Alsa-devel May 2014

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[alsa-devel] [PATCH 1/2] usbtv: split core and video implementation
by Federico Simoncelli 19 Aug '14

19 Aug '14

From: Federico Simoncelli <fsimonce(a)redhat.com> Signed-off-by: Federico Simoncelli <fsimonce(a)redhat.com> Reviewed-by: Lubomir Rintel <lkundrak(a)v3.sk> --- drivers/media/usb/usbtv/Makefile | 3 + drivers/media/usb/usbtv/usbtv-core.c | 136 +++++++++++++++++ drivers/media/usb/usbtv/{usbtv.c => usbtv-video.c} | 163 ++------------------- drivers/media/usb/usbtv/usbtv.h | 98 +++++++++++++ 4 files changed, 246 insertions(+), 154 deletions(-) create mode 100644 drivers/media/usb/usbtv/usbtv-core.c rename drivers/media/usb/usbtv/{usbtv.c => usbtv-video.c} (82%) create mode 100644 drivers/media/usb/usbtv/usbtv.h diff --git a/drivers/media/usb/usbtv/Makefile b/drivers/media/usb/usbtv/Makefile index 28b872f..775316a 100644 --- a/drivers/media/usb/usbtv/Makefile +++ b/drivers/media/usb/usbtv/Makefile @@ -1 +1,4 @@ +usbtv-y := usbtv-core.o \ + usbtv-video.o + obj-$(CONFIG_VIDEO_USBTV) += usbtv.o diff --git a/drivers/media/usb/usbtv/usbtv-core.c b/drivers/media/usb/usbtv/usbtv-core.c new file mode 100644 index 0000000..e89e48b --- /dev/null +++ b/drivers/media/usb/usbtv/usbtv-core.c @@ -0,0 +1,136 @@ +/* + * Fushicai USBTV007 Video Grabber Driver + * + * Product web site: + * http://www.fushicai.com/products_detail/&productId=d05449ee-b690-42f9-a661-… + * + * Following LWN articles were very useful in construction of this driver: + * Video4Linux2 API series: http://lwn.net/Articles/203924/ + * videobuf2 API explanation: http://lwn.net/Articles/447435/ + * Thanks go to Jonathan Corbet for providing this quality documentation. + * He is awesome. + * + * Copyright (c) 2013 Lubomir Rintel + * All rights reserved. + * No physical hardware was harmed running Windows during the + * reverse-engineering activity + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions, and the following disclaimer, + * without modification. + * 2. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * Alternatively, this software may be distributed under the terms of the + * GNU General Public License ("GPL"). + */ + +#include <linux/module.h> + +#include "usbtv.h" + +int usbtv_set_regs(struct usbtv *usbtv, const u16 regs[][2], int size) +{ + int ret; + int pipe = usb_rcvctrlpipe(usbtv->udev, 0); + int i; + + for (i = 0; i < size; i++) { + u16 index = regs[i][0]; + u16 value = regs[i][1]; + + ret = usb_control_msg(usbtv->udev, pipe, USBTV_REQUEST_REG, + USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, + value, index, NULL, 0, 0); + if (ret < 0) + return ret; + } + + return 0; +} + +static int usbtv_probe(struct usb_interface *intf, + const struct usb_device_id *id) +{ + int ret; + int size; + struct device *dev = &intf->dev; + struct usbtv *usbtv; + + /* Checks that the device is what we think it is. */ + if (intf->num_altsetting != 2) + return -ENODEV; + if (intf->altsetting[1].desc.bNumEndpoints != 4) + return -ENODEV; + + /* Packet size is split into 11 bits of base size and count of + * extra multiplies of it.*/ + size = usb_endpoint_maxp(&intf->altsetting[1].endpoint[0].desc); + size = (size & 0x07ff) * (((size & 0x1800) >> 11) + 1); + + /* Device structure */ + usbtv = kzalloc(sizeof(struct usbtv), GFP_KERNEL); + if (usbtv == NULL) + return -ENOMEM; + usbtv->dev = dev; + usbtv->udev = usb_get_dev(interface_to_usbdev(intf)); + + usbtv->iso_size = size; + + usb_set_intfdata(intf, usbtv); + + ret = usbtv_video_init(usbtv); + if (ret < 0) + goto usbtv_video_fail; + + /* for simplicity we exploit the v4l2_device reference counting */ + v4l2_device_get(&usbtv->v4l2_dev); + + dev_info(dev, "Fushicai USBTV007 Video Grabber\n"); + return 0; + +usbtv_video_fail: + kfree(usbtv); + + return ret; +} + +static void usbtv_disconnect(struct usb_interface *intf) +{ + struct usbtv *usbtv = usb_get_intfdata(intf); + usb_set_intfdata(intf, NULL); + + if (!usbtv) + return; + + usbtv_video_free(usbtv); + + usb_put_dev(usbtv->udev); + usbtv->udev = NULL; + + /* the usbtv structure will be deallocated when v4l2 will be + done using it */ + v4l2_device_put(&usbtv->v4l2_dev); +} + +struct usb_device_id usbtv_id_table[] = { + { USB_DEVICE(0x1b71, 0x3002) }, + {} +}; +MODULE_DEVICE_TABLE(usb, usbtv_id_table); + +MODULE_AUTHOR("Lubomir Rintel"); +MODULE_DESCRIPTION("Fushicai USBTV007 Video Grabber Driver"); +MODULE_LICENSE("Dual BSD/GPL"); + +struct usb_driver usbtv_usb_driver = { + .name = "usbtv", + .id_table = usbtv_id_table, + .probe = usbtv_probe, + .disconnect = usbtv_disconnect, +}; + +module_usb_driver(usbtv_usb_driver); diff --git a/drivers/media/usb/usbtv/usbtv.c b/drivers/media/usb/usbtv/usbtv-video.c similarity index 82% rename from drivers/media/usb/usbtv/usbtv.c rename to drivers/media/usb/usbtv/usbtv-video.c index 6222a4a..496bc2e 100644 --- a/drivers/media/usb/usbtv/usbtv.c +++ b/drivers/media/usb/usbtv/usbtv-video.c @@ -28,45 +28,10 @@ * GNU General Public License ("GPL"). */ -#include <linux/init.h> -#include <linux/list.h> -#include <linux/module.h> -#include <linux/slab.h> -#include <linux/usb.h> -#include <linux/videodev2.h> - -#include <media/v4l2-device.h> #include <media/v4l2-ioctl.h> #include <media/videobuf2-core.h> -#include <media/videobuf2-vmalloc.h> - -/* Hardware. */ -#define USBTV_VIDEO_ENDP 0x81 -#define USBTV_BASE 0xc000 -#define USBTV_REQUEST_REG 12 - -/* Number of concurrent isochronous urbs submitted. - * Higher numbers was seen to overly saturate the USB bus. */ -#define USBTV_ISOC_TRANSFERS 16 -#define USBTV_ISOC_PACKETS 8 - -#define USBTV_CHUNK_SIZE 256 -#define USBTV_CHUNK 240 - -/* Chunk header. */ -#define USBTV_MAGIC_OK(chunk) ((be32_to_cpu(chunk[0]) & 0xff000000) \ - == 0x88000000) -#define USBTV_FRAME_ID(chunk) ((be32_to_cpu(chunk[0]) & 0x00ff0000) >> 16) -#define USBTV_ODD(chunk) ((be32_to_cpu(chunk[0]) & 0x0000f000) >> 15) -#define USBTV_CHUNK_NO(chunk) (be32_to_cpu(chunk[0]) & 0x00000fff) - -#define USBTV_TV_STD (V4L2_STD_525_60 | V4L2_STD_PAL) - -/* parameters for supported TV norms */ -struct usbtv_norm_params { - v4l2_std_id norm; - int cap_width, cap_height; -}; + +#include "usbtv.h" static struct usbtv_norm_params norm_params[] = { { @@ -81,43 +46,6 @@ static struct usbtv_norm_params norm_params[] = { } }; -/* A single videobuf2 frame buffer. */ -struct usbtv_buf { - struct vb2_buffer vb; - struct list_head list; -}; - -/* Per-device structure. */ -struct usbtv { - struct device *dev; - struct usb_device *udev; - struct v4l2_device v4l2_dev; - struct video_device vdev; - struct vb2_queue vb2q; - struct mutex v4l2_lock; - struct mutex vb2q_lock; - - /* List of videobuf2 buffers protected by a lock. */ - spinlock_t buflock; - struct list_head bufs; - - /* Number of currently processed frame, useful find - * out when a new one begins. */ - u32 frame_id; - int chunks_done; - - enum { - USBTV_COMPOSITE_INPUT, - USBTV_SVIDEO_INPUT, - } input; - v4l2_std_id norm; - int width, height; - int n_chunks; - int iso_size; - unsigned int sequence; - struct urb *isoc_urbs[USBTV_ISOC_TRANSFERS]; -}; - static int usbtv_configure_for_norm(struct usbtv *usbtv, v4l2_std_id norm) { int i, ret = 0; @@ -142,26 +70,6 @@ static int usbtv_configure_for_norm(struct usbtv *usbtv, v4l2_std_id norm) return ret; } -static int usbtv_set_regs(struct usbtv *usbtv, const u16 regs[][2], int size) -{ - int ret; - int pipe = usb_rcvctrlpipe(usbtv->udev, 0); - int i; - - for (i = 0; i < size; i++) { - u16 index = regs[i][0]; - u16 value = regs[i][1]; - - ret = usb_control_msg(usbtv->udev, pipe, USBTV_REQUEST_REG, - USB_DIR_OUT | USB_TYPE_VENDOR | USB_RECIP_DEVICE, - value, index, NULL, 0, 0); - if (ret < 0) - return ret; - } - - return 0; -} - static int usbtv_select_input(struct usbtv *usbtv, int input) { int ret; @@ -560,12 +468,6 @@ start_fail: return ret; } -struct usb_device_id usbtv_id_table[] = { - { USB_DEVICE(0x1b71, 0x3002) }, - {} -}; -MODULE_DEVICE_TABLE(usb, usbtv_id_table); - static int usbtv_querycap(struct file *file, void *priv, struct v4l2_capability *cap) { @@ -759,33 +661,9 @@ static void usbtv_release(struct v4l2_device *v4l2_dev) kfree(usbtv); } -static int usbtv_probe(struct usb_interface *intf, - const struct usb_device_id *id) +int usbtv_video_init(struct usbtv *usbtv) { int ret; - int size; - struct device *dev = &intf->dev; - struct usbtv *usbtv; - - /* Checks that the device is what we think it is. */ - if (intf->num_altsetting != 2) - return -ENODEV; - if (intf->altsetting[1].desc.bNumEndpoints != 4) - return -ENODEV; - - /* Packet size is split into 11 bits of base size and count of - * extra multiplies of it.*/ - size = usb_endpoint_maxp(&intf->altsetting[1].endpoint[0].desc); - size = (size & 0x07ff) * (((size & 0x1800) >> 11) + 1); - - /* Device structure */ - usbtv = kzalloc(sizeof(struct usbtv), GFP_KERNEL); - if (usbtv == NULL) - return -ENOMEM; - usbtv->dev = dev; - usbtv->udev = usb_get_dev(interface_to_usbdev(intf)); - - usbtv->iso_size = size; (void)usbtv_configure_for_norm(usbtv, V4L2_STD_525_60); @@ -805,20 +683,18 @@ static int usbtv_probe(struct usb_interface *intf, usbtv->vb2q.lock = &usbtv->vb2q_lock; ret = vb2_queue_init(&usbtv->vb2q); if (ret < 0) { - dev_warn(dev, "Could not initialize videobuf2 queue\n"); - goto usbtv_fail; + dev_warn(usbtv->dev, "Could not initialize videobuf2 queue\n"); + return ret; } /* v4l2 structure */ usbtv->v4l2_dev.release = usbtv_release; - ret = v4l2_device_register(dev, &usbtv->v4l2_dev); + ret = v4l2_device_register(usbtv->dev, &usbtv->v4l2_dev); if (ret < 0) { - dev_warn(dev, "Could not register v4l2 device\n"); + dev_warn(usbtv->dev, "Could not register v4l2 device\n"); goto v4l2_fail; } - usb_set_intfdata(intf, usbtv); - /* Video structure */ strlcpy(usbtv->vdev.name, "usbtv", sizeof(usbtv->vdev.name)); usbtv->vdev.v4l2_dev = &usbtv->v4l2_dev; @@ -832,52 +708,31 @@ static int usbtv_probe(struct usb_interface *intf, video_set_drvdata(&usbtv->vdev, usbtv); ret = video_register_device(&usbtv->vdev, VFL_TYPE_GRABBER, -1); if (ret < 0) { - dev_warn(dev, "Could not register video device\n"); + dev_warn(usbtv->dev, "Could not register video device\n"); goto vdev_fail; } - dev_info(dev, "Fushicai USBTV007 Video Grabber\n"); return 0; vdev_fail: v4l2_device_unregister(&usbtv->v4l2_dev); v4l2_fail: vb2_queue_release(&usbtv->vb2q); -usbtv_fail: - kfree(usbtv); return ret; } -static void usbtv_disconnect(struct usb_interface *intf) +void usbtv_video_free(struct usbtv *usbtv) { - struct usbtv *usbtv = usb_get_intfdata(intf); - mutex_lock(&usbtv->vb2q_lock); mutex_lock(&usbtv->v4l2_lock); usbtv_stop(usbtv); - usb_set_intfdata(intf, NULL); video_unregister_device(&usbtv->vdev); v4l2_device_disconnect(&usbtv->v4l2_dev); - usb_put_dev(usbtv->udev); - usbtv->udev = NULL; mutex_unlock(&usbtv->v4l2_lock); mutex_unlock(&usbtv->vb2q_lock); v4l2_device_put(&usbtv->v4l2_dev); } - -MODULE_AUTHOR("Lubomir Rintel"); -MODULE_DESCRIPTION("Fushicai USBTV007 Video Grabber Driver"); -MODULE_LICENSE("Dual BSD/GPL"); - -struct usb_driver usbtv_usb_driver = { - .name = "usbtv", - .id_table = usbtv_id_table, - .probe = usbtv_probe, - .disconnect = usbtv_disconnect, -}; - -module_usb_driver(usbtv_usb_driver); diff --git a/drivers/media/usb/usbtv/usbtv.h b/drivers/media/usb/usbtv/usbtv.h new file mode 100644 index 0000000..536343d --- /dev/null +++ b/drivers/media/usb/usbtv/usbtv.h @@ -0,0 +1,98 @@ +/* + * Fushicai USBTV007 Video Grabber Driver + * + * Copyright (c) 2013 Lubomir Rintel + * All rights reserved. + * No physical hardware was harmed running Windows during the + * reverse-engineering activity + * + * Redistribution and use in source and binary forms, with or without + * modification, are permitted provided that the following conditions + * are met: + * 1. Redistributions of source code must retain the above copyright + * notice, this list of conditions, and the following disclaimer, + * without modification. + * 2. The name of the author may not be used to endorse or promote products + * derived from this software without specific prior written permission. + * + * Alternatively, this software may be distributed under the terms of the + * GNU General Public License ("GPL"). + */ + +#include <linux/slab.h> +#include <linux/usb.h> + +#include <media/v4l2-device.h> +#include <media/videobuf2-vmalloc.h> + +/* Hardware. */ +#define USBTV_VIDEO_ENDP 0x81 +#define USBTV_BASE 0xc000 +#define USBTV_REQUEST_REG 12 + +/* Number of concurrent isochronous urbs submitted. + * Higher numbers was seen to overly saturate the USB bus. */ +#define USBTV_ISOC_TRANSFERS 16 +#define USBTV_ISOC_PACKETS 8 + +#define USBTV_CHUNK_SIZE 256 +#define USBTV_CHUNK 240 + +/* Chunk header. */ +#define USBTV_MAGIC_OK(chunk) ((be32_to_cpu(chunk[0]) & 0xff000000) \ + == 0x88000000) +#define USBTV_FRAME_ID(chunk) ((be32_to_cpu(chunk[0]) & 0x00ff0000) >> 16) +#define USBTV_ODD(chunk) ((be32_to_cpu(chunk[0]) & 0x0000f000) >> 15) +#define USBTV_CHUNK_NO(chunk) (be32_to_cpu(chunk[0]) & 0x00000fff) + +#define USBTV_TV_STD (V4L2_STD_525_60 | V4L2_STD_PAL) + +/* parameters for supported TV norms */ +struct usbtv_norm_params { + v4l2_std_id norm; + int cap_width, cap_height; +}; + +/* A single videobuf2 frame buffer. */ +struct usbtv_buf { + struct vb2_buffer vb; + struct list_head list; +}; + +/* Per-device structure. */ +struct usbtv { + struct device *dev; + struct usb_device *udev; + + /* video */ + struct v4l2_device v4l2_dev; + struct video_device vdev; + struct vb2_queue vb2q; + struct mutex v4l2_lock; + struct mutex vb2q_lock; + + /* List of videobuf2 buffers protected by a lock. */ + spinlock_t buflock; + struct list_head bufs; + + /* Number of currently processed frame, useful find + * out when a new one begins. */ + u32 frame_id; + int chunks_done; + + enum { + USBTV_COMPOSITE_INPUT, + USBTV_SVIDEO_INPUT, + } input; + v4l2_std_id norm; + int width, height; + int n_chunks; + int iso_size; + unsigned int sequence; + struct urb *isoc_urbs[USBTV_ISOC_TRANSFERS]; +}; + +int usbtv_set_regs(struct usbtv *usbtv, const u16 regs[][2], int size); + +int usbtv_video_init(struct usbtv *usbtv); +void usbtv_video_free(struct usbtv *usbtv); -- 1.8.4.2

4 5

[alsa-devel] [PATCH 1/2 v2] ASoC: Add HA (HEAD acoustics) DSP codec driver template
by Stefan Roese 16 Aug '14

16 Aug '14

This codec driver template represents an I2C controlled multichannel audio codec that has many typical ASoC codec driver features like volume controls, mixer stages, mux selection, output power control, in-codec audio routings, codec bias management and DAI link configuration. This driver is based on an early version provided by Jarkko Nikula. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> Signed-off-by: Stefan Roese <sr(a)denx.de> Cc: Thorsten Eisbein <thorsten.eisbein(a)head-acoustics.de> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Mark Brown <broonie(a)kernel.org> --- v2: - Added/changed copyright line - Changed authorship (as suggested by Jarkko the original author) - Added Thorsten as maintainer - Remove ha_dsp_hw_params() and ha_dsp_set_dai_fmt() as its not used (only needed for CODEC as clock master which is currently not suported). - Removed some unneeded include files - "const char *const foo" used instead of "const char *foo" - SOC_MIXER_ARRAY() helper macro used - Removed ha_dsp_set_bias_level() and use default implementation - Use codec->dev instead of codec->dev->parent in dev_get_regmap() - Added CODEC reset to probe - Remove "ret" in ha_dsp_i2c_probe() sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ha-dsp.c | 333 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ha-dsp.h | 50 +++++++ 4 files changed, 389 insertions(+) create mode 100644 sound/soc/codecs/ha-dsp.c create mode 100644 sound/soc/codecs/ha-dsp.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0e8401..f357988 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -51,6 +51,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C select SND_SOC_BT_SCO + select SND_SOC_HA_DSP if I2C select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -343,6 +344,9 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HA_DSP + tristate + config SND_SOC_HDMI_CODEC tristate "HDMI stub CODEC" diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 3c4d275..f296bec 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -39,6 +39,7 @@ snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o +snd-soc-ha-dsp-objs := ha-dsp.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o snd-soc-l3-objs := l3.o @@ -190,6 +191,7 @@ obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o +obj-$(CONFIG_SND_SOC_HA_DSP) += snd-soc-ha-dsp.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ha-dsp.c b/sound/soc/codecs/ha-dsp.c new file mode 100644 index 0000000..5a3c7ef --- /dev/null +++ b/sound/soc/codecs/ha-dsp.c @@ -0,0 +1,333 @@ +/* + * ha-dsp.c -- HA DSP ALSA SoC Audio driver + * + * Copyright 2011-2014 HEAD acoustics GmbH + * + * Authors: + * Jarkko Nikula <jarkko.nikula(a)bitmer.com> + * Stefan Roese <sr(a)denx.de> + * Thorsten Eisbein <thorsten.eisbein(a)head-acoustics.de> + * + * 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. + * + * Maintainer: Thorsten Eisbein <thorsten.eisbein(a)head-acoustics.de> + */ + +#include <linux/module.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "ha-dsp.h" + +/* Reset default register values for soc-cache */ +static const struct reg_default ha_dsp_reg_defaults[] = { + { 0x00, 0x00 }, + { 0x01, 0x55 }, + { 0x02, 0x55 }, + { 0x03, 0x00 }, + { 0x04, 0x00 }, + { 0x05, 0x00 }, + { 0x06, 0x00 }, + { 0x07, 0x00 }, + { 0x08, 0x02 }, + { 0x09, 0x02 }, + { 0x0a, 0x02 }, + { 0x0b, 0x02 }, + { 0x0c, 0x02 }, + { 0x0d, 0x02 }, + { 0x0e, 0x02 }, + { 0x0f, 0x02 }, +}; + +/* DSP mode selection */ +static const char *const ha_dsp_mode_texts[] = {"Mode 1", "Mode 2"}; +static SOC_ENUM_SINGLE_DECL(ha_dsp_mode_enum, HA_DSP_CTRL, 0, + ha_dsp_mode_texts); + +/* Monitor output mux selection */ +static const char *const ha_dsp_monitor_texts[] = {"Off", "ADC", "DAC"}; +static SOC_ENUM_SINGLE_DECL(ha_dsp_monitor_enum, HA_DSP_CTRL, 1, + ha_dsp_monitor_texts); + +static const struct snd_kcontrol_new ha_dsp_monitor_control = + SOC_DAPM_ENUM("Route", ha_dsp_monitor_enum); + +/* Output mixers */ +static const struct snd_kcontrol_new ha_dsp_out1_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT1_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT1_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out2_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT2_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT2_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out3_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT3_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT3_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out4_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT4_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT4_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out5_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT5_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT5_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out6_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT6_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT6_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out7_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT7_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT1_CTRL, 2, 1, 0), +}; +static const struct snd_kcontrol_new ha_dsp_out8_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Switch", HA_DSP_OUT8_CTRL, 1, 1, 0), + SOC_DAPM_SINGLE("IN Bypass Switch", HA_DSP_OUT8_CTRL, 2, 1, 0), +}; + +static const struct snd_kcontrol_new ha_dsp_snd_controls[] = { + SOC_SINGLE("ADC Capture Volume", + HA_DSP_ADC_VOL, 0, 0x7f, 0), + SOC_SINGLE("ADC Capture Switch", + HA_DSP_ADC_VOL, 7, 0x01, 1), + + SOC_SINGLE("PCM Playback Volume", + HA_DSP_DAC_VOL, 0, 0x7f, 0), + SOC_SINGLE("PCM Playback Switch", + HA_DSP_DAC_VOL, 7, 0x01, 1), + + SOC_ENUM("DSP Mode", ha_dsp_mode_enum), +}; + +static const struct snd_soc_dapm_widget ha_dsp_widgets[] = { + SND_SOC_DAPM_ADC("ADC", "Capture", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_DAC("DAC", "Playback", SND_SOC_NOPM, 0, 0), + + SOC_MIXER_ARRAY("OUT1 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out1_mixer_controls), + SOC_MIXER_ARRAY("OUT2 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out2_mixer_controls), + SOC_MIXER_ARRAY("OUT3 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out3_mixer_controls), + SOC_MIXER_ARRAY("OUT4 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out4_mixer_controls), + SOC_MIXER_ARRAY("OUT5 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out5_mixer_controls), + SOC_MIXER_ARRAY("OUT6 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out6_mixer_controls), + SOC_MIXER_ARRAY("OUT7 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out7_mixer_controls), + SOC_MIXER_ARRAY("OUT8 Mixer", SND_SOC_NOPM, 0, 0, + ha_dsp_out8_mixer_controls), + + SND_SOC_DAPM_PGA("OUT1 PGA", HA_DSP_OUT1_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT2 PGA", HA_DSP_OUT2_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT3 PGA", HA_DSP_OUT3_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT4 PGA", HA_DSP_OUT4_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT5 PGA", HA_DSP_OUT5_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT6 PGA", HA_DSP_OUT6_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT7 PGA", HA_DSP_OUT7_CTRL, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("OUT8 PGA", HA_DSP_OUT8_CTRL, 0, 0, NULL, 0), + + SND_SOC_DAPM_MUX("Monitor Out Mux", SND_SOC_NOPM, 0, 0, + &ha_dsp_monitor_control), + + /* Input pins */ + SND_SOC_DAPM_INPUT("IN1"), + SND_SOC_DAPM_INPUT("IN2"), + SND_SOC_DAPM_INPUT("IN3"), + SND_SOC_DAPM_INPUT("IN4"), + SND_SOC_DAPM_INPUT("IN5"), + SND_SOC_DAPM_INPUT("IN6"), + SND_SOC_DAPM_INPUT("IN7"), + SND_SOC_DAPM_INPUT("IN8"), + + /* Output pins */ + SND_SOC_DAPM_OUTPUT("OUT1"), + SND_SOC_DAPM_OUTPUT("OUT2"), + SND_SOC_DAPM_OUTPUT("OUT3"), + SND_SOC_DAPM_OUTPUT("OUT4"), + SND_SOC_DAPM_OUTPUT("OUT5"), + SND_SOC_DAPM_OUTPUT("OUT6"), + SND_SOC_DAPM_OUTPUT("OUT7"), + SND_SOC_DAPM_OUTPUT("OUT8"), + SND_SOC_DAPM_OUTPUT("MONITOR"), +}; + +static const struct snd_soc_dapm_route ha_dsp_routes[] = { + /* Inputs to ADC */ + {"ADC", NULL, "IN1"}, + {"ADC", NULL, "IN2"}, + {"ADC", NULL, "IN3"}, + {"ADC", NULL, "IN4"}, + {"ADC", NULL, "IN5"}, + {"ADC", NULL, "IN6"}, + {"ADC", NULL, "IN7"}, + {"ADC", NULL, "IN8"}, + + /* DAC and input bypass paths to outputs */ + {"OUT1 Mixer", "DAC Switch", "DAC"}, + {"OUT1 Mixer", "IN Bypass Switch", "IN1"}, + {"OUT1 PGA", NULL, "OUT1 Mixer"}, + {"OUT1", NULL, "OUT1 PGA"}, + + {"OUT2 Mixer", "DAC Switch", "DAC"}, + {"OUT2 Mixer", "IN Bypass Switch", "IN2"}, + {"OUT2 PGA", NULL, "OUT2 Mixer"}, + {"OUT2", NULL, "OUT2 PGA"}, + + {"OUT3 Mixer", "DAC Switch", "DAC"}, + {"OUT3 Mixer", "IN Bypass Switch", "IN3"}, + {"OUT3 PGA", NULL, "OUT3 Mixer"}, + {"OUT3", NULL, "OUT3 PGA"}, + + {"OUT4 Mixer", "DAC Switch", "DAC"}, + {"OUT4 Mixer", "IN Bypass Switch", "IN4"}, + {"OUT4 PGA", NULL, "OUT4 Mixer"}, + {"OUT4", NULL, "OUT4 PGA"}, + + {"OUT5 Mixer", "DAC Switch", "DAC"}, + {"OUT5 Mixer", "IN Bypass Switch", "IN5"}, + {"OUT5 PGA", NULL, "OUT5 Mixer"}, + {"OUT5", NULL, "OUT5 PGA"}, + + {"OUT6 Mixer", "DAC Switch", "DAC"}, + {"OUT6 Mixer", "IN Bypass Switch", "IN6"}, + {"OUT6 PGA", NULL, "OUT6 Mixer"}, + {"OUT6", NULL, "OUT6 PGA"}, + + {"OUT7 Mixer", "DAC Switch", "DAC"}, + {"OUT7 Mixer", "IN Bypass Switch", "IN7"}, + {"OUT7 PGA", NULL, "OUT7 Mixer"}, + {"OUT7", NULL, "OUT7 PGA"}, + + {"OUT8 Mixer", "DAC Switch", "DAC"}, + {"OUT8 Mixer", "IN Bypass Switch", "IN8"}, + {"OUT8 PGA", NULL, "OUT8 Mixer"}, + {"OUT8", NULL, "OUT8 PGA"}, + + /* Monitor output */ + {"Monitor Out Mux", "ADC", "ADC"}, + {"Monitor Out Mux", "DAC", "DAC"}, + {"MONITOR", NULL, "Monitor Out Mux"}, +}; + +static struct snd_soc_dai_driver ha_dsp_dai = { + .name = "ha-dsp-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_96000, + /* We use only 32 Bits for Audio */ + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 16, + .rates = SNDRV_PCM_RATE_8000_96000, + /* We use only 32 Bits for Audio */ + .formats = SNDRV_PCM_FMTBIT_S32_LE, + }, +}; + +static int ha_dsp_probe(struct snd_soc_codec *codec) +{ + int ret; + + codec->control_data = dev_get_regmap(codec->dev, NULL); + ret = snd_soc_codec_set_cache_io(codec, codec->control_data); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + snd_soc_write(codec, HA_DSP_CTRL, HA_DSP_SW_RESET); + + return 0; +} + +static int ha_dsp_remove(struct snd_soc_codec *codec) +{ + snd_soc_write(codec, HA_DSP_CTRL, HA_DSP_SW_RESET); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ha_dsp = { + .probe = ha_dsp_probe, + .remove = ha_dsp_remove, + + .controls = ha_dsp_snd_controls, + .num_controls = ARRAY_SIZE(ha_dsp_snd_controls), + .dapm_widgets = ha_dsp_widgets, + .num_dapm_widgets = ARRAY_SIZE(ha_dsp_widgets), + .dapm_routes = ha_dsp_routes, + .num_dapm_routes = ARRAY_SIZE(ha_dsp_routes), +}; + +static const struct regmap_config ha_dsp_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = 0x0f, + .reg_defaults = ha_dsp_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ha_dsp_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static int ha_dsp_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(client, &ha_dsp_regmap); + if (IS_ERR(regmap)) { + dev_err(&client->dev, "Failed to create regmap: %ld\n", + PTR_ERR(regmap)); + return PTR_ERR(regmap); + } + + return snd_soc_register_codec(&client->dev, &soc_codec_dev_ha_dsp, + &ha_dsp_dai, 1); +} + +static int ha_dsp_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + + return 0; +} + +/* + * This name/ID is neded to match the DT node for the codec + */ +static const struct i2c_device_id ha_dsp_i2c_id[] = { + { "ha-dsp-audio", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ha_dsp_i2c_id); + +static struct i2c_driver ha_dsp_i2c_driver = { + .driver = { + .name = "ha-dsp-codec", + .owner = THIS_MODULE, + }, + .probe = ha_dsp_i2c_probe, + .remove = ha_dsp_i2c_remove, + .id_table = ha_dsp_i2c_id, +}; + +module_i2c_driver(ha_dsp_i2c_driver); + +MODULE_DESCRIPTION("ASoC HA DSP driver"); +MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula(a)bitmer.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ha-dsp.h b/sound/soc/codecs/ha-dsp.h new file mode 100644 index 0000000..6622f8a --- /dev/null +++ b/sound/soc/codecs/ha-dsp.h @@ -0,0 +1,50 @@ +/* + * ha-dsp.h -- HA DSP ALSA SoC Audio driver + * + * Copyright 2011-2014 HEAD acoustics GmbH + * + * Author: Jarkko Nikula <jhnikula(a)gmail.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. + */ + +#ifndef __HA_DSP_H__ +#define __HA_DSP_H__ + +/* Registers */ + +/* + * Bit 2-1: Monitor output selection: Off, ADC, DAC + * Bit 0: DSP Mode + */ +#define HA_DSP_CTRL 0x00 + +/* + * Bit 7: Mute + * Bit 6-0: Volume + */ +#define HA_DSP_DAC_VOL 0x01 +#define HA_DSP_ADC_VOL 0x02 + +/* + * Bit 2: INx Bypass to OUTx Switch + * Bit 1: DAC to OUTx switch + * Bit 0: Output power + */ +#define HA_DSP_OUT1_CTRL 0x08 +#define HA_DSP_OUT2_CTRL 0x09 +#define HA_DSP_OUT3_CTRL 0x0a +#define HA_DSP_OUT4_CTRL 0x0b +#define HA_DSP_OUT5_CTRL 0x0c +#define HA_DSP_OUT6_CTRL 0x0d +#define HA_DSP_OUT7_CTRL 0x0e +#define HA_DSP_OUT8_CTRL 0x0f + +/* Register bits and values */ + +/* HA_DSP_CTRL */ +#define HA_DSP_SW_RESET 0xff + +#endif -- 1.9.2

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[alsa-devel] [PATCH] ALSA: hda - initialize audio InfoFrame to be all zero
by mengdong.lin＠intel.com 23 Jul '14

23 Jul '14

From: Mengdong Lin <mengdong.lin(a)intel.com> This patch initialized the local audio InfoFrame variable 'ai' to be all zero, thus the data bytes will indicate "Refer to Stream Header" by default. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 5ef9503..3ab7063 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1062,6 +1062,7 @@ static void hdmi_pin_setup_infoframe(struct hda_codec *codec, { union audio_infoframe ai; + memset(&ai, 0, sizeof(ai)); if (conn_type == 0) { /* HDMI */ struct hdmi_audio_infoframe *hdmi_ai = &ai.hdmi; -- 1.8.1.2

4 5

[alsa-devel] wm8974 and sam9g45 atmel
by Michele Da Rold 11 Jul '14

11 Jul '14

Hello everybody, I'm starting a new development with a Atmel SAM9G45 and a WM8974. I'm using kernel 2.6.33.2 I have connected the SSC1 pin TD,TF,TK and the RD pin, so I want to use WM8974 like a slave. I wrote a interface driver based on sam9263_wm8731.c The driver seem to load ok, I recognize the codec but when I start a mp3 playing (I use mpg123) the playback not start. Putting some dedug messagge the driver not call the function "set_bias_level", and call the mute function with mute=0. Testing the MCLK signal I not see any clock on pin because I never call clk_enable!! There is a way to identify the problem? Someone could help me? Thank you in advance Michele Da Rold I put the code of driver (don't care about printk I used it for debug purpose!!) /* * sam9g45_wm8974 -- SoC audio for AT91SAM9G20-based * ATMEL AT91SAM9G20ek board. * * Copyright (C) 2005 SAN People * Copyright (C) 2008 Atmel * * Authors: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> * * Based on sam9263_wm8731.c. * Frank Mandarino <fmandarino(a)endrelia.com> * Copyright 2006 Endrelia Technologies Inc. * Based on corgi.c by: * Copyright 2005 Wolfson Microelectronics PLC. * Copyright 2005 Openedhand Ltd. * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/kernel.h> #include <linux/clk.h> #include <linux/timer.h> #include <linux/interrupt.h> #include <linux/platform_device.h> #include <linux/i2c.h> #include <linux/atmel-ssc.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <asm/mach-types.h> #include <mach/hardware.h> #include <mach/gpio.h> #include "../codecs/wm8974.h" #include "atmel-pcm.h" #include "atmel_ssc_dai.h" #define MCLK_RATE 12000000 /* * As shipped the board does not have inputs. However, it is relatively * straightforward to modify the board to hook them up so support is left * in the driver. */ #undef ENABLE_MIC_INPUT #define ENABLE_MIC_INPUT 1 static struct clk *mclk; static int at91sam9g45ek_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->dai->codec_dai; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_hw_params() called\n"); /* set codec DAI configuration */ ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS/*SND_SOC_DAIFMT_CBM_CFM*/); if (ret < 0) return ret; return 0; } static struct snd_soc_ops at91sam9g45ek_ops = { .hw_params = at91sam9g45ek_hw_params, }; static int at91sam9g45ek_set_bias_level(struct snd_soc_card *card, enum snd_soc_bias_level level) { static int mclk_on; int ret = 0; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45ek_set_bias_level() called %d\n",level); switch (level) { case SND_SOC_BIAS_ON: case SND_SOC_BIAS_PREPARE: if (!mclk_on) ret = clk_enable(mclk); if (ret == 0) mclk_on = 1; break; case SND_SOC_BIAS_OFF: case SND_SOC_BIAS_STANDBY: if (mclk_on) clk_disable(mclk); mclk_on = 0; break; } return ret; } #if 0 static const struct snd_soc_dapm_widget at91sam9g45ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), }; static const struct snd_soc_dapm_route intercon[] = { /* Speaker output mixer */ {"Speaker Mixer", "PCM Playback Switch", "DAC"}, {"Speaker Mixer", "Aux Playback Switch", "Aux Input"}, {"Speaker Mixer", "Line Bypass Switch", "Boost Mixer"}, /* speaker connected to SPKOUTP */ {"SpkN Out", NULL, "Speaker Mixer"}, {"SpkP Out", NULL, "Speaker Mixer"}, {"SPKOUTN", NULL, "SpkN Out"}, {"SPKOUTP", NULL, "SpkP Out"}, /* Input PGA */ {"Input PGA", "Aux Switch", "Aux Input"}, {"Input PGA", "MicN Switch", "MICN"}, {"Input PGA", "MicP Switch", "MICP"}, }; /* * Logic for a wm974 as connected on a at91sam9g45ek board. */ static int at91sam9g45ek_wm8974_init(struct snd_soc_codec *codec) { struct snd_soc_dai *codec_dai = &codec->dai[0]; int ret; printk(KERN_DEBUG "at91sam9g45_wm8974 " ": at91sam9g45_wm8974_init() called\n"); /* ret = snd_soc_dai_set_sysclk(codec_dai, 0, MCLK_RATE, SND_SOC_CLOCK_IN); if (ret < 0) { printk(KERN_ERR "Failed to set WM8974 SYSCLK: %d\n", ret); return ret; } */ /* Add specific widgets */ snd_soc_dapm_new_controls(codec, at91sam9g45ek_dapm_widgets, ARRAY_SIZE(at91sam9g45ek_dapm_widgets)); /* Set up specific audio path interconnects */ snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); /* not connected */ snd_soc_dapm_nc_pin(codec, "MONOOUT"); snd_soc_dapm_nc_pin(codec, "AUX"); #ifdef ENABLE_MIC_INPUT snd_soc_dapm_enable_pin(codec, "Int Mic"); #else snd_soc_dapm_nc_pin(codec, "Int Mic"); #endif /* always connected */ snd_soc_dapm_enable_pin(codec, "Ext Spk"); snd_soc_dapm_sync(codec); return 0; } #endif static struct snd_soc_dai_link at91sam9g45ek_dai = { .name = "WM8974", .stream_name = "WM8974 PCM", .cpu_dai = &atmel_ssc_dai[1], .codec_dai = &wm8974_dai, #if 0 .init = at91sam9g45ek_wm8974_init, #endif .ops = &at91sam9g45ek_ops, }; static struct snd_soc_card snd_soc_at91sam9g45ek = { .name = "AT91SAMG45", .platform = &atmel_soc_platform, .dai_link = &at91sam9g45ek_dai, .num_links = 1, .set_bias_level = at91sam9g45ek_set_bias_level, }; static struct snd_soc_device at91sam9g45ek_snd_devdata = { .card = &snd_soc_at91sam9g45ek, .codec_dev = &soc_codec_dev_wm8974, }; static struct platform_device *at91sam9g45ek_snd_device; static int __init at91sam9g45ek_init(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc = NULL; struct clk *main; int ret; at91_set_A_periph(AT91_PIN_PD27, 0); /* * Codec MCLK is supplied by PCK1 - set it up. */ mclk = clk_get(NULL, "pck1"); if (IS_ERR(mclk)) { printk(KERN_ERR "ASoC: Failed to get MCLK\n"); ret = PTR_ERR(mclk); goto err; } main = clk_get(NULL, "main"); if (IS_ERR(main)) { printk(KERN_ERR "ASoC: Failed to get PLLB\n"); ret = PTR_ERR(mclk); goto err_mclk; } ret = clk_set_parent(mclk, main); clk_put(main); if (ret != 0) { printk(KERN_ERR "ASoC: Failed to set MCLK parent\n"); goto err_mclk; } clk_set_rate(mclk, MCLK_RATE); /* * Request SSC device */ ssc = ssc_request(1); if (IS_ERR(ssc)) { printk(KERN_ERR "ASoC: Failed to request SSC 1\n"); ret = PTR_ERR(ssc); ssc = NULL; goto err_ssc; } ssc_p->ssc = ssc; at91sam9g45ek_snd_device = platform_device_alloc("soc-audio", -1); if (!at91sam9g45ek_snd_device) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); ret = -ENOMEM; } platform_set_drvdata(at91sam9g45ek_snd_device, &at91sam9g45ek_snd_devdata); at91sam9g45ek_snd_devdata.dev = &at91sam9g45ek_snd_device->dev; ret = platform_device_add(at91sam9g45ek_snd_device); if (ret) { printk(KERN_ERR "ASoC: Platform device allocation failed\n"); platform_device_put(at91sam9g45ek_snd_device); } return ret; err_ssc: ssc_free(ssc); ssc_p->ssc = NULL; err_mclk: clk_put(mclk); mclk = NULL; err: return ret; } static void __exit at91sam9g45ek_exit(void) { struct atmel_ssc_info *ssc_p = at91sam9g45ek_dai.cpu_dai->private_data; struct ssc_device *ssc; if (ssc_p != NULL) { ssc = ssc_p->ssc; if (ssc != NULL) ssc_free(ssc); ssc_p->ssc = NULL; } platform_device_unregister(at91sam9g45ek_snd_device); at91sam9g45ek_snd_device = NULL; clk_put(mclk); mclk = NULL; } module_init(at91sam9g45ek_init); module_exit(at91sam9g45ek_exit); /* Module information */ MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou(a)atmel.com>"); MODULE_DESCRIPTION("ALSA SoC AT91SAM9G45_WM8974"); MODULE_LICENSE("GPL");

3 8

[alsa-devel] [PATCH 1/2] doc: dt bindings: Document Odroid X2/U3 audio subsystem bindings
by Sylwester Nawrocki 04 Jul '14

04 Jul '14

Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- .../bindings/sound/samsung,odroidx2-max98090.txt | 32 ++++++++++++++++++++ 1 file changed, 32 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt diff --git a/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt new file mode 100644 index 0000000..b37e79a --- /dev/null +++ b/Documentation/devicetree/bindings/sound/samsung,odroidx2-max98090.txt @@ -0,0 +1,32 @@ +Odroid X2/U3 audio complex + +Required properties: +- compatible : "samsung,odroidx2-audio" +- samsung,i2s-controller : phandle to the I2S controller +- samsung,audio-codec : phandle to the audio codec + +Example: + + i2s0: i2s@03830000 { + ... + clocks = <&clock_audss EXYNOS_I2S_BUS>, + <&clock_audss EXYNOS_DOUT_AUD_BUS>; + clock-names = "iis", "i2s_opclk0"; + status = "okay"; + }; + + i2c@13870000 { + ... + max98090: max98090@10 { + compatible = "maxim,max98090"; + reg = <0x10>; + interrupt-parent = <&gpx0>; + interrupts = <1 0>; + }; + }; + + sound { + compatible = "samsung,odroidx2-audio"; + samsung,i2s-controller = <&i2s0>; + samsung,audio-codec = <&max98090>; + }; -- 1.7.9.5

3 10

[alsa-devel] [PATCHv2 0/9] DT support for N900 soundcard (rx51-audio)
by Sebastian Reichel 30 Jun '14

30 Jun '14

Hi, This patchset adds DT support in rx51-audio. I tested it on my Nokia N900 and was able to play a wav file using aplay. I have not tested the whole functionality, but output via speakers, headphones and the related enable controls seem to work. Changes since PATCHv1 [0]: * drop "ASoC: tlv320aic3x: fix shared reset pin for DT" (already applied) * drop "ASoC: RX-51: Convert to table based DAPM setup" (similar patch was applied to -next) * remove useless "card->dev = NULL;" from first patch * split patches from into smaller chunks * introduce new patch, which fixes a checkpatch warning * use gpiod API, which reduces the DT specific bits of the driver * rebase to newer -next checkout [0] https://lkml.org/lkml/2014/4/5/92 -- Sebastian Sebastian Reichel (9): ASoC: omap: rx51: Use static const char * const arrays ASoC: omap: rx51: Add module alias ASoC: omap: rx51: Use devm_snd_soc_register_card ASoC: Allow Aux Codecs to be specified using DT ASoC: omap: rx51: omap_mcbsp_st_add_controls: add id parameter ASoC: omap: rx51: get GPIO numbers via gpiod API ASoC: omap: rx51: Add some error messages ASoC: omap: rx51: Add DT support DTS: OMAP3-N900: Add sound support .../devicetree/bindings/sound/nokia,rx51.txt | 27 +++ arch/arm/boot/dts/omap3-n900.dts | 17 ++ include/sound/soc.h | 13 +- sound/soc/omap/omap-mcbsp.c | 5 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/rx51.c | 226 +++++++++++++++------ sound/soc/soc-core.c | 68 ++++--- 7 files changed, 271 insertions(+), 87 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/nokia,rx51.txt -- 1.9.2

4 24

[alsa-devel] [RFT v3 0/3] ASoC: core: Add support for DAI multicodec
by Benoit Cousson 23 Jun '14

23 Jun '14

Hi Mark, Liam and Lars, Sorry for the delay. Here is the third version of the multicodec series. I changed the way to store the multiple codec_dai as suggested by Lars and added the missing iterations in all the places pointed by Lars in his comments. Please note that I have a very simple setup based on BBB to check if the series is working, so it will be good to test it on various platforms to ensure this is not generating any regression. Hence this version is still RFT. The series is based on asoc/for-next (v3.15-rc2). Test and comments are welcome. Thanks, Benoit v1: http://comments.gmane.org/gmane.linux.alsa.devel/120532 v2: http://comments.gmane.org/gmane.linux.alsa.devel/121024 - Split the first version in several patches. - Remove the TDM fixup v3: - Fixed all the places that were not iterating over all the codecs. - Change the structure and split it. Benoit Cousson (2): ASoC: core: Add one dai_get_widget helper instead of two rtd based ones ASoC: core: Add support for DAI multicodec Misael Lopez Cruz (1): ASoC: core: Add helpers for dai link and aux dev init include/sound/soc-dai.h | 5 + include/sound/soc.h | 13 ++ sound/soc/soc-compress.c | 83 +++++--- sound/soc/soc-core.c | 401 +++++++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 ++++--- sound/soc/soc-pcm.c | 529 +++++++++++++++++++++++++++++++---------------- 6 files changed, 763 insertions(+), 339 deletions(-) -- 1.8.3.2

4 16

[alsa-devel] [PATCH v3] ASoC: davinci-mcasp: set up channel status bits for S/PDIF mode
by Daniel Mack 18 Jun '14

18 Jun '14

In DIT (S/PDIF) mode, program the transmitted user bits to reflect the configured sample rate, along with some other details. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- v3 just changes variable names. sound/soc/davinci/davinci-mcasp.c | 49 +++++++++++++++++++++++++++++++++++++-- 1 file changed, 47 insertions(+), 2 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 4500cf0..a21a7d4 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -27,6 +27,7 @@ #include <linux/of_platform.h> #include <linux/of_device.h> +#include <sound/asoundef.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/pcm_params.h> @@ -576,8 +577,12 @@ static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream) } /* S/PDIF */ -static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) +static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp, + unsigned int rate) { + u32 cs_value = 0; + u8 *cs_bytes = (u8*) &cs_value; + /* Set the TX format : 24 bit right rotation, 32 bit slot, Pad 0 and LSB first */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(6) | TXSSZ(15)); @@ -599,6 +604,46 @@ static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp) /* Enable the DIT */ mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN); + /* Set S/PDIF channel status bits */ + cs_bytes[0] = IEC958_AES0_CON_NOT_COPYRIGHT; + cs_bytes[1] = IEC958_AES1_CON_PCM_CODER; + + switch (rate) { + case 22050: + cs_bytes[3] |= IEC958_AES3_CON_FS_22050; + break; + case 24000: + cs_bytes[3] |= IEC958_AES3_CON_FS_24000; + break; + case 32000: + cs_bytes[3] |= IEC958_AES3_CON_FS_32000; + break; + case 44100: + cs_bytes[3] |= IEC958_AES3_CON_FS_44100; + break; + case 48000: + cs_bytes[3] |= IEC958_AES3_CON_FS_48000; + break; + case 88200: + cs_bytes[3] |= IEC958_AES3_CON_FS_88200; + break; + case 96000: + cs_bytes[3] |= IEC958_AES3_CON_FS_96000; + break; + case 176400: + cs_bytes[3] |= IEC958_AES3_CON_FS_176400; + break; + case 192000: + cs_bytes[3] |= IEC958_AES3_CON_FS_192000; + break; + default: + printk(KERN_WARNING "unsupported sampling rate: %d\n", rate); + return -EINVAL; + } + + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, cs_value); + mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, cs_value); + return 0; } @@ -634,7 +679,7 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, return ret; if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) - ret = mcasp_dit_hw_param(mcasp); + ret = mcasp_dit_hw_param(mcasp, params_rate(params)); else ret = mcasp_i2s_hw_param(mcasp, substream->stream); -- 1.8.5.3

4 4

[alsa-devel] [PATCH 1/2] ASoC: ak5386: Add regulators to documentation and fix sparse warning
by Daniel Mack 18 Jun '14

18 Jun '14

Document the newly added regulators to the DT binding document. Also, "static const char const *x" is not identical to "static const char * const x", which sparse now complains about. Fix it. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- Documentation/devicetree/bindings/sound/ak5386.txt | 4 ++++ sound/soc/codecs/ak5386.c | 2 +- 2 files changed, 5 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt index dc3914f..ec3df3a 100644 --- a/Documentation/devicetree/bindings/sound/ak5386.txt +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -10,10 +10,14 @@ Optional properties: - reset-gpio : a GPIO spec for the reset/power down pin. If specified, it will be deasserted at probe time. + - va-supply : a regulator spec, providing 5.0V + - vd-supply : a regulator spec, providing 3.3V Example: spdif: ak5386@0 { compatible = "asahi-kasei,ak5386"; reset-gpio = <&gpio0 23>; + va-supply = <&vdd_5v0_reg>; + vd-supply = <&vdd_3v3_reg>; }; diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c index a30be5c..8107a1c 100644 --- a/sound/soc/codecs/ak5386.c +++ b/sound/soc/codecs/ak5386.c @@ -19,7 +19,7 @@ #include <sound/pcm.h> #include <sound/initval.h> -static const char const *supply_names[] = { +static const char * const supply_names[] = { "va", "vd" }; -- 1.8.5.3

3 5

[alsa-devel] [RFC][PATCH v1] ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series
by Nicolin Chen 13 Jun '14

13 Jun '14

From: Nicolin Chen <b42378(a)freescale.com> This patch adds three main functions for DAI master mode: set_dai_fmt(), set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock accordingly. After appending this patch, the fsl_ssi driver on i.MX series has the ability to derive LRCLK and BCLK from baud clock source so as to support some audio Codecs which can only be used in slave mode. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 280 +++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_ssi.h | 2 + 2 files changed, 278 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f9f4569..b2ebaf8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include <linux/device.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> @@ -139,7 +140,10 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + bool baudclk_locked; u8 i2s_mode; + spinlock_t baudclk_lock; + struct clk *baudclk; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -434,13 +438,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + unsigned long flags; /* First, we only do fsl_ssi_setup() when SSI is going to be active. * Second, fsl_ssi_setup was already called by ac97_init earlier if * the driver is in ac97 mode. */ - if (!dai->active && !ssi_private->imx_ac97) + if (!dai->active && !ssi_private->imx_ac97) { fsl_ssi_setup(ssi_private); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } return 0; } @@ -502,6 +511,243 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } /** + * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format. + */ +static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 strcr = 0, stcr, srcr, scr, mask; + + scr = read_ssi(&ssi->scr) & ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK); + scr |= CCSR_SSI_SCR_NET; + + mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR | + CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL | + CCSR_SSI_STCR_TEFS; + stcr = read_ssi(&ssi->stcr) & ~mask; + srcr = read_ssi(&ssi->srcr) & ~mask; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + break; + default: + return -EINVAL; + } + scr |= ssi_private->i2s_mode; + + /* Data on rising edge of bclk, frame low, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + strcr ^= CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + strcr ^= CCSR_SSI_STCR_TFSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + strcr ^= CCSR_SSI_STCR_TSCKP; + strcr ^= CCSR_SSI_STCR_TFSI; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR; + scr |= CCSR_SSI_SCR_SYS_CLK_EN; + break; + case SND_SOC_DAIFMT_CBM_CFM: + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + stcr |= strcr; + srcr |= strcr; + + if (ssi_private->cpu_dai_drv.symmetric_rates) { + /* Need to clear RXDIR when using SYNC mode */ + srcr &= ~CCSR_SSI_SRCR_RXDIR; + scr |= CCSR_SSI_SCR_SYN; + } + + write_ssi(stcr, &ssi->stcr); + write_ssi(srcr, &ssi->srcr); + write_ssi(scr, &ssi->scr); + + return 0; +} + +/** + * fsl_ssi_set_dai_sysclk - configure Digital Audio Interface bit clock + * + * Note: This function can be only called when using SSI as DAI master + * + * Quick instruction for parameters: + * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels + * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + */ +static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + unsigned long flags, clkrate, baudrate, tmprate; + u64 sub, savesub = 100000; + + /* Don't apply it to any non-baudclk circumstance */ + if (IS_ERR(ssi_private->baudclk)) + return -EINVAL; + + /* It should be already enough to divide clock by setting pm alone */ + psr = 0; + div2 = 0; + + factor = (div2 + 1) * (7 * psr + 1) * 2; + + for (i = 0; i < 255; i++) { + /* The bclk rate must be smaller than 1/5 sysclk rate */ + if (factor * (i + 1) < 5) + continue; + + tmprate = freq * factor * (i + 2); + clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + + do_div(clkrate, factor); + afreq = (u32)clkrate / (i + 1); + + if (freq == afreq) + sub = 0; + else if (freq / afreq == 1) + sub = freq - afreq; + else if (afreq / freq == 1) + sub = afreq - freq; + else + continue; + + /* Calculate the fraction */ + sub *= 100000; + do_div(sub, freq); + + if (sub < savesub) { + baudrate = tmprate; + savesub = sub; + pm = i; + } + + /* We are lucky */ + if (savesub == 0) + break; + } + + /* No proper pm found if it is still remaining the initial value */ + if (pm == 999) { + dev_err(cpu_dai->dev, "failed to handle the required sysclk\n"); + return -EINVAL; + } + + stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) | + (psr ? CCSR_SSI_SxCCR_PSR : 0); + mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 | CCSR_SSI_SxCCR_PSR; + + if (dir == SND_SOC_CLOCK_OUT || synchronous) + write_ssi_mask(&ssi->stccr, mask, stccr); + else + write_ssi_mask(&ssi->srccr, mask, stccr); + + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + if (!ssi_private->baudclk_locked) { + ret = clk_set_rate(ssi_private->baudclk, baudrate); + if (ret) { + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + dev_err(cpu_dai->dev, "failed to set baudclk rate\n"); + return -EINVAL; + } + ssi_private->baudclk_locked = true; + } + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + + return 0; +} + +/** + * fsl_ssi_set_dai_tdm_slot - set TDM slot number + * + * Note: This function can be only called when using SSI as DAI master + */ +static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 val; + + /* The slot number should be >= 2 if using Network mode or I2S mode */ + val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); + if (val && slots < 2) { + dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); + return -EINVAL; + } + + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + + /* The register SxMSKs needs SSI to provide essential clock due to + * hardware design. So we here temporarily enable SSI to set them. + */ + val = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); + + write_ssi(tx_mask, &ssi->stmsk); + write_ssi(rx_mask, &ssi->srmsk); + + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, val); + + return 0; +} + +/** * fsl_ssi_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA @@ -517,6 +763,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sier_bits; + unsigned long flags; /* * Enable only the interrupts and DMA requests @@ -557,8 +804,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & - (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) { write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } break; default: @@ -585,6 +836,9 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, + .set_fmt = fsl_ssi_set_dai_fmt, + .set_sysclk = fsl_ssi_set_dai_sysclk, + .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; @@ -897,6 +1151,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + ssi_private->baudclk_locked = false; + spin_lock_init(&ssi_private->baudclk_lock); + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; @@ -914,6 +1171,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) goto error_irqmap; } + /* For those SLAVE implementations, we ingore non-baudclk cases + * and, instead, abandon MASTER mode that needs baud clock. + */ + ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); + if (IS_ERR(ssi_private->baudclk)) + dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + else + clk_prepare_enable(ssi_private->baudclk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -1059,8 +1325,11 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } error_irqmap: irq_dispose_mapping(ssi_private->irq); @@ -1076,8 +1345,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } irq_dispose_mapping(ssi_private->irq); return 0; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index e6b9a69..e6b6324 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -125,7 +125,9 @@ struct ccsr_ssi { #define CCSR_SSI_SRCR_REFS 0x00000001 /* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2_SHIFT 18 #define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR_SHIFT 17 #define CCSR_SSI_SxCCR_PSR 0x00020000 #define CCSR_SSI_SxCCR_WL_SHIFT 13 #define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 -- 1.8.4

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Alsa-devel May 2014