October 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: rt5640: Fill up the IN3's support
by Oder Chiou 20 Oct '15

20 Oct '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5640.txt | 9 ++++++--- include/sound/rt5640.h | 3 ++- sound/soc/codecs/rt5640.c | 22 +++++++++++++++++++++- 3 files changed, 29 insertions(+), 5 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/rt5640.txt b/Documentation/devicetree/bindings/sound/rt5640.txt index bac4d9a..9e62f6e 100644 --- a/Documentation/devicetree/bindings/sound/rt5640.txt +++ b/Documentation/devicetree/bindings/sound/rt5640.txt @@ -14,7 +14,8 @@ Optional properties: - realtek,in1-differential - realtek,in2-differential - Boolean. Indicate MIC1/2 input are differential, rather than single-ended. +- realtek,in3-differential + Boolean. Indicate MIC1/2/3 input are differential, rather than single-ended. - realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. @@ -24,9 +25,11 @@ Pins on the device (for linking into audio routes) for RT5639/RT5640: * DMIC2 * MICBIAS1 * IN1P - * IN1R + * IN1N * IN2P - * IN2R + * IN2N + * IN3P + * IN3N * HPOL * HPOR * LOUTL diff --git a/include/sound/rt5640.h b/include/sound/rt5640.h index 59d26dd..e3c84b9 100644 --- a/include/sound/rt5640.h +++ b/include/sound/rt5640.h @@ -12,9 +12,10 @@ #define __LINUX_SND_RT5640_H struct rt5640_platform_data { - /* IN1 & IN2 can optionally be differential */ + /* IN1 & IN2 & IN3 can optionally be differential */ bool in1_diff; bool in2_diff; + bool in3_diff; bool dmic_en; bool dmic1_data_pin; /* 0 = IN1P; 1 = GPIO3 */ diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e1ceeb8..f2beb1a 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -405,11 +405,14 @@ static const struct snd_kcontrol_new rt5640_snd_controls[] = { SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5640_DAC1_DIG_VOL, RT5640_L_VOL_SFT, RT5640_R_VOL_SFT, 175, 0, dac_vol_tlv), - /* IN1/IN2 Control */ + /* IN1/IN2/IN3 Control */ SOC_SINGLE_TLV("IN1 Boost", RT5640_IN1_IN2, RT5640_BST_SFT1, 8, 0, bst_tlv), SOC_SINGLE_TLV("IN2 Boost", RT5640_IN3_IN4, RT5640_BST_SFT2, 8, 0, bst_tlv), + SOC_SINGLE_TLV("IN3 Boost", RT5640_IN1_IN2, + RT5640_BST_SFT2, 8, 0, bst_tlv), + /* INL/INR Volume Control */ SOC_DOUBLE_TLV("IN Capture Volume", RT5640_INL_INR_VOL, RT5640_INL_VOL_SFT, RT5640_INR_VOL_SFT, @@ -598,6 +601,8 @@ static const struct snd_kcontrol_new rt5640_rec_l_mix[] = { RT5640_M_HP_L_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("INL Switch", RT5640_REC_L2_MIXER, RT5640_M_IN_L_RM_L_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_L2_MIXER, + RT5640_M_BST2_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_L2_MIXER, RT5640_M_BST4_RM_L_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_L2_MIXER, @@ -611,6 +616,8 @@ static const struct snd_kcontrol_new rt5640_rec_r_mix[] = { RT5640_M_HP_R_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("INR Switch", RT5640_REC_R2_MIXER, RT5640_M_IN_R_RM_R_SFT, 1, 1), + SOC_DAPM_SINGLE("BST3 Switch", RT5640_REC_R2_MIXER, + RT5640_M_BST2_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST2 Switch", RT5640_REC_R2_MIXER, RT5640_M_BST4_RM_R_SFT, 1, 1), SOC_DAPM_SINGLE("BST1 Switch", RT5640_REC_R2_MIXER, @@ -1065,6 +1072,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN1N"), SND_SOC_DAPM_INPUT("IN2P"), SND_SOC_DAPM_INPUT("IN2N"), + SND_SOC_DAPM_INPUT("IN3P"), + SND_SOC_DAPM_INPUT("IN3N"), SND_SOC_DAPM_PGA("DMIC L1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC R1", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("DMIC L2", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -1081,6 +1090,8 @@ static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { RT5640_PWR_BST1_BIT, 0, NULL, 0), SND_SOC_DAPM_PGA("BST2", RT5640_PWR_ANLG2, RT5640_PWR_BST4_BIT, 0, NULL, 0), + SND_SOC_DAPM_PGA("BST3", RT5640_PWR_ANLG2, + RT5640_PWR_BST2_BIT, 0, NULL, 0), /* Input Volume */ SND_SOC_DAPM_PGA("INL VOL", RT5640_PWR_VOL, RT5640_PWR_IN_L_BIT, 0, NULL, 0), @@ -1310,6 +1321,7 @@ static const struct snd_soc_dapm_widget rt5639_specific_dapm_widgets[] = { static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"IN1P", NULL, "LDO2"}, {"IN2P", NULL, "LDO2"}, + {"IN3P", NULL, "LDO2"}, {"DMIC L1", NULL, "DMIC1"}, {"DMIC R1", NULL, "DMIC1"}, @@ -1320,18 +1332,22 @@ static const struct snd_soc_dapm_route rt5640_dapm_routes[] = { {"BST1", NULL, "IN1N"}, {"BST2", NULL, "IN2P"}, {"BST2", NULL, "IN2N"}, + {"BST3", NULL, "IN3P"}, + {"BST3", NULL, "IN3N"}, {"INL VOL", NULL, "IN2P"}, {"INR VOL", NULL, "IN2N"}, {"RECMIXL", "HPOL Switch", "HPOL"}, {"RECMIXL", "INL Switch", "INL VOL"}, + {"RECMIXL", "BST3 Switch", "BST3"}, {"RECMIXL", "BST2 Switch", "BST2"}, {"RECMIXL", "BST1 Switch", "BST1"}, {"RECMIXL", "OUT MIXL Switch", "OUT MIXL"}, {"RECMIXR", "HPOR Switch", "HPOR"}, {"RECMIXR", "INR Switch", "INR VOL"}, + {"RECMIXR", "BST3 Switch", "BST3"}, {"RECMIXR", "BST2 Switch", "BST2"}, {"RECMIXR", "BST1 Switch", "BST1"}, {"RECMIXR", "OUT MIXR Switch", "OUT MIXR"}, @@ -2260,6 +2276,10 @@ static int rt5640_i2c_probe(struct i2c_client *i2c, regmap_update_bits(rt5640->regmap, RT5640_IN3_IN4, RT5640_IN_DF2, RT5640_IN_DF2); + if (rt5640->pdata.in3_diff) + regmap_update_bits(rt5640->regmap, RT5640_IN1_IN2, + RT5640_IN_DF2, RT5640_IN_DF2); + rt5640->hp_mute = 1; return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt5640, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ALSA: hda - Fix deadlock at error in building PCM
by Takashi Iwai 20 Oct '15

20 Oct '15

The HDA codec driver issues snd_hda_codec_reset() at the error path of PCM build. This was needed in the earlier code base, but the recent rewrite to use the standard bus binding made this a deadlock: modprobe D 0000000000000005 0 720 716 0x00000080 Call Trace: [<ffffffff816a5dbe>] schedule+0x3e/0x90 [<ffffffff816a61a5>] schedule_preempt_disabled+0x15/0x20 [<ffffffff816a7ae5>] __mutex_lock_slowpath+0xb5/0x120 [<ffffffff816a7b6b>] mutex_lock+0x1b/0x30 [<ffffffff8148656b>] device_release_driver+0x1b/0x30 [<ffffffff81485c15>] bus_remove_device+0x105/0x180 [<ffffffff814822b9>] device_del+0x139/0x260 [<ffffffffa05e0ec5>] snd_hdac_device_unregister+0x25/0x30 [snd_hda_core] [<ffffffffa074fa6a>] snd_hda_codec_reset+0x2a/0x70 [snd_hda_codec] [<ffffffffa075007b>] snd_hda_codec_build_pcms+0x18b/0x1b0 [snd_hda_codec] [<ffffffffa074a44e>] hda_codec_driver_probe+0xbe/0x140 [snd_hda_codec] [<ffffffff81486ac4>] driver_probe_device+0x1f4/0x460 [<ffffffff81486dc0>] __driver_attach+0x90/0xa0 [<ffffffff81484844>] bus_for_each_dev+0x64/0xa0 [<ffffffff814862de>] driver_attach+0x1e/0x20 [<ffffffff81485e7b>] bus_add_driver+0x1eb/0x280 [<ffffffff81487680>] driver_register+0x60/0xe0 [<ffffffffa074a0da>] __hda_codec_driver_register+0x5a/0x60 [snd_hda_codec] [<ffffffffa070a01e>] realtek_driver_init+0x1e/0x1000 [snd_hda_codec_realtek] [<ffffffff810002f3>] do_one_initcall+0xb3/0x200 [<ffffffff816a1fc5>] do_init_module+0x60/0x1f8 [<ffffffff810ee5c3>] load_module+0x1653/0x1bd0 [<ffffffff810eed48>] SYSC_finit_module+0x98/0xc0 [<ffffffff810eed8e>] SyS_finit_module+0xe/0x10 [<ffffffff816aa032>] entry_SYSCALL_64_fastpath+0x16/0x75 The simple fix is just to remove this call, since we don't need to think about unbinding at there any longer. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=948758 Cc: <stable(a)vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_codec.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 37f43a1b34ef..a249d5486889 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3367,10 +3367,8 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) int dev, err; err = snd_hda_codec_parse_pcms(codec); - if (err < 0) { - snd_hda_codec_reset(codec); + if (err < 0) return err; - } /* attach a new PCM streams */ list_for_each_entry(cpcm, &codec->pcm_list_head, list) { -- 2.6.1

1 0

[alsa-devel] [PATCH V4 0/4] Some fixes against previous comments
by han.lu＠intel.com 20 Oct '15

20 Oct '15

From: "Lu, Han" <han.lu(a)intel.com> Some fixes and adjustments against previous comments of Takashi Lu, Han (4): BAT: Remove redundant message strings BAT: Use colon instead of comma for separation BAT: Change comments and interface of usage() BAT: Use dynamic temp file bat/bat.c | 106 +++++++++++++++++++++++++++++++---------------------------- bat/common.h | 2 +- 2 files changed, 57 insertions(+), 51 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH 0/6] HD-audio modalias support patchset
by Takashi Iwai 20 Oct '15

20 Oct '15

Hi, this is a patchset to add the support of automatic modalias creation for HD-audio codec drivers like other standard drivers. Currently, the HDA codec driver has put module alias manually due to ugly historical reasons. This patch finally moves to the saner way. Also it adds the udev modalias support so that it can be autoloaded in udev. The patchset was initiated by Intel people for SKL ASoC drivers. Now I brushed up and made it working for legacy drivers. The current patchset is found in topic/hda-modalias branch of sound git tree, too. thanks, Takashi === Subhransu S. Prusty (2): ALSA: hda - Add hduadio support to DEVTABLE ALSA: hda - Add hdaudio bus modalias support Takashi Iwai (3): ALSA: hda - Add api_version to hda_device_id struct ALSA: hda - Add a common helper to give the codec modalias string ALSA: hda - convert to hda_device_id Thierry Reding (1): ALSA: hda - Advertise MODALIAS in uevent include/linux/mod_devicetable.h | 8 ++ include/sound/hdaudio.h | 12 +-- scripts/mod/devicetable-offsets.c | 5 + scripts/mod/file2alias.c | 17 +++ sound/hda/hda_bus_type.c | 13 +++ sound/hda/hdac_device.c | 15 +++ sound/hda/hdac_sysfs.c | 8 ++ sound/pci/hda/hda_bind.c | 32 +++--- sound/pci/hda/hda_codec.h | 19 ++-- sound/pci/hda/hda_generic.c | 7 +- sound/pci/hda/patch_analog.c | 37 ++++--- sound/pci/hda/patch_ca0110.c | 15 ++- sound/pci/hda/patch_ca0132.c | 9 +- sound/pci/hda/patch_cirrus.c | 21 ++-- sound/pci/hda/patch_cmedia.c | 15 ++- sound/pci/hda/patch_conexant.c | 118 ++++++--------------- sound/pci/hda/patch_hdmi.c | 201 +++++++++++++---------------------- sound/pci/hda/patch_realtek.c | 122 ++++++++++------------ sound/pci/hda/patch_si3054.c | 39 +++---- sound/pci/hda/patch_sigmatel.c | 214 +++++++++++++++++++------------------- sound/pci/hda/patch_via.c | 153 ++++++++++----------------- 21 files changed, 475 insertions(+), 605 deletions(-) -- 2.6.1

2 9

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Convert to devm_snd_soc_register_card
by Axel Lin 20 Oct '15

20 Oct '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/intel/boards/skl_rt286.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 488c98e..3b6f48c 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -240,18 +240,11 @@ static int skylake_audio_probe(struct platform_device *pdev) { skylake_rt286.dev = &pdev->dev; - return snd_soc_register_card(&skylake_rt286); -} - -static int skylake_audio_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&skylake_rt286); - return 0; + return devm_snd_soc_register_card(&skylake_rt286); } static struct platform_driver skylake_audio = { .probe = skylake_audio_probe, - .remove = skylake_audio_remove, .driver = { .name = "skl_alc286s_i2s", }, -- 2.1.4

1 0

[alsa-devel] Implementing sub-stream multiplexing in DAI
by Ughreja, Rakesh A 19 Oct '15

19 Oct '15

Hi Mark, In order to implement sub-stream multiplexing as a part of single BE DAI I have following observations. 1. All BE DAI ops gets called only once even if two different FE DAIs are connected to a BE DAI. 2. BE DAI ops are always called with sub-stream 0. are these expected ? What is the best way to allow calling of BE DAI ops with different sub-stream ID so that I can do link configuration based on sub-stream parameters ? can you give some pointers ? Regards, Rakesh

2 1

[alsa-devel] [PATCH] ASoC: rockchip: spdif: Convert to use devm_snd_dmaengine_pcm_register
by Axel Lin 19 Oct '15

19 Oct '15

Use resource managed API then we can remove snd_dmaengine_pcm_unregister() and snd_soc_unregister_component() calls in .probe error path and .remove. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/rockchip/rockchip_spdif.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/soc/rockchip/rockchip_spdif.c b/sound/soc/rockchip/rockchip_spdif.c index 9d5c470..a38a302 100644 --- a/sound/soc/rockchip/rockchip_spdif.c +++ b/sound/soc/rockchip/rockchip_spdif.c @@ -355,16 +355,14 @@ static int rk_spdif_probe(struct platform_device *pdev) goto err_pm_runtime; } - ret = snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); if (ret) { dev_err(&pdev->dev, "Could not register PCM\n"); - goto err_pcm_register; + goto err_pm_runtime; } return 0; -err_pcm_register: - snd_dmaengine_pcm_unregister(&pdev->dev); err_pm_runtime: pm_runtime_disable(&pdev->dev); @@ -381,8 +379,6 @@ static int rk_spdif_remove(struct platform_device *pdev) clk_disable_unprepare(spdif->mclk); clk_disable_unprepare(spdif->hclk); - snd_dmaengine_pcm_unregister(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); return 0; } -- 2.1.4

2 1

[alsa-devel] [PATCH] ASoC: ad193x-spi: Add adau1328 to ad193x_spi_id table
by Axel Lin 19 Oct '15

19 Oct '15

This driver also supports adau1328, thus add adau1328 to ad193x_spi_id. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ad193x-spi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c index b55055a..431f95d 100644 --- a/sound/soc/codecs/ad193x-spi.c +++ b/sound/soc/codecs/ad193x-spi.c @@ -41,6 +41,7 @@ static const struct spi_device_id ad193x_spi_id[] = { { "ad1934", AD1934 }, { "ad1938", AD193X }, { "ad1939", AD193X }, + { "adau1328", AD193X }, { } }; MODULE_DEVICE_TABLE(spi, ad193x_spi_id); -- 2.1.4

3 2

[alsa-devel] [PATCH v3] ASoC: Intel: Skylake - Add Skylake RT286 I2S machine driver
by Vinod Koul 19 Oct '15

19 Oct '15

From: Omair M Abdullah <omair.m.abdullah(a)intel.com> Add the SKL I2S machine driver using Realtek ALC286S codec in I2S mode. Signed-off-by: Omair M Abdullah <omair.m.abdullah(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- v3: fixed sparse warns v2: added depends on X86 and ACPI based on LKP reports sound/soc/intel/Kconfig | 13 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/skl_rt286.c | 266 +++++++++++++++++++++++++++++++++++++ 3 files changed, 281 insertions(+) create mode 100644 sound/soc/intel/boards/skl_rt286.c diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 664df1f44530..6d67357f2d74 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -140,3 +140,16 @@ config SND_SOC_INTEL_SKYLAKE select SND_HDA_EXT_CORE select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST + +config SND_SOC_INTEL_SKL_RT286_MACH + tristate "ASoC Audio driver for SKL with RT286 I2S mode" + depends on X86 && ACPI + select SND_SOC_INTEL_SST + select SND_SOC_INTEL_SKYLAKE + select SND_SOC_RT286 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Skylake platforms + with RT286 I2S audio codec. + Say Y if you have such a device + If unsure select "N". diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index cb94895c9edb..371c4565cad8 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -6,6 +6,7 @@ snd-soc-sst-bytcr-rt5640-objs := bytcr_rt5640.o snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o snd-soc-sst-cht-bsw-rt5645-objs := cht_bsw_rt5645.o snd-soc-sst-cht-bsw-max98090_ti-objs := cht_bsw_max98090_ti.o +snd-soc-skl_rt286-objs := skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -15,3 +16,4 @@ obj-$(CONFIG_SND_SOC_INTEL_BYTCR_RT5640_MACH) += snd-soc-sst-bytcr-rt5640.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5645_MACH) += snd-soc-sst-cht-bsw-rt5645.o obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH) += snd-soc-sst-cht-bsw-max98090_ti.o +obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c new file mode 100644 index 000000000000..488c98ee1ded --- /dev/null +++ b/sound/soc/intel/boards/skl_rt286.c @@ -0,0 +1,266 @@ +/* + * Intel Skylake I2S Machine Driver + * + * Copyright (C) 2014-2015, Intel Corporation. All rights reserved. + * + * Modified from: + * Intel Broadwell Wildcatpoint SST Audio + * + * Copyright (C) 2013, Intel Corporation. All rights reserved. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License version + * 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include "../../codecs/rt286.h" + +static struct snd_soc_jack skylake_headset; +/* Headset jack detection DAPM pins */ +static struct snd_soc_jack_pin skylake_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static const struct snd_kcontrol_new skylake_controls[] = { + SOC_DAPM_PIN_SWITCH("Speaker"), + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget skylake_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Speaker", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("DMIC2", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), +}; + +static const struct snd_soc_dapm_route skylake_rt286_map[] = { + /* speaker */ + {"Speaker", NULL, "SPOR"}, + {"Speaker", NULL, "SPOL"}, + + /* HP jack connectors - unknown if we have jack deteck */ + {"Headphone Jack", NULL, "HPO Pin"}, + + /* other jacks */ + {"MIC1", NULL, "Mic Jack"}, + + /* digital mics */ + {"DMIC1 Pin", NULL, "DMIC2"}, + {"DMIC AIF", NULL, "SoC DMIC"}, + + /* CODEC BE connections */ + { "AIF1 Playback", NULL, "ssp0 Tx"}, + { "ssp0 Tx", NULL, "codec0_out"}, + { "ssp0 Tx", NULL, "codec1_out"}, + + { "codec0_in", NULL, "ssp0 Rx" }, + { "codec1_in", NULL, "ssp0 Rx" }, + { "ssp0 Rx", NULL, "AIF1 Capture" }, + + { "dmic01_hifi", NULL, "DMIC01 Rx" }, + { "DMIC01 Rx", NULL, "Capture" }, + + { "hif1", NULL, "iDisp Tx"}, + { "iDisp Tx", NULL, "iDisp_out"}, + +}; + +static int skylake_rt286_codec_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_codec *codec = rtd->codec; + int ret; + + ret = snd_soc_card_jack_new(rtd->card, "Headset", + SND_JACK_HEADSET | SND_JACK_BTN_0, + &skylake_headset, + skylake_headset_pins, ARRAY_SIZE(skylake_headset_pins)); + + if (ret) + return ret; + + rt286_mic_detect(codec, &skylake_headset); + + return 0; +} + + +static int skylake_ssp0_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); + + /* The output is 48KHz, stereo, 16bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + params_set_format(params, SNDRV_PCM_FORMAT_S16_LE); + + return 0; +} + +static int skylake_rt286_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; + + ret = snd_soc_dai_set_sysclk(codec_dai, RT286_SCLK_S_PLL, 24000000, + SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(rtd->dev, "set codec sysclk failed: %d\n", ret); + + return ret; +} + +static struct snd_soc_ops skylake_rt286_ops = { + .hw_params = skylake_rt286_hw_params, +}; + +/* skylake digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link skylake_rt286_dais[] = { + /* Front End DAI links */ + { + .name = "Skl Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_playback = 1, + }, + { + .name = "Skl Audio Capture Port", + .stream_name = "Audio Record", + .cpu_dai_name = "System Pin", + .platform_name = "0000:00:1f.3", + .nonatomic = 1, + .dynamic = 1, + .codec_name = "snd-soc-dummy", + .codec_dai_name = "snd-soc-dummy-dai", + .trigger = { + SND_SOC_DPCM_TRIGGER_POST, + SND_SOC_DPCM_TRIGGER_POST + }, + .dpcm_capture = 1, + }, + { + .name = "Skl 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 = "0000:00:1f.3", + .init = NULL, + .dpcm_capture = 1, + .ignore_suspend = 1, + .nonatomic = 1, + .dynamic = 1, + }, + + /* Back End DAI links */ + { + /* SSP0 - Codec */ + .name = "SSP0-Codec", + .be_id = 0, + .cpu_dai_name = "SSP0 Pin", + .platform_name = "0000:00:1f.3", + .no_pcm = 1, + .codec_name = "i2c-INT343A:00", + .codec_dai_name = "rt286-aif1", + .init = skylake_rt286_codec_init, + .dai_fmt = SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS, + .ignore_suspend = 1, + .ignore_pmdown_time = 1, + .be_hw_params_fixup = skylake_ssp0_fixup, + .ops = &skylake_rt286_ops, + .dpcm_playback = 1, + .dpcm_capture = 1, + }, + { + .name = "dmic01", + .be_id = 1, + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .ignore_suspend = 1, + .dpcm_capture = 1, + .no_pcm = 1, + }, +}; + +/* skylake audio machine driver for SPT + RT286S */ +static struct snd_soc_card skylake_rt286 = { + .name = "skylake-rt286", + .owner = THIS_MODULE, + .dai_link = skylake_rt286_dais, + .num_links = ARRAY_SIZE(skylake_rt286_dais), + .controls = skylake_controls, + .num_controls = ARRAY_SIZE(skylake_controls), + .dapm_widgets = skylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(skylake_widgets), + .dapm_routes = skylake_rt286_map, + .num_dapm_routes = ARRAY_SIZE(skylake_rt286_map), + .fully_routed = true, +}; + +static int skylake_audio_probe(struct platform_device *pdev) +{ + skylake_rt286.dev = &pdev->dev; + + return snd_soc_register_card(&skylake_rt286); +} + +static int skylake_audio_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&skylake_rt286); + return 0; +} + +static struct platform_driver skylake_audio = { + .probe = skylake_audio_probe, + .remove = skylake_audio_remove, + .driver = { + .name = "skl_alc286s_i2s", + }, +}; + +module_platform_driver(skylake_audio) + +/* Module information */ +MODULE_AUTHOR("Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com>"); +MODULE_DESCRIPTION("Intel SST Audio for Skylake"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:skl_alc286s_i2s"); -- 1.9.1

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[alsa-devel] [PATCH 0/7] ALSA: USB-audio: Support Zoom R16 playback
by Ricard Wanderlof 19 Oct '15

19 Oct '15

The Zoom R16 has a nonstandard playback format where each isochronous packet contains a length descriptor in the first four bytes. Curiously, capture data does not contain this and requires no quirk. The quirk involves adding the extra length descriptor whenever outgoing isochronous packets are generated, both in pcm.c (outgoing audio) and endpoint.c (silent data). The calculation for the max packet size in endpoint.c is also affected. This patch series refactors some code in pcm.c and endpoint.c before the actual quirk implementation. A commit then cleans up the entry in the quirks table. Tested with a Zoom R16, both separately using arecord and aplay (8 channel capture and 2 channel playback, respectively), as well as capture and playback together together using the Ardour digitial audio workstation application. The R24 reportedly is compatible with the R16, but I don't have access to one so I can't test it, and consequently the patch series only mentions the R16. Also tested using an Edirol UA-5 in both class compliant (16-bit) and "advanced" (24 bit, forces the use of quirks) modes in order to do some form of regression testing. Ricard Wanderlof (7): Break out copying to urb from prepare_playback_urb() Also move out hwptr_done wrap from prepare_playback_urb() Break out creation of silent urbs from prepare_outbound_urb() Add offset parameter to copy_to_urb() Add quirk for Zoom R16 playback Adjust max packet size calculation for tx_length_quirk Remove mixer entry from Zoom R16/24 quirk sound/usb/card.h | 1 + sound/usb/endpoint.c | 73 +++++++++++++++++++++++++++++++++-------------- sound/usb/pcm.c | 74 +++++++++++++++++++++++++++++++++++++----------- sound/usb/quirks-table.h | 14 +++------ sound/usb/quirks.c | 3 ++ sound/usb/stream.c | 1 + sound/usb/usbaudio.h | 1 + 7 files changed, 119 insertions(+), 48 deletions(-) -- 2.1.4

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