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

[alsa-devel] [PATCH 00/12] Rework OMAP4+ HDMI audio support
by Jyri Sarha 15 Jul '14

15 Jul '14

The patches are based on linux 3.16-rc2 and OMAPDSS HDMI infoframe patches[1] from Tomi Valkeinen. The base, the patches, and couple of additional not-to-be-merged omap2plus_defconfig patches can be found here: git://git.ti.com/~jyrisarha/ti-linux-kernel/jyrisarhas-audio-video-linux-fe… omap-hdmi-audio The patch set fixes OMAP4+ HDMI audio. The structure of the implementation looks a bit different than before. Instead of creating a driver specific API for a separate ASoC component driver to connect to, this implementation integrates an the ASoC cpu-dai component driver into the HDMI driver. Also the other ASoC component drivers needed for operational audio are automatically registered by the HDMI driver. There is no need to add anything to the device tree as long as the nodes needed for HDMI video are there. Big part of the HDMI audio code is still unchanged and there is a need for a cleanup there. Also there is still probably something wrong with speaker mapping of multi-channel streams. I will get back to cleaning up these issues later. Best regards, Jyri [1] http://marc.info/?l=linux-fbdev&m=140360425130951&w=2 Jyri Sarha (12): ARM: OMAP2+: Remove non working OMAP HDMI audio initialization OMAPDSS: hdmi.h: Add HDMI_AUDIO_LAYOUT_6CH enum value OMAPDSS: hdmi_wp: Add function for getting hdmi_wp physical base address OMAPDSS: hdmi_audio: Integrated ASoC DAI component driver implementation OMAPDSS: Kconfig: Select audio dependencies if OMAP4_DSS_HDMI_AUDIO=y OMAPDSS: hdmi4: Register HDMI audio ASoC drivers from HDMI driver OMAPDSS: Kconfig: Select audio dependencies if OMAP5_DSS_HDMI_AUDIO=y OMAPDSS: hdmi5: Register HDMI audio ASoC drivers from HDMI driver ASoC: omap: Remove obsolete HDMI audio code and Kconfig options OMAPDSS: hdmi4: Remove callbacks for an external ASoC DAI driver OMAPDSS: hdmi5: Remove callbacks for an external ASoC DAI driver OMAPDSS: Remove all references to obsolete HDMI audio callbacks arch/arm/mach-omap2/devices.c | 28 -- .../fbdev/omap2/displays-new/connector-hdmi.c | 99 ------ .../fbdev/omap2/displays-new/encoder-tpd12s015.c | 56 --- drivers/video/fbdev/omap2/dss/Kconfig | 24 +- drivers/video/fbdev/omap2/dss/Makefile | 2 + drivers/video/fbdev/omap2/dss/hdmi.h | 43 ++- drivers/video/fbdev/omap2/dss/hdmi4.c | 158 +++------ drivers/video/fbdev/omap2/dss/hdmi5.c | 170 +++------ drivers/video/fbdev/omap2/dss/hdmi_audio.c | 317 +++++++++++++++++ drivers/video/fbdev/omap2/dss/hdmi_wp.c | 6 + include/video/omapdss.h | 31 -- sound/soc/omap/Kconfig | 13 - sound/soc/omap/Makefile | 4 - sound/soc/omap/omap-hdmi-card.c | 87 ----- sound/soc/omap/omap-hdmi.c | 364 -------------------- sound/soc/omap/omap-hdmi.h | 38 -- 16 files changed, 489 insertions(+), 951 deletions(-) create mode 100644 drivers/video/fbdev/omap2/dss/hdmi_audio.c delete mode 100644 sound/soc/omap/omap-hdmi-card.c delete mode 100644 sound/soc/omap/omap-hdmi.c delete mode 100644 sound/soc/omap/omap-hdmi.h -- 1.7.9.5

5 29

[alsa-devel] [RFC v2] ASoC: core: add a helper for extended byte controls using TLV
by Vinod Koul 15 Jul '14

15 Jul '14

From: Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com> ALSA supports arbitrary length TLVs for each kcontrol that can be used to pass metadata about the control (e.g. volumes, enum information). The same transport mechanism is now used for arbitrary length data by defining a new helper. Signed-off-by: Omair Mohammed Abdullah <omair.m.abdullah(a)intel.com> Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> --- As discussed in [1] we are adding a new approach to solve the byte control extensions by using existing TLVs and combining them with byte controls. The usermode on seeing byte control + TLV can treat it differently as it does for control + TLV combination today. This way we don't change kernel API and existing users will be happy, while providing embedded folks facility to pass large bytes data to kcontrols [1]: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069483.h… include/sound/soc.h | 14 +++++++++++++- sound/soc/soc-core.c | 22 ++++++++++++++++++++++ 2 files changed, 35 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ed9e2d7..7522221 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -270,7 +270,14 @@ .get = xhandler_get, .put = xhandler_put, \ .private_value = (unsigned long)&(struct soc_bytes_ext) \ {.max = xcount} } - +#define SND_SOC_BYTES_TLV(xname, xcount, xhandler_get, xhandler_put) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .access = SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE | \ + SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK, \ + .tlv.c = (snd_soc_bytes_tlv_callback), \ + .info = snd_soc_info_bytes_ext, \ + .private_value = (unsigned long)&(struct soc_bytes_ext) \ + {.max = xcount, .get = xhandler_get, .put = xhandler_put, }} #define SOC_SINGLE_XR_SX(xname, xregbase, xregcount, xnbits, \ xmin, xmax, xinvert) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ @@ -552,6 +559,8 @@ int snd_soc_bytes_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol); int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *ucontrol); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv); int snd_soc_info_xr_sx(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); int snd_soc_get_xr_sx(struct snd_kcontrol *kcontrol, @@ -1119,6 +1128,9 @@ struct soc_bytes { struct soc_bytes_ext { int max; + /* used for TLV byte control */ + int (*get)(unsigned int __user *bytes, unsigned int size); + int (*put)(const unsigned int __user *bytes, unsigned int size); }; /* multi register control */ diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b87d7d8..e6ad95b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3267,6 +3267,28 @@ int snd_soc_bytes_info_ext(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(snd_soc_bytes_info_ext); +int snd_soc_bytes_tlv_callback(struct snd_kcontrol *kcontrol, int op_flag, + unsigned int size, unsigned int __user *tlv) +{ + struct soc_bytes_ext *params = (void *)kcontrol->private_value; + unsigned int count = size < params->max ? size : params->max; + + switch (op_flag) { + case 0: + if (params->get) + params->get(tlv, count); + break; + case 1: + if (params->put) + params->put(tlv, count); + break; + default: + return -ENXIO; + } + return 0; +} +EXPORT_SYMBOL_GPL(snd_soc_bytes_tlv_callback); + /** * snd_soc_info_xr_sx - signed multi register info callback * @kcontrol: mreg control -- 1.7.0.4

2 2

[alsa-devel] [PATCH 1/2] ALSA: hda - Add the pin fixup for HP Envy TS bass speaker
by Takashi Iwai 15 Jul '14

15 Jul '14

NID 0x10 seems corresponding to the bass speaker. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 52bdbdc7786e..946ae4094aa3 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -103,6 +103,7 @@ enum { STAC_92HD83XXX_HP, STAC_HP_ENVY_BASS, STAC_HP_BNB13_EQ, + STAC_HP_ENVY_TS_BASS, STAC_92HD83XXX_MODELS }; @@ -2668,6 +2669,13 @@ static const struct hda_fixup stac92hd83xxx_fixups[] = { .chained = true, .chain_id = STAC_92HD83XXX_HP_MIC_LED, }, + [STAC_HP_ENVY_TS_BASS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x10, 0x92170111 }, + {} + }, + }, }; static const struct hda_model_fixup stac92hd83xxx_models[] = { @@ -2684,6 +2692,7 @@ static const struct hda_model_fixup stac92hd83xxx_models[] = { { .id = STAC_92HD83XXX_HEADSET_JACK, .name = "headset-jack" }, { .id = STAC_HP_ENVY_BASS, .name = "hp-envy-bass" }, { .id = STAC_HP_BNB13_EQ, .name = "hp-bnb13-eq" }, + { .id = STAC_HP_ENVY_TS_BASS, .name = "hp-envy-ts-bass" }, {} }; @@ -2739,6 +2748,8 @@ static const struct snd_pci_quirk stac92hd83xxx_fixup_tbl[] = { "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190A, "HP bNB13", STAC_HP_BNB13_EQ), + SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x190e, + "HP ENVY TS", STAC_HP_ENVY_TS_BASS), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1940, "HP bNB13", STAC_HP_BNB13_EQ), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1941, -- 2.0.1

1 1

[alsa-devel] [PATCH] ALSA: hda - Fix broken PM due to incomplete i915 initialization
by Takashi Iwai 15 Jul '14

15 Jul '14

When the initialization of Intel HDMI controller fails due to missing i915 kernel symbols (e.g. HD-audio is built in while i915 is module), the driver discontinues the probe. However, since the probe was done asynchronously, the driver object still remains, thus the relevant PM ops are still called at suspend/resume. This results in the bad access to the incomplete audio card object, eventually leads to Oops or stall at PM. This patch adds the missing checks of chip->init_failed flag at each PM callback in order to fix the problem above. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=79561 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d690c26a197c..83cd19017cf3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -596,7 +596,7 @@ static int azx_suspend(struct device *dev) struct azx *chip = card->private_data; struct azx_pcm *p; - if (chip->disabled) + if (chip->disabled || chip->init_failed) return 0; snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); @@ -628,7 +628,7 @@ static int azx_resume(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - if (chip->disabled) + if (chip->disabled || chip->init_failed) return 0; if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) { @@ -665,7 +665,7 @@ static int azx_runtime_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - if (chip->disabled) + if (chip->disabled || chip->init_failed) return 0; if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) @@ -692,7 +692,7 @@ static int azx_runtime_resume(struct device *dev) struct hda_codec *codec; int status; - if (chip->disabled) + if (chip->disabled || chip->init_failed) return 0; if (!(chip->driver_caps & AZX_DCAPS_PM_RUNTIME)) @@ -729,7 +729,7 @@ static int azx_runtime_idle(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; - if (chip->disabled) + if (chip->disabled || chip->init_failed) return 0; if (!power_save_controller || -- 2.0.1

1 0

[alsa-devel] Support for ASP DM6446 in main-line Linux Kernel
by abhijit naik 15 Jul '14

15 Jul '14

Hi, I am studying ASP driver of DM6446 EVM in main line kernel 3.14. I referred to both sound/soc/davinci/davinci-mcasp.c and sound/soc/davinci/davinci-i2s.c Which one among the two is correct for DM6446 ASP and I am seeing that there is a device tree code in sound/soc/davinci/davinci-evm.c but corresponding support code is not present in sound/soc/davinci/davinci-mcasp.c Please let me know whether this support is complete or can some one guide me to working code for DM6446 ASP -- Best wishes, Abhijit Naik

2 1

[alsa-devel] [PATCH] ASoC: Intel: Update FW version readback
by Jie Yang 15 Jul '14

15 Jul '14

Update FW version readback. IPC_GLB_GET_FW_VERSION reads back the ABI version whilst the release version is in the mailbox. Update to use mailbox version for info logging. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 20 +++++++++++++++++--- 1 file changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 4342363..a8a2cf8 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -183,7 +183,7 @@ struct sst_hsw_ipc_fw_ready { u32 inbox_size; u32 outbox_size; u32 fw_info_size; - u8 fw_info[1]; + u8 fw_info[IPC_MAX_MAILBOX_BYTES - 5 * sizeof(u32)]; } __attribute__((packed)); struct ipc_message { @@ -569,6 +569,9 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) { struct sst_hsw_ipc_fw_ready fw_ready; u32 offset; + u8 fw_info[IPC_MAX_MAILBOX_BYTES-5*sizeof(u32)]; + char *tmp[5], *pinfo; + int i = 0; offset = (header & 0x1FFFFFFF) << 3; @@ -589,6 +592,19 @@ static void hsw_fw_ready(struct sst_hsw *hsw, u32 header) fw_ready.inbox_offset, fw_ready.inbox_size); dev_dbg(hsw->dev, " mailbox downstream 0x%x - size 0x%x\n", fw_ready.outbox_offset, fw_ready.outbox_size); + if (fw_ready.fw_info_size < sizeof(fw_ready.fw_info)) { + fw_ready.fw_info[fw_ready.fw_info_size] = 0; + dev_dbg(hsw->dev, " Firmware info: %s \n", fw_ready.fw_info); + + /* log the FW version info got from the mailbox here. */ + memcpy(fw_info, fw_ready.fw_info, fw_ready.fw_info_size); + pinfo = &fw_info[0]; + for (i = 0; i < sizeof(tmp) / sizeof(char *); i++) + tmp[i] = strsep(&pinfo, " "); + dev_info(hsw->dev, "FW loaded, mailbox readback FW info: type %s, - " + "version: %s.%s, build %s, source commit id: %s\n", + tmp[0], tmp[1], tmp[2], tmp[3], tmp[4]); + } } static void hsw_notification_work(struct work_struct *work) @@ -1775,8 +1791,6 @@ int sst_hsw_dsp_init(struct device *dev, struct sst_pdata *pdata) /* get the FW version */ sst_hsw_fw_get_version(hsw, &version); - dev_info(hsw->dev, "FW loaded: type %d - version: %d.%d build %d\n", - version.type, version.major, version.minor, version.build); /* get the globalmixer */ ret = sst_hsw_mixer_get_info(hsw); -- 1.8.3.2

1 0

[alsa-devel] [PATCH v6] ASoC: tas2552: Support TI TAS2552 Amplifier
by Dan Murphy 14 Jul '14

14 Jul '14

Support the TI TAS2552 Class D amplifier. The TAS2552 is a high efficiency Class-D audio power amplifier with advanced battery current management and an integrated Class-G boost The device constantly measures the current and voltage across the load and provides a digital stream of this information. Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- v6 - Addressed comments from v5 also added PLL algorithim instead of hard coding PLL registers - https://patchwork.kernel.org/patch/4476001/ v5 - Consolidated dai_fmt call back to write serial control register once - https://patchwork.kernel.org/patch/4474531/ v4 - Added pm_runtime support, removed magical numbers, added regulator support, changed from legacy gpio to new gpiod calls, fixed default register values, flipped on battery tracking support and added suspend/resume support - https://patchwork.kernel.org/patch/4465611/ v3 - Updated bindings doc per comments, rearranged probe pdata vs np check - https://patchwork.kernel.org/patch/4453481/ v2 - Address RFC comments- Added regmap, and snd_soc calls removed debug code, address checkpatch errors -https://patchwork.kernel.org/patch/4378281/ .../devicetree/bindings/sound/tas2552.txt | 25 + include/sound/tas2552-plat.h | 25 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2552.c | 538 ++++++++++++++++++++ sound/soc/codecs/tas2552.h | 129 +++++ 6 files changed, 724 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tas2552.txt create mode 100644 include/sound/tas2552-plat.h create mode 100644 sound/soc/codecs/tas2552.c create mode 100644 sound/soc/codecs/tas2552.h diff --git a/Documentation/devicetree/bindings/sound/tas2552.txt b/Documentation/devicetree/bindings/sound/tas2552.txt new file mode 100644 index 0000000..00223dc --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tas2552.txt @@ -0,0 +1,25 @@ +Texas Instruments - tas2552 Codec module + +The tas2552 serial control bus communicates through I2C protocols + +Required properties: + +- compatible - One of: + "ti,tas2552" - TAS2552 + +- reg - I2C slave address + +Optional properties: + +- enable-gpio - gpio pin to enable/disable the device + +Example: + +tas2552: tas2552@41 { + compatible = "ti,tas2552"; + reg = <0x41>; + enable-gpio = <&gpio4 2 GPIO_ACTIVE_HIGH>; +}; + +For more product information please see the link below: +http://www.ti.com/product/TAS2552 diff --git a/include/sound/tas2552-plat.h b/include/sound/tas2552-plat.h new file mode 100644 index 0000000..65e7627 --- /dev/null +++ b/include/sound/tas2552-plat.h @@ -0,0 +1,25 @@ +/* + * TAS2552 driver platform header + * + * Copyright (C) 2014 Texas Instruments Inc. + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * 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. + */ + +#ifndef TAS2552_PLAT_H +#define TAS2552_PLAT_H + +struct tas2552_platform_data { + int enable_gpio; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0b9571c..cc09261 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -99,6 +99,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C + select SND_SOC_TAS2552 if I2C select SND_SOC_TLV320DAC33 if I2C select SND_SOC_TWL4030 if TWL4030_CORE select SND_SOC_TWL6040 if TWL6040_CORE @@ -754,4 +755,8 @@ config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C +config SND_SOC_TAS2552 + tristate "Texas Instruments TAS2552 Mono Audio amplifier" + depends on I2C + endmenu diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1bd6e1c..33bc7228 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -162,6 +162,7 @@ snd-soc-wm-hubs-objs := wm_hubs.o # Amp snd-soc-max9877-objs := max9877.o snd-soc-tpa6130a2-objs := tpa6130a2.o +snd-soc-tas2552-objs := tas2552.o obj-$(CONFIG_SND_SOC_88PM860X) += snd-soc-88pm860x.o obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o @@ -325,3 +326,4 @@ obj-$(CONFIG_SND_SOC_WM_HUBS) += snd-soc-wm-hubs.o # Amp obj-$(CONFIG_SND_SOC_MAX9877) += snd-soc-max9877.o obj-$(CONFIG_SND_SOC_TPA6130A2) += snd-soc-tpa6130a2.o +obj-$(CONFIG_SND_SOC_TAS2552) += snd-soc-tas2552.o diff --git a/sound/soc/codecs/tas2552.c b/sound/soc/codecs/tas2552.c new file mode 100644 index 0000000..7b762d4 --- /dev/null +++ b/sound/soc/codecs/tas2552.c @@ -0,0 +1,538 @@ +/* + * tas2552.c - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * 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/errno.h> +#include <linux/device.h> +#include <linux/i2c.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/slab.h> + +#include <linux/gpio/consumer.h> +#include <linux/regulator/consumer.h> + +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/tlv.h> +#include <sound/tas2552-plat.h> + +#include "tas2552.h" + +static struct reg_default tas2552_reg_defs[] = { + {TAS2552_CFG_1, 0x22}, + {TAS2552_CFG_3, 0x80}, + {TAS2552_DOUT, 0x00}, + {TAS2552_OUTPUT_DATA, 0xc0}, + {TAS2552_PDM_CFG, 0x01}, + {TAS2552_PGA_GAIN, 0x00}, + {TAS2552_BOOST_PT_CTRL, 0x0f}, + {TAS2552_RESERVED_0D, 0x00}, + {TAS2552_LIMIT_RATE_HYS, 0x08}, + {TAS2552_CFG_2, 0xef}, + {TAS2552_SER_CTRL_1, 0x00}, + {TAS2552_SER_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_1, 0x10}, + {TAS2552_PLL_CTRL_2, 0x00}, + {TAS2552_PLL_CTRL_3, 0x00}, + {TAS2552_BTIP, 0x8f}, + {TAS2552_BTS_CTRL, 0x80}, + {TAS2552_LIMIT_RELEASE, 0x04}, + {TAS2552_LIMIT_INT_COUNT, 0x00}, + {TAS2552_EDGE_RATE_CTRL, 0x40}, + {TAS2552_VBAT_DATA, 0x00}, +}; + +#define TAS2552_NUM_SUPPLIES 3 +static const char *tas2552_supply_names[TAS2552_NUM_SUPPLIES] = { + "vbat", /* vbat voltage */ + "iovdd", /* I/O Voltage */ + "avdd", /* Analog DAC Voltage */ +}; + +struct tas2552_data { + struct snd_soc_codec *codec; + struct regmap *regmap; + struct i2c_client *tas2552_client; + struct regulator_bulk_data supplies[TAS2552_NUM_SUPPLIES]; + struct gpio_desc *enable_gpio; + unsigned char regs[TAS2552_VBAT_DATA]; + unsigned int mclk; +}; + +static void tas2552_sw_shutdown(struct tas2552_data *tas_data, int sw_shutdown) +{ + u8 cfg1_reg; + + if (sw_shutdown) + cfg1_reg = 0; + else + cfg1_reg = TAS2552_SWS_MASK; + + snd_soc_update_bits(tas_data->codec, TAS2552_CFG_1, + TAS2552_SWS_MASK, cfg1_reg); +} + +static int tas2552_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev); + int sample_rate, pll_clk; + int d; + u8 p, j; + + /* Turn on Class D amplifier */ + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN_MASK, + TAS2552_CLASSD_EN); + + if (!tas2552->mclk) + return -EINVAL; + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, 0); + + if (tas2552->mclk == TAS2552_245MHZ_CLK || + tas2552->mclk == TAS2552_225MHZ_CLK) { + /* By pass the PLL configuration */ + snd_soc_update_bits(codec, TAS2552_PLL_CTRL_2, + TAS2552_PLL_BYPASS_MASK, + TAS2552_PLL_BYPASS); + } else { + /* Fill in the PLL control registers for J & D + * PLL_CLK = (.5 * freq * J.D) / 2^p + * Need to fill in J and D here based on incoming freq + */ + p = snd_soc_read(codec, TAS2552_PLL_CTRL_1); + p = (p >> 7); + sample_rate = params_rate(params); + + if (sample_rate == 48000) + pll_clk = TAS2552_245MHZ_CLK; + else if (sample_rate == 44100) + pll_clk = TAS2552_225MHZ_CLK; + else { + dev_vdbg(codec->dev, "Substream sample rate is not found %i\n", + params_rate(params)); + return -EINVAL; + } + + j = (pll_clk * 2 * (1 << p)) / tas2552->mclk; + d = (pll_clk * 2 * (1 << p)) % tas2552->mclk; + + snd_soc_update_bits(codec, TAS2552_PLL_CTRL_1, + TAS2552_PLL_J_MASK, j); + snd_soc_write(codec, TAS2552_PLL_CTRL_2, + (d >> 7) & TAS2552_PLL_D_UPPER_MASK); + snd_soc_write(codec, TAS2552_PLL_CTRL_3, + d & TAS2552_PLL_D_LOWER_MASK); + + } + + snd_soc_update_bits(codec, TAS2552_CFG_2, TAS2552_PLL_ENABLE, + TAS2552_PLL_ENABLE); + + return 0; +} + +static int tas2552_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 serial_format; + u8 serial_control_mask; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + serial_format = 0x00; + break; + case SND_SOC_DAIFMT_CBS_CFM: + serial_format = TAS2552_WORD_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFS: + serial_format = TAS2552_BIT_CLK_MASK; + break; + case SND_SOC_DAIFMT_CBM_CFM: + serial_format = (TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK); + break; + default: + dev_vdbg(codec->dev, "DAI Format master is not found\n"); + return -EINVAL; + } + + serial_control_mask = TAS2552_BIT_CLK_MASK | TAS2552_WORD_CLK_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + serial_format &= TAS2552_DAIFMT_I2S_MASK; + break; + case SND_SOC_DAIFMT_DSP_A: + serial_format |= TAS2552_DAIFMT_DSP; + break; + case SND_SOC_DAIFMT_RIGHT_J: + serial_format |= TAS2552_DAIFMT_RIGHT_J; + break; + case SND_SOC_DAIFMT_LEFT_J: + serial_format |= TAS2552_DAIFMT_LEFT_J; + break; + default: + dev_vdbg(codec->dev, "DAI Format is not found\n"); + return -EINVAL; + } + + if (fmt & SND_SOC_DAIFMT_FORMAT_MASK) + serial_control_mask |= TAS2552_DATA_FORMAT_MASK; + + snd_soc_update_bits(codec, TAS2552_SER_CTRL_1, serial_control_mask, + serial_format); + + return 0; +} + +static int tas2552_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas2552_data *tas2552 = dev_get_drvdata(codec->dev); + + tas2552->mclk = freq; + + return 0; +} + +static int tas2552_mute(struct snd_soc_dai *dai, int mute) +{ + u8 cfg1_reg; + struct snd_soc_codec *codec = dai->codec; + + if (mute) + cfg1_reg = TAS2552_MUTE_MASK; + else + cfg1_reg = ~TAS2552_MUTE_MASK; + + snd_soc_update_bits(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK, cfg1_reg); + + return 0; +} + +#ifdef CONFIG_PM_RUNTIME +static int tas2552_runtime_suspend(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + tas2552_sw_shutdown(tas2552, 0); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + regcache_cache_only(tas2552->regmap, true); + regcache_mark_dirty(tas2552->regmap); + + return 0; +} + +static int tas2552_runtime_resume(struct device *dev) +{ + struct tas2552_data *tas2552 = dev_get_drvdata(dev); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 1); + + tas2552_sw_shutdown(tas2552, 1); + + regcache_cache_only(tas2552->regmap, false); + regcache_sync(tas2552->regmap); + + return 0; +} +#endif + +static const struct dev_pm_ops tas2552_pm = { + SET_RUNTIME_PM_OPS(tas2552_runtime_suspend, tas2552_runtime_resume, + NULL) +}; + +static int tas2552_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + pm_runtime_get_sync(codec->dev); + + return 0; +} + +static void tas2552_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + + pm_runtime_put(codec->dev); +} + +static struct snd_soc_dai_ops tas2552_speaker_dai_ops = { + .hw_params = tas2552_hw_params, + .set_sysclk = tas2552_set_dai_sysclk, + .set_fmt = tas2552_set_dai_fmt, + .startup = tas2552_startup, + .shutdown = tas2552_shutdown, + .digital_mute = tas2552_mute, +}; + +/* Formats supported by TAS2552 driver. */ +#define TAS2552_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) + +/* TAS2552 dai structure. */ +static struct snd_soc_dai_driver tas2552_dai[] = { + { + .name = "tas2552-amplifier", + .playback = { + .stream_name = "Speaker", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = TAS2552_FORMATS, + }, + .ops = &tas2552_speaker_dai_ops, + }, +}; + +/* + * DAC digital volumes. From -7 to 24 dB in 1 dB steps + */ +static DECLARE_TLV_DB_SCALE(dac_tlv, -7, 100, 24); + +static const struct snd_kcontrol_new tas2552_snd_controls[] = { + SOC_SINGLE_TLV("Speaker Driver Playback Volume", + TAS2552_PGA_GAIN, 0, 0x1f, 1, dac_tlv), +}; + +static const struct reg_default tas2552_init_regs[] = { + { TAS2552_RESERVED_0D, 0xc0 }, +}; + +static int tas2552_codec_probe(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + tas2552->codec = codec; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + return ret; + } + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 1); + + pm_runtime_get_sync(codec->dev); + + snd_soc_write(codec, TAS2552_CFG_1, TAS2552_MUTE_MASK | + TAS2552_PLL_SRC_BCLK); + snd_soc_write(codec, TAS2552_CFG_3, TAS2552_I2S_OUT_SEL | + TAS2552_DIN_SRC_SEL_AVG_L_R | TAS2552_88_96KHZ); + snd_soc_write(codec, TAS2552_DOUT, TAS2552_PDM_DATA_I); + snd_soc_write(codec, TAS2552_OUTPUT_DATA, TAS2552_PDM_DATA_V_I | 0x8); + snd_soc_write(codec, TAS2552_PDM_CFG, TAS2552_PDM_BCLK_SEL); + snd_soc_write(codec, TAS2552_BOOST_PT_CTRL, TAS2552_APT_DELAY_200 | + TAS2552_APT_THRESH_2_1_7); + + ret = regmap_register_patch(tas2552->regmap, tas2552_init_regs, + ARRAY_SIZE(tas2552_init_regs)); + if (ret != 0) + dev_err(codec->dev, "Failed to write init registers: %d\n", + ret); + + snd_soc_write(codec, TAS2552_CFG_2, TAS2552_CLASSD_EN | + TAS2552_BOOST_EN | TAS2552_APT_EN | + TAS2552_PLL_ENABLE | TAS2552_LIM_EN); + return 0; +} + +static int tas2552_codec_remove(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + + if (tas2552->enable_gpio) + gpiod_set_value(tas2552->enable_gpio, 0); + + return 0; +}; + +#ifdef CONFIG_PM +static int tas2552_suspend(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + pm_runtime_put(codec->dev); + + ret = regulator_bulk_disable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) + dev_err(codec->dev, "Failed to disable supplies: %d\n", + ret); + return 0; +} + +static int tas2552_resume(struct snd_soc_codec *codec) +{ + struct tas2552_data *tas2552 = snd_soc_codec_get_drvdata(codec); + int ret; + + ret = regulator_bulk_enable(ARRAY_SIZE(tas2552->supplies), + tas2552->supplies); + + if (ret != 0) { + dev_err(codec->dev, "Failed to enable supplies: %d\n", + ret); + } + + pm_runtime_get_sync(codec->dev); + + return 0; +} +#else +#define tas2552_suspend NULL +#define tas2552_resume NULL +#endif + +static struct snd_soc_codec_driver soc_codec_dev_tas2552 = { + .probe = tas2552_codec_probe, + .remove = tas2552_codec_remove, + .suspend = tas2552_suspend, + .resume = tas2552_resume, + .controls = tas2552_snd_controls, + .num_controls = ARRAY_SIZE(tas2552_snd_controls), +}; + +static const struct regmap_config tas2552_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = TAS2552_MAX_REG, + .reg_defaults = tas2552_reg_defs, + .num_reg_defaults = ARRAY_SIZE(tas2552_reg_defs), + .cache_type = REGCACHE_RBTREE, +}; + +static int tas2552_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device *dev; + struct tas2552_data *data; + int ret; + int i; + + dev = &client->dev; + data = devm_kzalloc(&client->dev, sizeof(*data), GFP_KERNEL); + if (data == NULL) + return -ENOMEM; + + data->enable_gpio = devm_gpiod_get(dev, "enable"); + if (IS_ERR(data->enable_gpio)) { + ret = PTR_ERR(data->enable_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return ret; + + data->enable_gpio = NULL; + } else { + gpiod_direction_output(data->enable_gpio, 0); + } + + data->tas2552_client = client; + data->regmap = devm_regmap_init_i2c(client, &tas2552_regmap_config); + if (IS_ERR(data->regmap)) { + ret = PTR_ERR(data->regmap); + dev_err(&client->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tas2552_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), + data->supplies); + if (ret != 0) + dev_err(dev, "Failed to request supplies: %d\n", ret); + + pm_runtime_set_active(&client->dev); + pm_runtime_set_autosuspend_delay(&client->dev, 1000); + pm_runtime_use_autosuspend(&client->dev); + pm_runtime_enable(&client->dev); + pm_runtime_mark_last_busy(&client->dev); + pm_runtime_put_sync_autosuspend(&client->dev); + + dev_set_drvdata(&client->dev, data); + + ret = snd_soc_register_codec(&client->dev, + &soc_codec_dev_tas2552, + tas2552_dai, ARRAY_SIZE(tas2552_dai)); + if (ret < 0) + dev_err(&client->dev, "Failed to register codec: %d\n", ret); + + return 0; +} + +static int tas2552_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id tas2552_id[] = { + { "tas2552-codec", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas2552_id); + +#if IS_ENABLED(CONFIG_OF) +static const struct of_device_id tas2552_of_match[] = { + { .compatible = "ti,tas2552", }, + {}, +}; +MODULE_DEVICE_TABLE(of, tas2552_of_match); +#endif + +static struct i2c_driver tas2552_i2c_driver = { + .driver = { + .name = "tas2552", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas2552_of_match), + .pm = &tas2552_pm, + }, + .probe = tas2552_probe, + .remove = tas2552_i2c_remove, + .id_table = tas2552_id, +}; + +module_i2c_driver(tas2552_i2c_driver); + +MODULE_AUTHOR("Dan Muprhy <dmurphy(a)ti.com>"); +MODULE_DESCRIPTION("TAS2552 Audio amplifier driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2552.h b/sound/soc/codecs/tas2552.h new file mode 100644 index 0000000..6cea8f3 --- /dev/null +++ b/sound/soc/codecs/tas2552.h @@ -0,0 +1,129 @@ +/* + * tas2552.h - ALSA SoC Texas Instruments TAS2552 Mono Audio Amplifier + * + * Copyright (C) 2014 Texas Instruments Incorporated - http://www.ti.com + * + * Author: Dan Murphy <dmurphy(a)ti.com> + * + * 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. + */ + +#ifndef __TAS2552_H__ +#define __TAS2552_H__ + +/* Register Address Map */ +#define TAS2552_DEVICE_STATUS 0x00 +#define TAS2552_CFG_1 0x01 +#define TAS2552_CFG_2 0x02 +#define TAS2552_CFG_3 0x03 +#define TAS2552_DOUT 0x04 +#define TAS2552_SER_CTRL_1 0x05 +#define TAS2552_SER_CTRL_2 0x06 +#define TAS2552_OUTPUT_DATA 0x07 +#define TAS2552_PLL_CTRL_1 0x08 +#define TAS2552_PLL_CTRL_2 0x09 +#define TAS2552_PLL_CTRL_3 0x0a +#define TAS2552_BTIP 0x0b +#define TAS2552_BTS_CTRL 0x0c +#define TAS2552_RESERVED_0D 0x0d +#define TAS2552_LIMIT_RATE_HYS 0x0e +#define TAS2552_LIMIT_RELEASE 0x0f +#define TAS2552_LIMIT_INT_COUNT 0x10 +#define TAS2552_PDM_CFG 0x11 +#define TAS2552_PGA_GAIN 0x12 +#define TAS2552_EDGE_RATE_CTRL 0x13 +#define TAS2552_BOOST_PT_CTRL 0x14 +#define TAS2552_VER_NUM 0x16 +#define TAS2552_VBAT_DATA 0x19 +#define TAS2552_MAX_REG 0x20 + +/* CFG1 Register Masks */ +#define TAS2552_MUTE_MASK (1 << 2) +#define TAS2552_SWS_MASK (1 << 1) +#define TAS2552_WCLK_MASK 0x07 +#define TAS2552_CLASSD_EN_MASK (1 << 7) + +/* CFG2 Register Masks */ +#define TAS2552_CLASSD_EN (1 << 7) +#define TAS2552_BOOST_EN (1 << 6) +#define TAS2552_APT_EN (1 << 5) +#define TAS2552_PLL_ENABLE (1 << 3) +#define TAS2552_LIM_EN (1 << 2) +#define TAS2552_IVSENSE_EN (1 << 1) + +/* CFG3 Register Masks */ +#define TAS2552_WORD_CLK_MASK (1 << 7) +#define TAS2552_BIT_CLK_MASK (1 << 6) +#define TAS2552_DATA_FORMAT_MASK (0x11 << 2) + +#define TAS2552_DAIFMT_I2S_MASK 0xf3 +#define TAS2552_DAIFMT_DSP (1 << 3) +#define TAS2552_DAIFMT_RIGHT_J (1 << 4) +#define TAS2552_DAIFMT_LEFT_J (0x11 << 3) + +#define TAS2552_PLL_SRC_MCLK 0x00 +#define TAS2552_PLL_SRC_BCLK (1 << 3) +#define TAS2552_PLL_SRC_IVCLKIN (1 << 4) +#define TAS2552_PLL_SRC_1_8_FIXED (0x11 << 3) + +#define TAS2552_DIN_SRC_SEL_MUTED 0x00 +#define TAS2552_DIN_SRC_SEL_LEFT (1 << 4) +#define TAS2552_DIN_SRC_SEL_RIGHT (1 << 5) +#define TAS2552_DIN_SRC_SEL_AVG_L_R (0x11 << 4) + +#define TAS2552_PDM_IN_SEL (1 << 5) +#define TAS2552_I2S_OUT_SEL (1 << 6) +#define TAS2552_ANALOG_IN_SEL (1 << 7) + +/* CFG3 WCLK Dividers */ +#define TAS2552_8KHZ 0x00 +#define TAS2552_11_12KHZ (1 << 1) +#define TAS2552_16KHZ (1 << 2) +#define TAS2552_22_24KHZ (1 << 3) +#define TAS2552_32KHZ (1 << 4) +#define TAS2552_44_48KHZ (1 << 5) +#define TAS2552_88_96KHZ (1 << 6) +#define TAS2552_176_192KHZ (1 << 7) + +/* OUTPUT_DATA register */ +#define TAS2552_PDM_DATA_I 0x00 +#define TAS2552_PDM_DATA_V (1 << 6) +#define TAS2552_PDM_DATA_I_V (1 << 7) +#define TAS2552_PDM_DATA_V_I (0x11 << 6) + +/* PDM CFG Register */ +#define TAS2552_PDM_DATA_ES_RISE 0x4 + +#define TAS2552_PDM_PLL_CLK_SEL 0x00 +#define TAS2552_PDM_IV_CLK_SEL (1 << 1) +#define TAS2552_PDM_BCLK_SEL (1 << 2) +#define TAS2552_PDM_MCLK_SEL (1 << 3) + +/* Boost pass-through register */ +#define TAS2552_APT_DELAY_50 0x00 +#define TAS2552_APT_DELAY_75 (1 << 1) +#define TAS2552_APT_DELAY_125 (1 << 2) +#define TAS2552_APT_DELAY_200 (1 << 3) + +#define TAS2552_APT_THRESH_2_5 0x00 +#define TAS2552_APT_THRESH_1_7 (1 << 3) +#define TAS2552_APT_THRESH_1_4_1_1 (1 << 4) +#define TAS2552_APT_THRESH_2_1_7 (0x11 << 2) + +/* PLL Control Register */ +#define TAS2552_245MHZ_CLK 24576000 +#define TAS2552_225MHZ_CLK 22579200 +#define TAS2552_PLL_J_MASK 0x7f +#define TAS2552_PLL_D_UPPER_MASK 0x3f +#define TAS2552_PLL_D_LOWER_MASK 0xff +#define TAS2552_PLL_BYPASS_MASK 0x80 +#define TAS2552_PLL_BYPASS 0x80 + +#endif -- 1.7.9.5

3 4

[alsa-devel] [PATCH RESEND] ASoC: rt286: make rt286_i2c_driver static
by bardliao＠realtek.com 14 Jul '14

14 Jul '14

From: Bard Liao <bardliao(a)realtek.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index 8103315..218f86e 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -1206,7 +1206,7 @@ static int rt286_i2c_remove(struct i2c_client *i2c) } -struct i2c_driver rt286_i2c_driver = { +static struct i2c_driver rt286_i2c_driver = { .driver = { .name = "rt286", .owner = THIS_MODULE, -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH v2 0/4] ASoC: Intel: Enable BDW/HSW SRAM power gating
by Jie Yang 14 Jul '14

14 Jul '14

Hi Mark, Here is the patch series for enable Intel Broadwell/Haswell ADSP SRAM power gating feature. Patches 0001~0002 merge wcp DRAM regions and correct the SRAM bit shift for it; Patch 0003 add dummy read to workaround SRAM write missing bytes issue; Patch 0004 will make SRAM start with all memory banks disabled, and enabled required banks during boot procedure. thanks, Keyon Changes in v2: * remove the Add persistent area alloc patch, for it is not so related with power gating feature. Jie Yang (4): ASoC: Intel: Merge wild cat point ADSP DRAM regions ASoC: Intel: Use a table for ADSP SRAM shift ASoC: Intel: Add dummy read for SRAM block enable ASoC: Intel: Start with all memory banks disabled sound/soc/intel/sst-haswell-dsp.c | 61 +++++++++++++++++++++++++++++---------- 1 file changed, 46 insertions(+), 15 deletions(-) -- 1.8.3.2

2 8

[alsa-devel] [PATCH] ASoC: Intel: Cleanup HSW pcm format support
by Jie Yang 14 Jul '14

14 Jul '14

This change removes unsupported formats from System, Capture and Loopback FE DAIs. Also it fixes S24_LE support on all DAIs. While at this fix 24 bit flag for BYT as well. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-baytrail-pcm.c | 2 +- sound/soc/intel/sst-haswell-pcm.c | 27 ++++++++++++++++++--------- 2 files changed, 19 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 8eab973..599401c 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -32,7 +32,7 @@ static const struct snd_pcm_hardware sst_byt_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, .formats = SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FORMAT_S24_LE, + SNDRV_PCM_FMTBIT_S24_LE, .period_bytes_min = 384, .period_bytes_max = 48000, .periods_min = 2, diff --git a/sound/soc/intel/sst-haswell-pcm.c b/sound/soc/intel/sst-haswell-pcm.c index 7ec97d7..9ee82ae 100644 --- a/sound/soc/intel/sst-haswell-pcm.c +++ b/sound/soc/intel/sst-haswell-pcm.c @@ -81,7 +81,7 @@ static const struct snd_pcm_hardware hsw_pcm_hardware = { SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_NO_PERIOD_WAKEUP, - .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE | + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, .period_bytes_min = PAGE_SIZE, .period_bytes_max = (HSW_PCM_PERIODS_MAX / HSW_PCM_PERIODS_MIN) * PAGE_SIZE, @@ -420,7 +420,15 @@ static int hsw_pcm_hw_params(struct snd_pcm_substream *substream, sst_hsw_stream_set_valid(hsw, pcm_data->stream, 16); break; case SNDRV_PCM_FORMAT_S24_LE: - bits = SST_HSW_DEPTH_24BIT; + bits = SST_HSW_DEPTH_32BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 24); + break; + case SNDRV_PCM_FORMAT_S8: + bits = SST_HSW_DEPTH_8BIT; + sst_hsw_stream_set_valid(hsw, pcm_data->stream, 8); + break; + case SNDRV_PCM_FORMAT_S32_LE: + bits = SST_HSW_DEPTH_32BIT; sst_hsw_stream_set_valid(hsw, pcm_data->stream, 32); break; default: @@ -710,8 +718,9 @@ static int hsw_pcm_new(struct snd_soc_pcm_runtime *rtd) } #define HSW_FORMATS \ - (SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ - SNDRV_PCM_FMTBIT_S32_LE) + (SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S8) static struct snd_soc_dai_driver hsw_dais[] = { { @@ -721,7 +730,7 @@ static struct snd_soc_dai_driver hsw_dais[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -752,8 +761,8 @@ static struct snd_soc_dai_driver hsw_dais[] = { .stream_name = "Loopback Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = HSW_FORMATS, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, { @@ -762,8 +771,8 @@ static struct snd_soc_dai_driver hsw_dais[] = { .stream_name = "Analog Capture", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000, - .formats = HSW_FORMATS, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S16_LE, }, }, }; -- 1.8.3.2

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