October 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 2/2] ARM: tegra: Enable the mic-detect gpio on Acer Chromebook 13
by Dylan Reid 02 Dec '14

02 Dec '14

Enables the gpio-base mic detection on the Acer Chromebook 13. This gpio is set by the jack-detection chip when it notices either of the TRRS type headsets with a microphone. Signed-off-by: Dylan Reid <dgreid(a)chromium.org> --- arch/arm/boot/dts/tegra124-nyan-big.dts | 2 ++ 1 file changed, 2 insertions(+) diff --git a/arch/arm/boot/dts/tegra124-nyan-big.dts b/arch/arm/boot/dts/tegra124-nyan-big.dts index 7d0784c..1c68ff9 100644 --- a/arch/arm/boot/dts/tegra124-nyan-big.dts +++ b/arch/arm/boot/dts/tegra124-nyan-big.dts @@ -1130,6 +1130,8 @@ clock-names = "pll_a", "pll_a_out0", "mclk"; nvidia,hp-det-gpios = <&gpio TEGRA_GPIO(I, 7) GPIO_ACTIVE_HIGH>; + nvidia,mic-det-gpios = + <&gpio TEGRA_GPIO(R, 7) GPIO_ACTIVE_HIGH>; }; }; -- 2.1.2.330.g565301e

3 2

[alsa-devel] [PATCH 0/2] ASoC: fsl: Use dynamic slot width for ESAI.
by Nicolin Chen 25 Nov '14

25 Nov '14

This series of patches add the support of dynamic slot width for ESAI driver. To do this, the PATCH-1 removes the fixed width. In order not to break those platforms that use fixed width, there comes the PATCH-2 to apply an override of slot_width. @Shengjiu, Will you be available to test this series on Sabre Auto for both Master and Slave cases? I'd like to wait for your Test-by. Thanks. Nicolin Chen (2): ASoC: fsl_esai: Use dynamic slot width as default ASoC: fsl-asoc-card: Add slot_width setting for cpu-dai sound/soc/fsl/fsl-asoc-card.c | 12 ++++++++++++ sound/soc/fsl/fsl_esai.c | 12 +++++++----- 2 files changed, 19 insertions(+), 5 deletions(-) -- 1.9.1

2 9

[alsa-devel] [PATCH 00/29 v2] ALSA: Enhancement for existed FireWire drivers
by Takashi Sakamoto 24 Nov '14

24 Nov '14

This patchset is to enable ALSA Dice driver to handle more models, to improve ALSA firewire-speakers driver as OXFW driver. This patchset is for linux-next branch and all of my work for Linux 3.19. == Features: Currently, ALSA Dice driver is limited for playback-only devices. The aim of this patchset is to add supports for: - Full-duplex streams with synchronization - Both PCM capture and playback - Both MIDI capture and playback Currently, ALSA firewire-speakers driver is only for two playback-only models. And this driver has a bug for one of them. The aim of this patchset is to fix the bug and to add supports for more models with: - Both PCM capture and playback - Both MIDI capture and playback The renew drivers doesn't support: - Synchronization between multiple devices on the same bus For this purpose, please join in user-space driver project (FFADO) - Any control functionality for internal DSP For this purpose, please use ffado-dbus-server/ffado-mixer, or write your own application == Updates from my previous post: [alsa-devel] [PATCH 00/14] ALSA: Dice enhancement http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/081890.ht… - Fix typo - Minor code change [alsa-devel] [RFC][PATCH 00/15 v4] OXFW driver, a succesor to firewire-speakers http://mailman.alsa-project.org/pipermail/alsa-devel/2014-May/076581.html - Support more Mackie (Loud) models by name detection - Fix a bug to handle FireWave - Use AV/C Stream Format Information command to set sampling rate == Remarks: - When the Dice driver is loaded, user-space driver fails to register notification address, thus cannot handle device for streaming. See: http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081731.… - OXFW chipset seems to transmit wrong values for timestamp, thus the driver don't apply full-duplex streams with synchronization == Supported Dice-based models (as long as I know): TC Electronic - Desktop Konnekt 6 - Impact Twin - Konnekt 8/24D/Live - Studio Konnekt 48 PreSonus - FireStudio 26x26/Mobile/Project/Tube/LightPipe - StudioLive 1602/1642/2442 - FireStation Mackie - Onyx Blackbird - Onyx i-series (latter models) Focusrite - Saffire Pro 14/24/24 DSP/26/40/56 M-Audio - Profire 610/2626 Alessis - IO 24/26 - MultiMix 12/16 Weiss - ADC2 - Vesta - Mineva - AFI 1 - TAG DAC 1 - INT 202 - DAC 202 - Flex 5000 DnR - Axus FireWire I/O card 16x16 Lexicon - I-ONIX FW 810S Allen and Heath - Zed R16 Midas - Venice F32 == Supported OXFW-based models (as long as I know): Griffin: - FireWave LaCie: - Firewire Speakers Behringer: - F-Control Audio 202 Mackie (Loud): - Onyx-i series (former models) - Onyx Satellite - Tapco Link.Firewire - d.2 pro/d.4 pro - U.420/U.420d Takashi Sakamoto (29): ALSA: dice: Rename structure and its members ALSA: dice: Move file to its own directory ALSA: dice: Split transaction functionality into a file ALSA: dice: Split stream functionality into a file ALSA: dice: Split PCM functionality into a file ALSA: dice: Split hwdep functionality into a file ALSA: dice: Split proc interface into a file ALSA: dice: Add new functions for constraints of PCM parameters ALSA: dice: Change the way to start stream ALSA: dice: Add support for duplex streams with synchronization ALSA: dice: Support for non SYT-Match sampling clock source mode ALSA: dice: Add support for capturing PCM samples ALSA: dice: Add support for MIDI capture/playback ALSA: dice: remove experimental state ALSA: speakers: Rename to oxfw and rename some members ALSA: oxfw: Move to its own directory ALSA: oxfw: Split stream functionality to a new file and add a header file ALSA: oxfw: Split PCM functionality to a new file ALSA: oxfw: Split control functionality to a new file ALSA: oxfw: Change the way to name card ALSA: oxfw: Add support for AV/C stream format command to get/set supported stream formation ALSA: oxfw: Change the way to make PCM rules/constraints ALSA: oxfw: Add proc interface for debugging purpose ALSA: oxfw: Change the way to start stream ALSA: oxfw: Add support for Behringer/Mackie devices ALSA: oxfw: Add support AMDTP in-stream ALSA: oxfw: add support for capturing PCM samples ALSA: oxfw: Add support for capture/playback MIDI messages ALSA: oxfw: Add hwdep interface include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 3 +- sound/firewire/Kconfig | 26 +- sound/firewire/Makefile | 6 +- sound/firewire/dice.c | 1511 ---------------------------- sound/firewire/dice/Makefile | 3 + sound/firewire/dice/dice-hwdep.c | 190 ++++ sound/firewire/{ => dice}/dice-interface.h | 0 sound/firewire/dice/dice-midi.c | 176 ++++ sound/firewire/dice/dice-pcm.c | 404 ++++++++ sound/firewire/dice/dice-proc.c | 252 +++++ sound/firewire/dice/dice-stream.c | 423 ++++++++ sound/firewire/dice/dice-transaction.c | 382 +++++++ sound/firewire/dice/dice.c | 359 +++++++ sound/firewire/dice/dice.h | 189 ++++ sound/firewire/oxfw/Makefile | 3 + sound/firewire/oxfw/oxfw-command.c | 153 +++ sound/firewire/oxfw/oxfw-control.c | 283 ++++++ sound/firewire/oxfw/oxfw-hwdep.c | 190 ++++ sound/firewire/oxfw/oxfw-midi.c | 189 ++++ sound/firewire/oxfw/oxfw-pcm.c | 399 ++++++++ sound/firewire/oxfw/oxfw-proc.c | 113 +++ sound/firewire/oxfw/oxfw-stream.c | 702 +++++++++++++ sound/firewire/oxfw/oxfw.c | 310 ++++++ sound/firewire/oxfw/oxfw.h | 145 +++ sound/firewire/speakers.c | 792 --------------- 26 files changed, 4887 insertions(+), 2319 deletions(-) delete mode 100644 sound/firewire/dice.c create mode 100644 sound/firewire/dice/Makefile create mode 100644 sound/firewire/dice/dice-hwdep.c rename sound/firewire/{ => dice}/dice-interface.h (100%) create mode 100644 sound/firewire/dice/dice-midi.c create mode 100644 sound/firewire/dice/dice-pcm.c create mode 100644 sound/firewire/dice/dice-proc.c create mode 100644 sound/firewire/dice/dice-stream.c create mode 100644 sound/firewire/dice/dice-transaction.c create mode 100644 sound/firewire/dice/dice.c create mode 100644 sound/firewire/dice/dice.h create mode 100644 sound/firewire/oxfw/Makefile create mode 100644 sound/firewire/oxfw/oxfw-command.c create mode 100644 sound/firewire/oxfw/oxfw-control.c create mode 100644 sound/firewire/oxfw/oxfw-hwdep.c create mode 100644 sound/firewire/oxfw/oxfw-midi.c create mode 100644 sound/firewire/oxfw/oxfw-pcm.c create mode 100644 sound/firewire/oxfw/oxfw-proc.c create mode 100644 sound/firewire/oxfw/oxfw-stream.c create mode 100644 sound/firewire/oxfw/oxfw.c create mode 100644 sound/firewire/oxfw/oxfw.h delete mode 100644 sound/firewire/speakers.c -- 1.9.1

4 42

[alsa-devel] [PATCH] ASoC: Intel: Add Cherrytrail & Braswell machine driver cht_bsw_rt5672
by mengdong.lin＠intel.com 21 Nov '14

21 Nov '14

From: Mengdong Lin <mengdong.lin(a)intel.com> Add machine driver for two Intel Cherryview-based platforms, Cherrytrail and Braswell, with RT5672 codec. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2a3af88..7479ce0 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -16,6 +16,9 @@ config SND_SST_MFLD_PLATFORM config SND_SST_IPC tristate +config SND_SST_MACHINE + tristate + config SND_SOC_INTEL_SST tristate "ASoC support for Intel(R) Smart Sound Technology" select SND_SOC_INTEL_SST_ACPI if ACPI @@ -76,3 +79,20 @@ config SND_SOC_INTEL_BROADWELL_MACH Ultrabook platforms. Say Y if you have such a device If unsure select "N". + +config SND_SOC_INTEL_CHT_BSW_RT5672_MACH + tristate "ASoC Audio driver for Intel Cherrytrail & Braswell with RT5672 codec" + depends on X86_INTEL_LPSS + select SND_SOC_RT5670 + select SND_SST_MFLD_PLATFORM + select SND_SOC_INTEL_SST + select SND_SST_IPC + select SND_SST_MACHINE + default n + + help + This adds support for ASoC machine driver for Intel(R) Cherrytrail & Braswell + platforms with RT5672 audio codec. + Say Y if you have such a device + If unsure select "N". + diff --git a/sound/soc/intel/Makefile b/sound/soc/intel/Makefile index 9ab43be..4069d3f 100644 --- a/sound/soc/intel/Makefile +++ b/sound/soc/intel/Makefile @@ -31,6 +31,7 @@ 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 obj-$(CONFIG_SND_SOC_INTEL_BYT_MAX98090_MACH) += snd-soc-sst-byt-max98090-mach.o obj-$(CONFIG_SND_SOC_INTEL_BROADWELL_MACH) += snd-soc-sst-broadwell.o +obj-$(CONFIG_SND_SST_MACHINE) += board/ # DSP driver obj-$(CONFIG_SND_SST_IPC) += sst/ diff --git a/sound/soc/intel/board/Makefile b/sound/soc/intel/board/Makefile new file mode 100644 index 0000000..9ecc227 --- /dev/null +++ b/sound/soc/intel/board/Makefile @@ -0,0 +1,2 @@ +snd-soc-sst-cht-bsw-rt5672-objs := cht_bsw_rt5672.o +obj-$(CONFIG_SND_SOC_INTEL_CHT_BSW_RT5672_MACH) += snd-soc-sst-cht-bsw-rt5672.o diff --git a/sound/soc/intel/board/cht_bsw_rt5672.c b/sound/soc/intel/board/cht_bsw_rt5672.c new file mode 100644 index 0000000..dffd8b1 --- /dev/null +++ b/sound/soc/intel/board/cht_bsw_rt5672.c @@ -0,0 +1,286 @@ +/* + * cht_bsw_rt5672.c - ASoc Machine driver for Intel Cherryview-based platforms + * Cherrytrail and Braswell, with RT5672 codec. + * + * Copyright (C) 2014 Intel Corp + * Author: Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> + * Mengdong Lin <mengdong.lin(a)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; version 2 of the License. + * + * 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 <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt5670.h" +#include "../sst-atom-controls.h" + +#define CHT_PLAT_CLK_3_HZ 19200000 + +static const struct snd_soc_dapm_widget cht_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static const struct snd_soc_dapm_route cht_audio_map[] = { + {"IN1P", NULL, "Headset Mic"}, + {"IN1N", NULL, "Headset Mic"}, + {"DMIC L1", NULL, "Int Mic"}, + {"DMIC R1", NULL, "Int Mic"}, + {"Headphone", NULL, "HPOL"}, + {"Headphone", NULL, "HPOR"}, + {"Ext Spk", NULL, "SPOLP"}, + {"Ext Spk", NULL, "SPOLN"}, + {"Ext Spk", NULL, "SPORP"}, + {"Ext Spk", NULL, "SPORN"}, + {"AIF1 Playback", NULL, "ssp2 Tx"}, + {"ssp2 Tx", NULL, "codec_out0"}, + {"ssp2 Tx", NULL, "codec_out1"}, + {"codec_in0", NULL, "ssp2 Rx"}, + {"codec_in1", NULL, "ssp2 Rx"}, + {"ssp2 Rx", NULL, "AIF1 Capture"}, +}; + +static const struct snd_kcontrol_new cht_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), + SOC_DAPM_PIN_SWITCH("Ext Spk"), +}; + +static int cht_aif1_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; + unsigned int fmt; + + if (strncmp(codec_dai->name, "rt5670-aif1", 11)) + return 0; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, + SNDRV_PCM_FORMAT_GSM); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + + /* TDM slave Mode */ + fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS; + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec DAI fmt %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5670_PLL1_S_MCLK, + CHT_PLAT_CLK_3_HZ, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5670_SCLK_S_PLL1, + params_rate(params) * 512, + SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + return 0; +} + +static int cht_codec_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 DSP will covert the FE rate to 48k, stereo, 24bits */ + rate->min = rate->max = 48000; + channels->min = channels->max = 2; + + /* set SSP2 to 24-bit */ + snd_mask_set(&params->masks[SNDRV_PCM_HW_PARAM_FORMAT - + SNDRV_PCM_HW_PARAM_FIRST_MASK], + SNDRV_PCM_FORMAT_S24_LE); + return 0; +} + +static int cht_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + case SND_SOC_BIAS_OFF: + break; + default: + dev_err(card->dev, "Invalid bias level=%d\n", level); + return -EINVAL; + } + card->dapm.bias_level = level; + return 0; +} + +static int cht_init(struct snd_soc_pcm_runtime *runtime) +{ + int ret; + struct snd_soc_card *card = runtime->card; + + /* Set card bias level */ + cht_set_bias_level(card, &card->dapm, SND_SOC_BIAS_OFF); + card->dapm.idle_bias_off = true; + + ret = snd_soc_dapm_sync(&card->dapm); + if (ret) { + dev_err(card->dev, "unable to sync dapm\n"); + return ret; + } + return ret; +} + +static unsigned int rates_48000[] = { + 48000, +}; + +static struct snd_pcm_hw_constraint_list constraints_48000 = { + .count = ARRAY_SIZE(rates_48000), + .list = rates_48000, +}; + +static int cht_aif1_startup(struct snd_pcm_substream *substream) +{ + return snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, + &constraints_48000); +} + +static struct snd_soc_ops cht_aif1_ops = { + .startup = cht_aif1_startup, +}; + +static struct snd_soc_ops cht_be_ssp2_ops = { + .hw_params = cht_aif1_hw_params, +}; + +static struct snd_soc_dai_link cht_dailink[] = { + /* Front End DAI links */ + [MERR_DPCM_AUDIO] = { + .name = "Audio Port", + .stream_name = "Audio", + .cpu_dai_name = "media-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + .init = cht_init, + .ignore_suspend = 1, + .dynamic = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_aif1_ops, + }, + [MERR_DPCM_COMPR] = { + .name = "Compressed Port", + .stream_name = "Compress", + .cpu_dai_name = "compress-cpu-dai", + .codec_dai_name = "snd-soc-dummy-dai", + .codec_name = "snd-soc-dummy", + .platform_name = "sst-mfld-platform", + }, + + /* Back End DAI links */ + { + /* SSP2 - Codec */ + .name = "SSP2-Codec", + .be_id = 1, + .cpu_dai_name = "ssp2-port", + .platform_name = "sst-mfld-platform", + .no_pcm = 1, + .codec_dai_name = "rt5670-aif1", + .codec_name = "i2c-10EC5670:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBS_CFS, + .be_hw_params_fixup = cht_codec_fixup, + .ignore_suspend = 1, + .dpcm_playback = 1, + .dpcm_capture = 1, + .ops = &cht_be_ssp2_ops, + }, +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cht = { + .name = "cherrytrailcraudio", + .dai_link = cht_dailink, + .num_links = ARRAY_SIZE(cht_dailink), + .set_bias_level = cht_set_bias_level, + .dapm_widgets = cht_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets), + .dapm_routes = cht_audio_map, + .num_dapm_routes = ARRAY_SIZE(cht_audio_map), + .controls = cht_mc_controls, + .num_controls = ARRAY_SIZE(cht_mc_controls), +}; + +static int snd_cht_mc_probe(struct platform_device *pdev) +{ + int ret_val = 0; + + /* register the soc card */ + snd_soc_card_cht.dev = &pdev->dev; + ret_val = snd_soc_register_card(&snd_soc_card_cht); + if (ret_val) { + dev_err(&pdev->dev, + "snd_soc_register_card failed %d\n", ret_val); + return ret_val; + } + platform_set_drvdata(pdev, &snd_soc_card_cht); + return ret_val; +} + +static int snd_cht_mc_remove(struct platform_device *pdev) +{ + struct snd_soc_card *soc_card = platform_get_drvdata(pdev); + + snd_soc_card_set_drvdata(soc_card, NULL); + snd_soc_unregister_card(soc_card); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_cht_mc_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "cht-bsw-rt5672", + .pm = &snd_soc_pm_ops, + }, + .probe = snd_cht_mc_probe, + .remove = snd_cht_mc_remove, +}; + +module_platform_driver(snd_cht_mc_driver); + +MODULE_DESCRIPTION("ASoC Intel(R) Baytrail CR Machine driver"); +MODULE_AUTHOR("Subhransu S. Prusty, Mengdong Lin"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cht-bsw-rt5672"); -- 1.9.1

4 24

[alsa-devel] [PATCH] pcm: add new 32-bit DSD sample format
by Jurgen Kramer 21 Nov '14

21 Nov '14

Add the new DSD_U32_LE sample format to alsa-lib. NB include/pcm.h and include/sound/asound.h are updated so a new sync with the kernel headers is not needed Signed-off-by: Jurgen Kramer <gtmkramer(a)xs4all.nl> --- include/pcm.h | 4 +++- include/sound/asound.h | 3 ++- src/pcm/pcm.c | 2 ++ src/pcm/pcm_misc.c | 6 ++++++ 4 files changed, 13 insertions(+), 2 deletions(-) diff --git a/include/pcm.h b/include/pcm.h index 11e9f0d..db88ad5 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -211,7 +211,9 @@ typedef enum _snd_pcm_format { SND_PCM_FORMAT_DSD_U8, /* Direct Stream Digital (DSD) in 2-byte samples (x16) */ SND_PCM_FORMAT_DSD_U16_LE, - SND_PCM_FORMAT_LAST = SND_PCM_FORMAT_DSD_U16_LE, + /* Direct Stream Digital (DSD) in 4-byte samples (x32) */ + SND_PCM_FORMAT_DSD_U32_LE, + SND_PCM_FORMAT_LAST = SND_PCM_FORMAT_DSD_U32_LE, #if __BYTE_ORDER == __LITTLE_ENDIAN /** Signed 16 bit CPU endian */ diff --git a/include/sound/asound.h b/include/sound/asound.h index 32168f7..6ee5867 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -219,7 +219,8 @@ typedef int __bitwise snd_pcm_format_t; #define SNDRV_PCM_FORMAT_G723_40_1B ((__force snd_pcm_format_t) 47) /* 1 sample in 1 byte */ #define SNDRV_PCM_FORMAT_DSD_U8 ((__force snd_pcm_format_t) 48) /* DSD, 1-byte samples DSD (x8) */ #define SNDRV_PCM_FORMAT_DSD_U16_LE ((__force snd_pcm_format_t) 49) /* DSD, 2-byte samples DSD (x16), little endian */ -#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U16_LE +#define SNDRV_PCM_FORMAT_DSD_U32_LE ((__force snd_pcm_format_t) 50) /* DSD, 4-byte samples DSD (x32), little endian */ +#define SNDRV_PCM_FORMAT_LAST SNDRV_PCM_FORMAT_DSD_U32_LE #ifdef SNDRV_LITTLE_ENDIAN #define SNDRV_PCM_FORMAT_S16 SNDRV_PCM_FORMAT_S16_LE diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 1399a5b..2e24338 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -1565,6 +1565,7 @@ static const char *const snd_pcm_format_names[] = { FORMAT(G723_40_1B), FORMAT(DSD_U8), FORMAT(DSD_U16_LE), + FORMAT(DSD_U32_LE), }; static const char *const snd_pcm_format_aliases[SND_PCM_FORMAT_LAST+1] = { @@ -1624,6 +1625,7 @@ static const char *const snd_pcm_format_descriptions[] = { FORMATD(G723_40_1B, "G.723 (ADPCM) 40 kbit/s, 1 sample in 1 byte"), FORMATD(DSD_U8, "Direct Stream Digital, 1-byte (x8), oldest bit in MSB"), FORMATD(DSD_U16_LE, "Direct Stream Digital, 2-byte (x16), little endian, oldest bits in MSB"), + FORMATD(DSD_U32_LE, "Direct Stream Digital, 4-byte (x32), little endian, oldest bits in MSB"), }; static const char *const snd_pcm_type_names[] = { diff --git a/src/pcm/pcm_misc.c b/src/pcm/pcm_misc.c index 46fc771..9272179 100644 --- a/src/pcm/pcm_misc.c +++ b/src/pcm/pcm_misc.c @@ -64,6 +64,7 @@ int snd_pcm_format_signed(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3BE: case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0; default: return -EINVAL; @@ -154,6 +155,7 @@ int snd_pcm_format_little_endian(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U18_3BE: case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0; default: return -EINVAL; @@ -232,6 +234,7 @@ int snd_pcm_format_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_U32_BE: case SNDRV_PCM_FORMAT_FLOAT_LE: case SNDRV_PCM_FORMAT_FLOAT_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 32; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -292,6 +295,7 @@ int snd_pcm_format_physical_width(snd_pcm_format_t format) case SNDRV_PCM_FORMAT_FLOAT_BE: case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_LE: case SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 32; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -348,6 +352,7 @@ ssize_t snd_pcm_format_size(snd_pcm_format_t format, size_t samples) case SNDRV_PCM_FORMAT_U32_BE: case SNDRV_PCM_FORMAT_FLOAT_LE: case SNDRV_PCM_FORMAT_FLOAT_BE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return samples * 4; case SNDRV_PCM_FORMAT_FLOAT64_LE: case SNDRV_PCM_FORMAT_FLOAT64_BE: @@ -394,6 +399,7 @@ u_int64_t snd_pcm_format_silence_64(snd_pcm_format_t format) return 0x8080808080808080ULL; case SNDRV_PCM_FORMAT_DSD_U8: case SNDRV_PCM_FORMAT_DSD_U16_LE: + case SNDRV_PCM_FORMAT_DSD_U32_LE: return 0x6969696969696969ULL; #ifdef SNDRV_LITTLE_ENDIAN case SNDRV_PCM_FORMAT_U16_LE: -- 1.9.3

3 7

[alsa-devel] AKAI EIE PRO
by Martin Svensson 20 Nov '14

20 Nov '14

Dont know if its ok to ask a question here so please let me know if its not ok then i wont ask again ;) i just bought an external usb AKAI EIE PRO sound card, is there any future plans to implement it in Alsa? Sincerally Martin aka "ztealmax"

4 7

[alsa-devel] [PATCH v2 0/6] media token resource framework
by Shuah Khan 18 Nov '14

18 Nov '14

Add media token device resource framework to allow sharing resources such as tuner, dma, audio etc. across media drivers and non-media sound drivers that control media hardware. The Media token resource is created at the main struct device that is common to all drivers that claim various pieces of the main media device, which allows them to find the resource using the main struct device. As an example, digital, analog, and snd-usb-audio drivers can use the media token resource API using the main struct device for the interface the media device is attached to. This patch series consists of media token resource framework and changes to use it in dvb-core, v4l2-core, au0828 driver, and snd-usb-audio driver. With these changes dvb and v4l2 can share the tuner without disrupting each other. Used tvtime, xawtv, kaffeine, and vlc, vlc audio capture option, arecord/aplay during development to identify v4l2 vb2 and vb1 ioctls and file operations that disrupt the digital stream and would require changes to check tuner ownership prior to changing the tuner configuration. vb2 changes are made in the v4l2-core and vb1 changes are made in the au0828 driver to encourage porting drivers to vb2 to advantage of the new media token resource framework with changes in the core. In this patch v2 series, fixed problems identified in the patch v1 series. Important ones are changing snd-usb-audio to use media tokens, holding tuner lock in VIDIOC_ENUMINPUT, and VIDIOC_QUERYSTD. Shuah Khan (6): media: add media token device resource framework media: v4l2-core changes to use media token api media: au0828-video changes to use media token api media: dvb-core changes to use media token api sound/usb: pcm changes to use media token api media: au0828-core changes to create and destroy media MAINTAINERS | 2 + drivers/media/dvb-core/dvb_frontend.c | 14 +- drivers/media/usb/au0828/au0828-core.c | 23 +++ drivers/media/usb/au0828/au0828-video.c | 42 +++++- drivers/media/v4l2-core/v4l2-fh.c | 7 + drivers/media/v4l2-core/v4l2-ioctl.c | 61 ++++++++ include/linux/media_tknres.h | 50 +++++++ lib/Makefile | 2 + lib/media_tknres.c | 237 +++++++++++++++++++++++++++++++ sound/usb/pcm.c | 9 ++ 10 files changed, 445 insertions(+), 2 deletions(-) create mode 100644 include/linux/media_tknres.h create mode 100644 lib/media_tknres.c -- 1.7.10.4

11 56

[alsa-devel] [PATCH 00/15] new locomo driver
by Dmitry Eremin-Solenikov 14 Nov '14

14 Nov '14

Sharp Zaurus SL-5500 and SL-5600 use special companion Gate Array. Current drivers present in Linux kernel has some problems: * It uses custom bus instead of platform bus/mfd core. * Device drivers are not well layered/separated. * It uses custom gpio accessors instead of using GPIOLIB. With this patchset I tried to modernise and restructure the LoCoMo driver. Also I added an experimental SPI driver that will be used by mmc-spi on Sharp SL-5500 collie devices. ---------------------------------------------------------------- Dmitry Eremin-Solenikov (15): mfd: add new driver for Sharp LoCoMo GPIO: port LoCoMo gpio support from old driver leds: port locomo leds driver to new locomo core input: convert LoCoMo keyboard driver to use new locomo core video: backlight: add new locomo backlight driver video: lcd: add LoCoMo LCD driver video: backlight: drop old locomo bl/lcd driver ARM: sa1100: make collie use new locomo drivers ARM: sa1100: don't preallocate IRQ space for locomo ARM: pxa: poodle: use new LoCoMo driver sound: soc: poodle: make use of new locomo GPIO interface ARM: pxa: poodle: don't preallocate IRQ space for locomo ARM: drop old LoCoMo driver gpio: locomo: implement per-pin irq handling spi: add locomo SPI driver arch/arm/common/Kconfig | 3 - arch/arm/common/Makefile | 1 - arch/arm/common/locomo.c | 914 --------------------- arch/arm/mach-pxa/Kconfig | 1 - arch/arm/mach-pxa/include/mach/poodle.h | 16 +- arch/arm/mach-pxa/poodle.c | 23 +- arch/arm/mach-sa1100/Kconfig | 1 - arch/arm/mach-sa1100/collie.c | 112 +-- arch/arm/mach-sa1100/include/mach/collie.h | 25 +- arch/arm/mach-sa1100/include/mach/irqs.h | 19 +- drivers/gpio/Kconfig | 7 + drivers/gpio/Makefile | 1 + drivers/gpio/gpio-locomo.c | 386 +++++++++ drivers/input/keyboard/Kconfig | 1 - drivers/input/keyboard/locomokbd.c | 165 ++-- drivers/leds/Kconfig | 1 - drivers/leds/leds-locomo.c | 104 ++- drivers/mfd/Kconfig | 8 + drivers/mfd/Makefile | 1 + drivers/mfd/locomo.c | 644 +++++++++++++++ drivers/spi/Kconfig | 8 + drivers/spi/Makefile | 1 + drivers/spi/spi-locomo.c | 370 +++++++++ drivers/video/backlight/Kconfig | 14 +- drivers/video/backlight/Makefile | 3 +- drivers/video/backlight/locomo_bl.c | 171 ++++ drivers/video/backlight/locomo_lcd.c | 224 +++++ drivers/video/backlight/locomolcd.c | 255 ------ .../asm/hardware => include/linux/mfd}/locomo.h | 148 ++-- sound/soc/pxa/poodle.c | 51 +- 30 files changed, 2185 insertions(+), 1493 deletions(-) delete mode 100644 arch/arm/common/locomo.c create mode 100644 drivers/gpio/gpio-locomo.c create mode 100644 drivers/mfd/locomo.c create mode 100644 drivers/spi/spi-locomo.c create mode 100644 drivers/video/backlight/locomo_bl.c create mode 100644 drivers/video/backlight/locomo_lcd.c delete mode 100644 drivers/video/backlight/locomolcd.c rename {arch/arm/include/asm/hardware => include/linux/mfd}/locomo.h (55%)

9 53

[alsa-devel] [PATCH v2] Scarlett mixer interface inclusion
by Chris J Arges 13 Nov '14

13 Nov '14

This is a followup to David Henningsson's PATCH [1] to include the work of Tobias Hoffman and Robin Gareus in getting the Scarlett mixer interface working. I've read through the comments and tried to include fixes where appropriate. Overall, I'd like to get a good base commit and start working on other commits that address any major structual changes such as making functions more generic or any other cleanup. I've tested this on my Scarlett 18i8 mixer and it does work, however there are still a few issues which need to be fixed: 1) When loading a device one gets the following messages: snd-usb-audio 1-1.1:1.0: control 2:0:0:Master Playback Switch:0 is already present snd-usb-audio: probe of 1-1.1:1.3 failed with error -5 2) When unloading a device there are numerous sysfs_remove_group issues: usb 1-1.1: USB disconnect, device number 6 ------------[ cut here ]------------ WARNING: CPU: 0 PID: 54 at /build/buildd/linux-3.16.0/fs/sysfs/group.c:219 sysfs_remove_group+0x99/0xa0() sysfs group ffffffff82cbd6e0 not found for kobject 'midi1' Modules linked in: snd_usb_audio(OE) snd_usbmidi_lib(OE) ctr ccm xt_conntrack ipt_REJECT xt_CHECKSUM iptable_mangle ipt_MASQUERADE iptable_nat nf_conntrack_ipv4 nf_defrag_ipv4 nf_nat_ipv4 nf_nat nf_conntrack xt_tcpudp bridge stp llc iptable_filter ip_tables x_tables arc4 iwldvm snd_hda_codec_hdmi mac80211 snd_hda_codec_conexant snd_hda_codec_generic iwlwifi snd_hda_intel uvcvideo snd_hda_controller videobuf2_vmalloc snd_hda_codec videobuf2_memops videobuf2_core snd_hwdep v4l2_common snd_pcm intel_rapl x86_pkg_temp_thermal videodev intel_powerclamp media coretemp kvm_intel bnep kvm btusb rfcomm cfg80211 joydev bluetooth serio_raw 6lowpan_iphc thinkpad_acpi nvram snd_seq_midi lpc_ich snd_seq_midi_event snd_rawmidi shpchp snd_seq mei_me snd_seq_device snd_timer mei snd soundcore parport_pc ppdev mac_hid lp parport binfmt_misc nls_iso8859_1 dm_crypt crct10dif_pclmul crc32_pclmul ghash_clmulni_intel aesni_intel aes_x86_64 lrw gf128mul glue_helper ablk_helper cryptd i915 psmouse ahci libahci firewire_ohci sdhci_pci firewire_core sdhci crc_itu_t i2c_algo_bit e1000e drm_kms_helper drm ptp pps_core wmi video CPU: 0 PID: 54 Comm: khubd Tainted: G OE 3.16.0-23-generic #30-Ubuntu Hardware name: LENOVO 4177CTO/4177CTO, BIOS 83ET76WW (1.46 ) 07/05/2013 0000000000000009 ffff880407b4f920 ffffffff8277fcbc ffff880407b4f968 ffff880407b4f958 ffffffff8206fd8d 0000000000000000 ffffffff82cbd6e0 ffff880405c7b410 0000000000000001 ffff8803eb059cf8 ffff880407b4f9b8 Call Trace: [<ffffffff8277fcbc>] dump_stack+0x45/0x56 [<ffffffff8206fd8d>] warn_slowpath_common+0x7d/0xa0 [<ffffffff8206fdfc>] warn_slowpath_fmt+0x4c/0x50 [<ffffffff82256538>] ? kernfs_find_and_get_ns+0x48/0x60 [<ffffffff8225a259>] sysfs_remove_group+0x99/0xa0 [<ffffffff824d6927>] dpm_sysfs_remove+0x57/0x60 [<ffffffff824cb625>] device_del+0x45/0x1d0 [<ffffffff824cb7ce>] device_unregister+0x1e/0x70 [<ffffffff824cb89c>] device_destroy+0x3c/0x50 [<ffffffffc045f34b>] sound_remove_unit+0x5b/0x90 [soundcore] [<ffffffffc045f3aa>] unregister_sound_special+0x2a/0x30 [soundcore] [<ffffffffc04cc9b0>] snd_unregister_oss_device+0x60/0x100 [snd] [<ffffffffc0550ca9>] snd_rawmidi_dev_disconnect+0x119/0x160 [snd_rawmidi] [<ffffffffc04cc14c>] __snd_device_disconnect.part.1+0x1c/0x60 [snd] [<ffffffffc04cc437>] snd_device_disconnect_all+0x47/0x60 [snd] [<ffffffffc04c59f7>] snd_card_disconnect+0x147/0x1f0 [snd] [<ffffffffc09cf5a8>] usb_audio_disconnect+0x78/0x1a0 [snd_usb_audio] [<ffffffff825867b4>] usb_unbind_interface+0x74/0x2b0 [<ffffffff824cf76f>] __device_release_driver+0x7f/0xf0 [<ffffffff824cf803>] device_release_driver+0x23/0x30 [<ffffffff824cf0f8>] bus_remove_device+0x108/0x180 [<ffffffff824cb709>] device_del+0x129/0x1d0 [<ffffffff82583f21>] usb_disable_device+0x91/0x290 [<ffffffff82578688>] usb_disconnect+0x98/0x2f0 [<ffffffff8257a6cf>] hub_port_connect+0x7f/0xac0 [<ffffffff8257b9a3>] hub_events+0x893/0xda0 [<ffffffff8257beed>] hub_thread+0x3d/0x160 [<ffffffff820b9590>] ? prepare_to_wait_event+0x100/0x100 [<ffffffff8257beb0>] ? hub_events+0xda0/0xda0 [<ffffffff82094aeb>] kthread+0xdb/0x100 [<ffffffff82094a10>] ? kthread_create_on_node+0x1c0/0x1c0 [<ffffffff82787c3c>] ret_from_fork+0x7c/0xb0 [<ffffffff82094a10>] ? kthread_create_on_node+0x1c0/0x1c0 ---[ end trace 7a8cdc4a9c65ddf5 ]--- However this issue occured even without the patch applied. Please follow up with additional comments and review and I'll be happy to incorporate them into the next patch version. [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2014-October/081986.ht… Chris J Arges (1): Scarlett mixer interface for 6i6, 18i6, 18i8 and 18i20 sound/usb/Makefile | 1 + sound/usb/mixer.c | 27 +- sound/usb/quirks-table.h | 51 -- sound/usb/scarlett/scarlettmixer.c | 1264 ++++++++++++++++++++++++++++++++++++ sound/usb/scarlett/scarlettmixer.h | 6 + 5 files changed, 1293 insertions(+), 56 deletions(-) create mode 100644 sound/usb/scarlett/scarlettmixer.c create mode 100644 sound/usb/scarlett/scarlettmixer.h -- 2.1.0

6 59

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add ASRC support
by bardliao＠realtek.com 11 Nov '14

11 Nov '14

From: Bard Liao <bardliao(a)realtek.com> This patch add ASRC support for rt5645 codec. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 144 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 144 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index bc442f4..3131e91 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -443,6 +443,65 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); +static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + unsigned int u_bit = 0, p_bit = 0; + struct soc_enum *em = + (struct soc_enum *)kcontrol->private_value; + + switch (em->reg) { + case RT5645_ASRC_2: + switch (em->shift_l) { + case 0: + u_bit = 0x8; + p_bit = RT5645_PWR_ADC_S1F; + break; + case 4: + u_bit = 0x100; + p_bit = RT5645_PWR_DAC_MF_R; + break; + case 8: + u_bit = 0x200; + p_bit = RT5645_PWR_DAC_MF_L; + break; + case 12: + u_bit = 0x400; + p_bit = RT5645_PWR_DAC_S1F; + break; + } + break; + case RT5645_ASRC_3: + switch (em->shift_l) { + case 0: + u_bit = 0x1; + p_bit = RT5645_PWR_ADC_MF_R; + break; + case 4: + u_bit = 0x2; + p_bit = RT5645_PWR_ADC_MF_L; + break; + } + break; + } + + if (u_bit || p_bit) { + switch (ucontrol->value.integer.value[0]) { + case 1 ... 4: /*enable*/ + if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) + snd_soc_update_bits(codec, + RT5645_ASRC_1, u_bit, u_bit); + break; + default: /*disable*/ + snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); + break; + } + } + + return snd_soc_put_enum_double(kcontrol, ucontrol); +} + static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, @@ -554,6 +613,53 @@ static int is_sys_clk_from_pll(struct snd_soc_dapm_widget *source, return 0; } +static int is_using_asrc(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + + switch (source->shift) { + case 0: + reg = RT5645_ASRC_3; + shift = 0; + break; + case 1: + reg = RT5645_ASRC_3; + shift = 4; + break; + case 3: + reg = RT5645_ASRC_2; + shift = 0; + break; + case 8: + reg = RT5645_ASRC_2; + shift = 4; + break; + case 9: + reg = RT5645_ASRC_2; + shift = 8; + break; + case 10: + reg = RT5645_ASRC_2; + shift = 12; + break; + default: + return 0; + } + + val = (snd_soc_read(source->codec, reg) >> shift) & 0xf; + switch (val) { + case 1: + case 2: + case 3: + case 4: + return 1; + default: + return 0; + } + +} + /* Digital Mixer */ static const struct snd_kcontrol_new rt5645_sto1_adc_l_mix[] = { SOC_DAPM_SINGLE("ADC1 Switch", RT5645_STO1_ADC_MIXER, @@ -1246,6 +1352,30 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("Mic Det Power", RT5645_PWR_VOL, RT5645_PWR_MIC_DET_BIT, 0, NULL, 0), + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("I2S1 ASRC", 1, RT5645_ASRC_1, + 11, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("I2S2 ASRC", 1, RT5645_ASRC_1, + 12, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC STO ASRC", 1, RT5645_ASRC_1, + 10, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO L ASRC", 1, RT5645_ASRC_1, + 9, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DAC MONO R ASRC", 1, RT5645_ASRC_1, + 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC STO1 ASRC", 1, RT5645_ASRC_1, + 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO L ASRC", 1, RT5645_ASRC_1, + 5, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC MONO R ASRC", 1, RT5645_ASRC_1, + 4, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5645_ASRC_1, + 3, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO L ASRC", 1, RT5645_ASRC_1, + 1, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC MONO R ASRC", 1, RT5645_ASRC_1, + 0, 0, NULL, 0), + /* Input Side */ /* micbias */ SND_SOC_DAPM_MICBIAS("micbias1", RT5645_PWR_ANLG2, @@ -1504,6 +1634,17 @@ static const struct snd_soc_dapm_widget rt5645_dapm_widgets[] = { }; static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { + { "adc stereo1 filter", NULL, "ADC STO1 ASRC", is_using_asrc }, + { "adc stereo2 filter", NULL, "ADC STO2 ASRC", is_using_asrc }, + { "adc mono left filter", NULL, "ADC MONO L ASRC", is_using_asrc }, + { "adc mono right filter", NULL, "ADC MONO R ASRC", is_using_asrc }, + { "dac mono left filter", NULL, "DAC MONO L ASRC", is_using_asrc }, + { "dac mono right filter", NULL, "DAC MONO R ASRC", is_using_asrc }, + { "dac stereo1 filter", NULL, "DAC STO ASRC", is_using_asrc }, + + { "I2S1", NULL, "I2S1 ASRC" }, + { "I2S2", NULL, "I2S2 ASRC" }, + { "IN1P", NULL, "LDO2" }, { "IN2P", NULL, "LDO2" }, @@ -1550,12 +1691,15 @@ static const struct snd_soc_dapm_route rt5645_dapm_routes[] = { { "Stereo1 DMIC Mux", "DMIC1", "DMIC1" }, { "Stereo1 DMIC Mux", "DMIC2", "DMIC2" }, + { "Stereo1 DMIC Mux", NULL, "DMIC STO1 ASRC" }, { "Mono DMIC L Mux", "DMIC1", "DMIC L1" }, { "Mono DMIC L Mux", "DMIC2", "DMIC L2" }, + { "Mono DMIC L Mux", NULL, "DMIC MONO L ASRC" }, { "Mono DMIC R Mux", "DMIC1", "DMIC R1" }, { "Mono DMIC R Mux", "DMIC2", "DMIC R2" }, + { "Mono DMIC R Mux", NULL, "DMIC MONO R ASRC" }, { "Stereo1 ADC L2 Mux", "DMIC", "Stereo1 DMIC Mux" }, { "Stereo1 ADC L2 Mux", "DAC MIX", "DAC MIXL" }, -- 1.8.1.1.439.g50a6b54

4 5