jack->fd, &buf, 1); return 1; } -- 2.7.4

1 0

[alsa-devel] [PATCH - JACK IO plug 0/3] snd_pcm_drain support
by twischer＠de.adit-jv.com 03 Jul '18

03 Jul '18

The following patches are required to support draining with the JACK IO plugin. Without these patches $ speaker-test -D jack -c2 will freeze. It is required to support the draining feature implemented in commit ce2095c41f2 (pcm: ioplug: Implement proper drain behavior)

2 8

[alsa-devel] [PATCH] snd_interval_refine_first/last: exclude value only if also excluded before
by twischer＠de.adit-jv.com 03 Jul '18

03 Jul '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this commit the following intervals [x y), (x y) were be replaced to (y-1 y) by snd_interval_refine_last(). This was also done if y-1 is part of the previous interval. With this changes it will be replaced with [y-1 y) in case of y-1 is part of the previous interval. A similar behavior will be used for snd_interval_refine_first(). This solves the issue reported here: https://bugzilla.opensuse.org/show_bug.cgi?id=1033179 and work arounded with commit e736715 ("pcm: dmix: Disable var_periodsize as default"). I am able to reproduce the issue with a simplified aplay use case using the following configuration: pcm_slave.adr3_tdm_8ch { pcm { type hw card "Loopback" device 0 } rate 48000 period_size 128 buffer_size 1024 channels 2 } pcm.dshare_Playback_3 { type dmix ipc_key 600 ipc_perm 0660 ipc_gid audio var_periodsize true slave adr3_tdm_8ch } pcm.AdevAcousticoutSpeech { type rate slave.pcm dshare_Playback_3 slave.rate 48000 } $ modprobe snd_aloop $ aplay -v --period-size=352 -c2 -fS16_LE -r22500 -DAdevAcousticoutSpeech /dev/urandom ... Rule 9 (0xffff91d1f230): PERIODS=(0 2) -> NONE BUFFER_SIZE=480 PERIOD_SIZE=[240 240] refine_soft 'AdevAcousticoutSpeech' (end--22) ... aplay: ../../alsa-utils-1.1.5/aplay/aplay.c:1390: set_params: Assertion `err >= 0' failed. Aborted by signal Aborted... The following stack trace shows where the -EINVAL will be thrown: __snd_pcm_hw_params_set_period_size_near() snd1_pcm_hw_param_set_near() snd1_pcm_hw_param_set_last() snd1_pcm_hw_refine_slave() snd1_pcm_hw_refine_soft() snd_pcm_hw_rule_div() snd1_interval_refine() This issue exists due to PERIODS does not include 2 Rule 9 (0xffff91d1f230): PERIODS=(0 9) -> (0 2) BUFFER_SIZE=[120 480] PERIOD_SIZE=(240 241) because of an invalid integer inverval of PERIOD_SIZE of (240 241). This interval is set by snd_interval_refine_last(). Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- src/pcm/interval.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/src/pcm/interval.c b/src/pcm/interval.c index 95cedd5..ef4c2ed 100644 --- a/src/pcm/interval.c +++ b/src/pcm/interval.c @@ -200,27 +200,33 @@ int snd_interval_refine(snd_interval_t *i, const snd_interval_t *v) int snd_interval_refine_first(snd_interval_t *i) { + const unsigned int last_max = i->max; + if (snd_interval_empty(i)) return -ENOENT; if (snd_interval_single(i)) return 0; i->max = i->min; - i->openmax = i->openmin; - if (i->openmax) + if (i->openmin) i->max++; + /* only exclude max value if also excluded before refine */ + i->openmax = (i->openmax && i->max >= last_max); return 1; } int snd_interval_refine_last(snd_interval_t *i) { + const unsigned int last_min = i->min; + if (snd_interval_empty(i)) return -ENOENT; if (snd_interval_single(i)) return 0; i->min = i->max; - i->openmin = i->openmax; - if (i->openmin) + if (i->openmax) i->min--; + /* only exclude min value if also excluded before refine */ + i->openmin = (i->openmin && i->min <= last_min); return 1; } -- 2.7.4

2 1

[alsa-devel] [PATCH v3 24/27] devicetree: fix a series of wrong file references
by Mauro Carvalho Chehab 03 Jul '18

03 Jul '18

As files got renamed, their references broke. Manually fix a series of broken refs at the DT bindings. Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> --- .../devicetree/bindings/input/rmi4/rmi_2d_sensor.txt | 2 +- Documentation/devicetree/bindings/mfd/sun6i-prcm.txt | 2 +- .../devicetree/bindings/pci/hisilicon-pcie.txt | 2 +- Documentation/devicetree/bindings/pci/kirin-pcie.txt | 2 +- .../devicetree/bindings/pci/pci-keystone.txt | 4 ++-- .../devicetree/bindings/sound/st,stm32-i2s.txt | 2 +- .../devicetree/bindings/sound/st,stm32-sai.txt | 2 +- MAINTAINERS | 12 ++++++------ 8 files changed, 14 insertions(+), 14 deletions(-) diff --git a/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt b/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt index f2c30c8b725d..9afffbdf6e28 100644 --- a/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt +++ b/Documentation/devicetree/bindings/input/rmi4/rmi_2d_sensor.txt @@ -12,7 +12,7 @@ Additional documentation for F11 can be found at: http://www.synaptics.com/sites/default/files/511-000136-01-Rev-E-RMI4-Inter… Optional Touch Properties: -Description in Documentation/devicetree/bindings/input/touch +Description in Documentation/devicetree/bindings/input/touchscreen - touchscreen-inverted-x - touchscreen-inverted-y - touchscreen-swapped-x-y diff --git a/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt b/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt index 4d21ffdb0fc1..daa091c2e67b 100644 --- a/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt +++ b/Documentation/devicetree/bindings/mfd/sun6i-prcm.txt @@ -8,7 +8,7 @@ Required properties: - reg: The PRCM registers range The prcm node may contain several subdevices definitions: - - see Documentation/devicetree/clk/sunxi.txt for clock devices + - see Documentation/devicetree/bindings/clock/sunxi.txt for clock devices - see Documentation/devicetree/bindings/reset/allwinner,sunxi-clock-reset.txt for reset controller devices diff --git a/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt b/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt index 7bf9df047a1e..0dcb87d6554f 100644 --- a/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt +++ b/Documentation/devicetree/bindings/pci/hisilicon-pcie.txt @@ -3,7 +3,7 @@ HiSilicon Hip05 and Hip06 PCIe host bridge DT description HiSilicon PCIe host controller is based on the Synopsys DesignWare PCI core. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt. +Documentation/devicetree/bindings/pci/designware-pcie.txt. Additional properties are described here: diff --git a/Documentation/devicetree/bindings/pci/kirin-pcie.txt b/Documentation/devicetree/bindings/pci/kirin-pcie.txt index 6e217c63123d..6bbe43818ad5 100644 --- a/Documentation/devicetree/bindings/pci/kirin-pcie.txt +++ b/Documentation/devicetree/bindings/pci/kirin-pcie.txt @@ -3,7 +3,7 @@ HiSilicon Kirin SoCs PCIe host DT description Kirin PCIe host controller is based on the Synopsys DesignWare PCI core. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt. +Documentation/devicetree/bindings/pci/designware-pcie.txt. Additional properties are described here: diff --git a/Documentation/devicetree/bindings/pci/pci-keystone.txt b/Documentation/devicetree/bindings/pci/pci-keystone.txt index 7e05487544ed..3d4a209b0fd0 100644 --- a/Documentation/devicetree/bindings/pci/pci-keystone.txt +++ b/Documentation/devicetree/bindings/pci/pci-keystone.txt @@ -3,9 +3,9 @@ TI Keystone PCIe interface Keystone PCI host Controller is based on the Synopsys DesignWare PCI hardware version 3.65. It shares common functions with the PCIe DesignWare core driver and inherits common properties defined in -Documentation/devicetree/bindings/pci/designware-pci.txt +Documentation/devicetree/bindings/pci/designware-pcie.txt -Please refer to Documentation/devicetree/bindings/pci/designware-pci.txt +Please refer to Documentation/devicetree/bindings/pci/designware-pcie.txt for the details of DesignWare DT bindings. Additional properties are described here as well as properties that are not applicable. diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt index 4bda52042402..58c341300552 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt @@ -18,7 +18,7 @@ Required properties: See Documentation/devicetree/bindings/dma/stm32-dma.txt. - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt Optional properties: - resets: Reference to a reset controller asserting the reset controller diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt index f301cdf0b7e6..3a3fc506e43a 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt @@ -37,7 +37,7 @@ SAI subnodes required properties: "tx": if sai sub-block is configured as playback DAI "rx": if sai sub-block is configured as capture DAI - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/pinctrl-stm32.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt SAI subnodes Optional properties: - st,sync: specify synchronization mode. diff --git a/MAINTAINERS b/MAINTAINERS index 67641f5bb373..69c9e9924902 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6965,7 +6965,7 @@ IIO MULTIPLEXER M: Peter Rosin <peda(a)axentia.se> L: linux-iio(a)vger.kernel.org S: Maintained -F: Documentation/devicetree/bindings/iio/multiplexer/iio-mux.txt +F: Documentation/devicetree/bindings/iio/multiplexer/io-channel-mux.txt F: drivers/iio/multiplexer/iio-mux.c IIO SUBSYSTEM AND DRIVERS @@ -9695,7 +9695,7 @@ MXSFB DRM DRIVER M: Marek Vasut <marex(a)denx.de> S: Supported F: drivers/gpu/drm/mxsfb/ -F: Documentation/devicetree/bindings/display/mxsfb-drm.txt +F: Documentation/devicetree/bindings/display/mxsfb.txt MYRICOM MYRI-10G 10GbE DRIVER (MYRI10GE) M: Chris Lee <christopher.lee(a)cspi.com> @@ -10884,7 +10884,7 @@ M: Will Deacon <will.deacon(a)arm.com> L: linux-pci(a)vger.kernel.org L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/pci/controller-generic-pci.txt +F: Documentation/devicetree/bindings/pci/host-generic-pci.txt F: drivers/pci/controller/pci-host-common.c F: drivers/pci/controller/pci-host-generic.c @@ -11065,7 +11065,7 @@ M: Xiaowei Song <songxiaowei(a)hisilicon.com> M: Binghui Wang <wangbinghui(a)hisilicon.com> L: linux-pci(a)vger.kernel.org S: Maintained -F: Documentation/devicetree/bindings/pci/pcie-kirin.txt +F: Documentation/devicetree/bindings/pci/kirin-pcie.txt F: drivers/pci/controller/dwc/pcie-kirin.c PCIE DRIVER FOR HISILICON STB @@ -12456,7 +12456,7 @@ L: linux-crypto(a)vger.kernel.org L: linux-samsung-soc(a)vger.kernel.org S: Maintained F: drivers/crypto/exynos-rng.c -F: Documentation/devicetree/bindings/crypto/samsung,exynos-rng4.txt +F: Documentation/devicetree/bindings/rng/samsung,exynos4-rng.txt SAMSUNG EXYNOS TRUE RANDOM NUMBER GENERATOR (TRNG) DRIVER M: Łukasz Stelmach <l.stelmach(a)samsung.com> @@ -13570,7 +13570,7 @@ F: drivers/*/stm32-*timer* F: drivers/pwm/pwm-stm32* F: include/linux/*/stm32-*tim* F: Documentation/ABI/testing/*timer-stm32 -F: Documentation/devicetree/bindings/*/stm32-*timer +F: Documentation/devicetree/bindings/*/stm32-*timer* F: Documentation/devicetree/bindings/pwm/pwm-stm32* STMMAC ETHERNET DRIVER -- 2.17.1

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[alsa-devel] [PATCH 0/5] ASoC: simple/audio card: convert to SPDX identifiers
by Kuninori Morimoto 03 Jul '18

03 Jul '18

Hi Mark These convert to SPDX identifiers for simple/audio cards Kuninori Morimoto (5): ASoC: simple-card: convert to SPDX identifiers ASoC: simple-card-utils: convert to SPDX identifiers ASoC: simple-scu-card.c: convert to SPDX identifiers ASoC: audio-graph-card.c: convert to SPDX identifiers ASoC: audio-graph-scu-card.c: convert to SPDX identifiers include/sound/simple_card.h | 7 ++----- include/sound/simple_card_utils.h | 8 +++----- sound/soc/generic/audio-graph-card.c | 21 +++++++++------------ sound/soc/generic/audio-graph-scu-card.c | 25 +++++++++++-------------- sound/soc/generic/simple-card-utils.c | 15 ++++++--------- sound/soc/generic/simple-card.c | 17 +++++++---------- sound/soc/generic/simple-scu-card.c | 21 +++++++++------------ 7 files changed, 47 insertions(+), 67 deletions(-) -- 2.7.4

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Re: [alsa-devel] linux-next: Tree for Jul 2 (sound/soc/pxa/)
by Randy Dunlap 02 Jul '18

02 Jul '18

[add other maintainers] On 07/02/18 09:09, Randy Dunlap wrote: > On 07/01/18 21:51, Stephen Rothwell wrote: >> Hi all, >> >> Changes since 20180629: >> > > on i386 or x86_64: > > builtin: > sound/soc/pxa/pxa-ssp.o:(.rodata+0x228): undefined reference to `pxa2xx_soc_pcm_new' > sound/soc/pxa/pxa-ssp.o:(.rodata+0x22c): undefined reference to `pxa2xx_pcm_free_dma_buffers' > sound/soc/pxa/pxa-ssp.o:(.rodata+0x250): undefined reference to `pxa2xx_pcm_ops' > /home/rdunlap/lnx/next/linux-next-20180702/Makefile:1023: recipe for target 'vmlinux' failed > > modular: > ERROR: "pxa2xx_pcm_ops" [sound/soc/pxa/snd-soc-pxa-ssp.ko] undefined! > ERROR: "pxa2xx_pcm_free_dma_buffers" [sound/soc/pxa/snd-soc-pxa-ssp.ko] undefined! > ERROR: "pxa2xx_soc_pcm_new" [sound/soc/pxa/snd-soc-pxa-ssp.ko] undefined! > > > x86_64 [modular] randconfig file is attached. > -- ~Randy

2 1

[alsa-devel] [PATCH v2] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 02 Jul '18

02 Jul '18

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Changes since V1 :- o Remove API method and alow drivers to set directly. o Validate time is driver supplied times. include/sound/pcm.h | 1 + sound/core/pcm_lib.c | 10 ++++++++-- 2 files changed, 9 insertions(+), 2 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..edad4a506b93 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + unsigned wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 44b5ae833082..6909b896a6a1 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1832,12 +1832,18 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = substream->wait_time; + } else { + wait_time = 10 * 1000; /* 10 secs */ + } + if (runtime->rate) { long t = runtime->period_size * 2 / runtime->rate; wait_time = max(t, wait_time); } - wait_time = msecs_to_jiffies(wait_time * 1000); + wait_time = msecs_to_jiffies(wait_time); } for (;;) { -- 2.17.1

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[alsa-devel] [sound:for-linus 15/15] sound/pci/hda/patch_hdmi.c:1638:18: error: implicit declaration of function 'pm_runtime_suspended'; did you mean 'pm_generic_suspend'?
by kbuild test robot 02 Jul '18

02 Jul '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-linus head: b0526c9f124fa2eada1c262743bf09edc3799a20 commit: b0526c9f124fa2eada1c262743bf09edc3799a20 [15/15] ALSA: hda - Handle pm failure during hotplug config: x86_64-randconfig-x010-201826 (attached as .config) compiler: gcc-7 (Debian 7.3.0-16) 7.3.0 reproduce: git checkout b0526c9f124fa2eada1c262743bf09edc3799a20 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): In file included from include/linux/init.h:5:0, from sound/pci/hda/patch_hdmi.c:32: sound/pci/hda/patch_hdmi.c: In function 'hdmi_present_sense': >> sound/pci/hda/patch_hdmi.c:1638:18: error: implicit declaration of function 'pm_runtime_suspended'; did you mean 'pm_generic_suspend'? [-Werror=implicit-function-declaration] if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { ^ include/linux/compiler.h:58:30: note: in definition of macro '__trace_if' if (__builtin_constant_p(!!(cond)) ? !!(cond) : \ ^~~~ sound/pci/hda/patch_hdmi.c:1638:3: note: in expansion of macro 'if' if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { ^~ cc1: some warnings being treated as errors vim +1638 sound/pci/hda/patch_hdmi.c 1629 1630 static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll) 1631 { 1632 struct hda_codec *codec = per_pin->codec; 1633 int ret; 1634 1635 /* no temporary power up/down needed for component notifier */ 1636 if (!codec_has_acomp(codec)) { 1637 ret = snd_hda_power_up_pm(codec); > 1638 if (ret < 0 && pm_runtime_suspended(hda_codec_dev(codec))) { 1639 snd_hda_power_down_pm(codec); 1640 return false; 1641 } 1642 } 1643 1644 if (codec_has_acomp(codec)) { 1645 sync_eld_via_acomp(codec, per_pin); 1646 ret = false; /* don't call snd_hda_jack_report_sync() */ 1647 } else { 1648 ret = hdmi_present_sense_via_verbs(per_pin, repoll); 1649 } 1650 1651 if (!codec_has_acomp(codec)) 1652 snd_hda_power_down_pm(codec); 1653 1654 return ret; 1655 } 1656 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH REPOST 0/2] ALSA: use irqsave in USB's completion callback
by Sebastian Andrzej Siewior 02 Jul '18

02 Jul '18

This series is about using _irqsave() primitives in the completion callback in order to get rid of local_irq_save() in __usb_hcd_giveback_urb(). Sebastian

4 5

[alsa-devel] alsa-lib-devel sources for centos 7 and backward compatibility
by maxim maxim 02 Jul '18

02 Jul '18

Hello, I have some issues with alsa-lib-devel for centos 7 , I'm working with gcc7.2.1 , I achieved the alsa-lb and compiled with gcc-7.2.1, my question where I can find alsa-lib-devel sources to compile it with my compiler? Additional question that I have it's about backward compatibility I saw that alsa-lib-devel-1.1.4.1-2.el7.x86_64.rpm doesn't have file alsa/iatomic.h as it was for Centos 5, could someone explain me please were I can achieve the functions that were in alsa/iatomic.h Thanks in advance, Max

1 0

[alsa-devel] [PATCH 0/5] ASoC: es7134: driver updates
by Jerome Brunet 02 Jul '18

02 Jul '18

This patchset provides several updates for the es7134 driver It removes unsupported sample rates, add runtime check of the mclk-fs ratio as documented in the datasheet and adds the component power supplies. Finally support for the es7154 is added. Jerome Brunet (5): ASoC: es7134: remove 64kHz rate from the supported rates ASoC: es7134: check if mclk rate is valid ASoC: es7134: update DT binding with new compatible and supplies ASoC: es7134: Add VDD and AVDD power supplies ASoC: es7134: add support for the es7154 .../devicetree/bindings/sound/everest,es7134.txt | 8 +- sound/soc/codecs/es7134.c | 176 ++++++++++++++++++++- 2 files changed, 179 insertions(+), 5 deletions(-) -- 2.14.4

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[alsa-devel] [PATCH v2 0/2] ASoC: add es7241 support
by Jerome Brunet 02 Jul '18

02 Jul '18

Add support for the es7241 which is a simple 2 channels analog to digital converter. Changes since v1: [0] * Add missing break as reported by smatch [1] My device is hardwired in slave mode, which probably why this slipped through unnoticed. [0]: https://lkml.kernel.org/r/20180629152444.21525-1-jbrunet@baylibre.com [1]: https://lkml.kernel.org/r/20180630082829.qwnhkk4f76n32omm@mwanda Jerome Brunet (2): ASoC: es7241: add dt-bindings documentation for the es7241 adc ASoC: es7241: add es7241 codec support .../devicetree/bindings/sound/everest,es7241.txt | 23 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es7241.c | 308 +++++++++++++++++++++ 4 files changed, 337 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/everest,es7241.txt create mode 100644 sound/soc/codecs/es7241.c -- 2.14.4

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[alsa-devel] [PATCH 00/14] ASoC: soc-xxx: convert to SPDX identifiers
by Kuninori Morimoto 02 Jul '18

02 Jul '18

Hi Mark These convert to SPDX identifiers for soc-xxx files Kuninori Morimoto (14): ASoC: soc-io.c: convert to SPDX identifiers ASoC: soc-dai.h: convert to SPDX identifiers ASoC: soc-ops.c: convert to SPDX identifiers ASoC: soc-pcm.c: convert to SPDX identifiers ASoC: soc-dpcm.h: convert to SPDX identifiers ASoC: soc-jack.c: convert to SPDX identifiers ASoC: soc-utils.c: convert to SPDX identifiers ASoC: soc-devres.c: convert to SPDX identifiers ASoC: soc-acpi: convert to SPDX identifiers ASoC: soc-core: convert to SPDX identifiers ASoC: soc-dapm: convert to SPDX identifiers ASoC: soc-topology: convert to SPDX identifiers ASoC: soc-compress: convert to SPDX identifiers ASoC: soc-generic-dmaengine-pcm: convert to SPDX identifiers include/sound/compress_driver.h | 21 +++------------- include/sound/dmaengine_pcm.h | 14 +++-------- include/sound/soc-acpi-intel-match.h | 14 ++--------- include/sound/soc-acpi.h | 13 ++-------- include/sound/soc-dai.h | 7 ++---- include/sound/soc-dapm.h | 11 +++----- include/sound/soc-dpcm.h | 7 ++---- include/sound/soc-topology.h | 7 ++---- include/sound/soc.h | 11 +++----- sound/soc/soc-acpi.c | 20 ++++----------- sound/soc/soc-compress.c | 24 +++++++----------- sound/soc/soc-core.c | 41 ++++++++++++++---------------- sound/soc/soc-dapm.c | 42 ++++++++++++++----------------- sound/soc/soc-devres.c | 15 ++++------- sound/soc/soc-generic-dmaengine-pcm.c | 19 ++++---------- sound/soc/soc-io.c | 19 ++++++-------- sound/soc/soc-jack.c | 19 ++++++-------- sound/soc/soc-ops.c | 29 +++++++++------------ sound/soc/soc-pcm.c | 28 ++++++++------------- sound/soc/soc-topology.c | 47 ++++++++++++++++------------------- sound/soc/soc-utils.c | 22 ++++++---------- 21 files changed, 150 insertions(+), 280 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: ac97: convert to SPDX identifiers
by Kuninori Morimoto 02 Jul '18

02 Jul '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- include/sound/ac97/codec.h | 8 +++----- include/sound/ac97/compat.h | 9 +++------ include/sound/ac97/controller.h | 8 +++----- include/sound/ac97/regs.h | 20 ++------------------ include/sound/ac97_codec.h | 25 +++++-------------------- sound/soc/soc-ac97.c | 29 ++++++++++++----------------- 6 files changed, 28 insertions(+), 71 deletions(-) diff --git a/include/sound/ac97/codec.h b/include/sound/ac97/codec.h index ec04be9..9792d25 100644 --- a/include/sound/ac97/codec.h +++ b/include/sound/ac97/codec.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * 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. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> */ + #ifndef __SOUND_AC97_CODEC2_H #define __SOUND_AC97_CODEC2_H diff --git a/include/sound/ac97/compat.h b/include/sound/ac97/compat.h index 1351cba..57e19af 100644 --- a/include/sound/ac97/compat.h +++ b/include/sound/ac97/compat.h @@ -1,14 +1,11 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * 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. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> * * This file is for backward compatibility with snd_ac97 structure and its * multiple usages, such as the snd_ac97_bus and snd_ac97_build_ops. - * */ + #ifndef AC97_COMPAT_H #define AC97_COMPAT_H diff --git a/include/sound/ac97/controller.h b/include/sound/ac97/controller.h index b36ecdd..06b5afb 100644 --- a/include/sound/ac97/controller.h +++ b/include/sound/ac97/controller.h @@ -1,10 +1,8 @@ -/* - * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> +/* SPDX-License-Identifier: GPL-2.0 * - * 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. + * Copyright (C) 2016 Robert Jarzmik <robert.jarzmik(a)free.fr> */ + #ifndef AC97_CONTROLLER_H #define AC97_CONTROLLER_H diff --git a/include/sound/ac97/regs.h b/include/sound/ac97/regs.h index 9a4fa0c..843f73f 100644 --- a/include/sound/ac97/regs.h +++ b/include/sound/ac97/regs.h @@ -1,27 +1,11 @@ -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.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. - * - * 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 - * */ - /* * AC'97 codec registers */ diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h index 89d311a..cc38399 100644 --- a/include/sound/ac97_codec.h +++ b/include/sound/ac97_codec.h @@ -1,30 +1,15 @@ -#ifndef __SOUND_AC97_CODEC_H -#define __SOUND_AC97_CODEC_H - -/* +/* SPDX-License-Identifier: GPL-2.0+ + * * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * Universal interface for Audio Codec '97 * * For more details look to AC '97 component specification revision 2.1 * by Intel Corporation (http://developer.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. - * - * 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 - * */ +#ifndef __SOUND_AC97_CODEC_H +#define __SOUND_AC97_CODEC_H + #include <linux/bitops.h> #include <linux/device.h> #include <linux/workqueue.h> diff --git a/sound/soc/soc-ac97.c b/sound/soc/soc-ac97.c index 3f424f2..c086786 100644 --- a/sound/soc/soc-ac97.c +++ b/sound/soc/soc-ac97.c @@ -1,20 +1,15 @@ -/* - * soc-ac97.c -- ALSA SoC Audio Layer AC97 support - * - * Copyright 2005 Wolfson Microelectronics PLC. - * Copyright 2005 Openedhand Ltd. - * Copyright (C) 2010 Slimlogic Ltd. - * Copyright (C) 2010 Texas Instruments Inc. - * - * Author: Liam Girdwood <lrg(a)slimlogic.co.uk> - * with code, comments and ideas from :- - * Richard Purdie <richard(a)openedhand.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. - */ +// SPDX-License-Identifier: GPL-2.0+ +// +// soc-ac97.c -- ALSA SoC Audio Layer AC97 support +// +// Copyright 2005 Wolfson Microelectronics PLC. +// Copyright 2005 Openedhand Ltd. +// Copyright (C) 2010 Slimlogic Ltd. +// Copyright (C) 2010 Texas Instruments Inc. +// +// Author: Liam Girdwood <lrg(a)slimlogic.co.uk> +// with code, comments and ideas from :- +// Richard Purdie <richard(a)openedhand.com> #include <linux/ctype.h> #include <linux/delay.h> -- 2.7.4

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[alsa-devel] [PATCH] ASoC: pxa: select SND_PXA2XX_LIB for drivers that depend on it
by Daniel Mack 02 Jul '18

02 Jul '18

Commit d767d3ce5c48b ("ASoC: pxa: provide PCM ops for ssp, i2s and ac97 components") created a build-time dependency to SND_PXA2XX_LIB but missed to reflect that in Kconfig. Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/Kconfig | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 95dcf97a6271..2fc02c227f69 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -24,15 +24,18 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS + select SND_PXA2XX_LIB select SND_PXA2XX_LIB_AC97 select SND_SOC_AC97_BUS config SND_PXA2XX_SOC_I2S + select SND_PXA2XX_LIB tristate config SND_PXA_SOC_SSP tristate "Soc Audio via PXA2xx/PXA3xx SSP ports" select PXA_SSP + select SND_PXA2XX_LIB config SND_MMP_SOC_SSPA tristate -- 2.17.1

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