b71428d7ab
No surprises in this development cycle, and most of works are about the fixes and the improvements of the existing code, while a new LED control layer and a few new drivers have been introduced. Here are some highlights: Core: - A common mute-LED framework was introduced; used by HD-audio for now, more adaption will follow later. The former "Mic Mute-LED Mode" mixer control has been replaced with the corresponding sysfs now. - User-control management was changed to count consumed bytes instead of capping by number of elements; this will allow more controls in the normal usage pattern while avoiding the possible memory exhaustion DoS ASoC: - Continued refactoring and cleanups in ASoC core and generic card drivers - Wide range of small cppcheck and warning fixes - New drivers for Freescale i.MX DMA over rpmsg, Mediatek MT6358 accessory detection, and Realtek RT1019, RT1316, RT711 and RT715 USB-audio: - Continued improvements and fixes of the implicit feedback mode, including better support for Pioneer and Roland/BOSS devices HD-audio: - Default back to non-buffer preallocation on x86 - Cirrus codec improvements, more quirks for Realtek codecs Others: - New virtio sound driver - FireWire Bebob updates Note that this PR includes a couple of changes in reset and SPI drivers, too, and some merge conflicts might happen. -----BEGIN PGP SIGNATURE----- iQJCBAABCAAsFiEEIXTw5fNLNI7mMiVaLtJE4w1nLE8FAmCMJaAOHHRpd2FpQHN1 c2UuZGUACgkQLtJE4w1nLE/T6A//Ti0SAWYnAr5l/7ccuwS4zljHcuHngwvIxRPY BokU1ZUlagi+Ro2HLUq13G8T4AlUAQ8r2ecz7EJQHHl9tkrIg7Cc0+fiBPHju1Yu 0F3Vjc78/JsJHvAR2DPll2rwhsdD3usSQXFo181k38J098X02iNcrzsj3kW5Bpzb DBvXzOBIAg/PPfPa4edSYsSurqYeZTkhshedTohlwOCnVbW9NN5b5T9yoXP+t5na rvK1Vhu0He8nVMBPDrzjKgE5rjm7Kn0FNXZ6CMDekU9sRVzm/PbgAqqmRnn6bUKa GDpcQzlaiDrw8a7/uTVgUZy85F9kMXMMnfYpBy4bBXOt6RWOplXY1yMxy1RXV+op 3qC9k5R+IsjSWFQZ2z5bIHtGBNCG0698z9fQcvpsWTv+R68rUyfj+jeO/G9zzvpi qpQTloBfI28NoP+iGis7wtrlQ15ut47YMCQS8QiOEvLmd5/3xKXRut4Ac/VmvDpS q7fLivL8MZ/SMoXY74q/kByMBkXNpryQCAN+xAslaJ5P0aefNYJJdBt/sJlsDd9J Ya2VIxHoP+Sb1MG6OLq1Y8c53Di9lwY80pOtF3plcz/ZWgzipirf6BhFj0OttiKP a6+VewXA7zZcWEdw+Ik4dWP2dybWL+CuNl7Bwug8SyG9iWqg8Ph7FgoCTWAi93Fx KKUJxsc= =YT3U -----END PGP SIGNATURE----- Merge tag 'sound-5.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "No surprises in this development cycle, and most of work is about the fixes and the improvements of the existing code, while a new LED control layer and a few new drivers have been introduced. Here are some highlights: Core: - A common mute-LED framework was introduced. It is used by HD-audio for now, more adaption will follow later. The former "Mic Mute-LED Mode" mixer control has been replaced with the corresponding sysfs now. - User-control management was changed to count consumed bytes instead of capping by number of elements; this will allow more controls in the normal usage pattern while avoiding the possible memory exhaustion DoS ASoC: - Continued refactoring and cleanups in ASoC core and generic card drivers - Wide range of small cppcheck and warning fixes - New drivers for Freescale i.MX DMA over rpmsg, Mediatek MT6358 accessory detection, and Realtek RT1019, RT1316, RT711 and RT715 USB-audio: - Continued improvements and fixes of the implicit feedback mode, including better support for Pioneer and Roland/BOSS devices HD-audio: - Default back to non-buffer preallocation on x86 - Cirrus codec improvements, more quirks for Realtek codecs Others: - New virtio sound driver - FireWire Bebob updates" * tag 'sound-5.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (587 commits) ALSA: hda/conexant: Re-order CX5066 quirk table entries ALSA: hda/realtek: Remove redundant entry for ALC861 Haier/Uniwill devices ALSA: hda/realtek: Re-order ALC662 quirk table entries ALSA: hda/realtek: Re-order remaining ALC269 quirk table entries ALSA: hda/realtek: Re-order ALC269 Lenovo quirk table entries ALSA: hda/realtek: Re-order ALC269 Sony quirk table entries ALSA: hda/realtek: Re-order ALC269 ASUS quirk table entries ALSA: hda/realtek: Re-order ALC269 Dell quirk table entries ALSA: hda/realtek: Re-order ALC269 Acer quirk table entries ALSA: hda/realtek: Re-order ALC269 HP quirk table entries ALSA: hda/realtek: Re-order ALC882 Clevo quirk table entries ALSA: hda/realtek: Re-order ALC882 Sony quirk table entries ALSA: hda/realtek: Re-order ALC882 Acer quirk table entries ALSA: usb-audio: Remove redundant assignment to len ALSA: hda/realtek: Add quirk for Intel Clevo PCx0Dx ALSA: virtio: fix kernel-doc ALSA: hda/cirrus: Use CS8409 filter to fix abnormal sounds on Bullseye ALSA: hda/cirrus: Set Initial DMIC volume for Bullseye to -26 dB ALSA: sb: Fix two use after free in snd_sb_qsound_build ALSA: emu8000: Fix a use after free in snd_emu8000_create_mixer ...
610 lines
16 KiB
C
610 lines
16 KiB
C
// SPDX-License-Identifier: GPL-2.0-only
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/*
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* ams-delta.c -- SoC audio for Amstrad E3 (Delta) videophone
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*
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* Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
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*
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* Initially based on sound/soc/omap/osk5912.x
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* Copyright (C) 2008 Mistral Solutions
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*/
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#include <linux/gpio/consumer.h>
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#include <linux/spinlock.h>
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#include <linux/tty.h>
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#include <linux/module.h>
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#include <sound/soc.h>
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#include <sound/jack.h>
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#include <asm/mach-types.h>
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#include <linux/platform_data/asoc-ti-mcbsp.h>
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#include "omap-mcbsp.h"
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#include "../codecs/cx20442.h"
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static struct gpio_desc *handset_mute;
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static struct gpio_desc *handsfree_mute;
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static int ams_delta_event_handset(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event));
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return 0;
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}
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static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w,
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struct snd_kcontrol *k, int event)
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{
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gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event));
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return 0;
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}
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/* Board specific DAPM widgets */
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static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
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/* Handset */
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SND_SOC_DAPM_MIC("Mouthpiece", NULL),
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SND_SOC_DAPM_HP("Earpiece", ams_delta_event_handset),
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/* Handsfree/Speakerphone */
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SND_SOC_DAPM_MIC("Microphone", NULL),
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SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree),
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};
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/* How they are connected to codec pins */
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static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
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{"TELIN", NULL, "Mouthpiece"},
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{"Earpiece", NULL, "TELOUT"},
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{"MIC", NULL, "Microphone"},
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{"Speaker", NULL, "SPKOUT"},
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};
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/*
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* Controls, functional after the modem line discipline is activated.
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*/
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/* Virtual switch: audio input/output constellations */
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static const char *ams_delta_audio_mode[] =
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{"Mixed", "Handset", "Handsfree", "Speakerphone"};
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/* Selection <-> pin translation */
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#define AMS_DELTA_MOUTHPIECE 0
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#define AMS_DELTA_EARPIECE 1
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#define AMS_DELTA_MICROPHONE 2
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#define AMS_DELTA_SPEAKER 3
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#define AMS_DELTA_AGC 4
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#define AMS_DELTA_MIXED ((1 << AMS_DELTA_EARPIECE) | \
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(1 << AMS_DELTA_MICROPHONE))
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#define AMS_DELTA_HANDSET ((1 << AMS_DELTA_MOUTHPIECE) | \
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(1 << AMS_DELTA_EARPIECE))
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#define AMS_DELTA_HANDSFREE ((1 << AMS_DELTA_MICROPHONE) | \
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(1 << AMS_DELTA_SPEAKER))
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#define AMS_DELTA_SPEAKERPHONE (AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
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static const unsigned short ams_delta_audio_mode_pins[] = {
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AMS_DELTA_MIXED,
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AMS_DELTA_HANDSET,
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AMS_DELTA_HANDSFREE,
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AMS_DELTA_SPEAKERPHONE,
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};
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static unsigned short ams_delta_audio_agc;
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/*
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* Used for passing a codec structure pointer
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* from the board initialization code to the tty line discipline.
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*/
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static struct snd_soc_component *cx20442_codec;
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static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
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struct snd_soc_dapm_context *dapm = &card->dapm;
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struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
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unsigned short pins;
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int pin, changed = 0;
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/* Refuse any mode changes if we are not able to control the codec. */
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if (!cx20442_codec->card->pop_time)
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return -EUNATCH;
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if (ucontrol->value.enumerated.item[0] >= control->items)
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return -EINVAL;
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snd_soc_dapm_mutex_lock(dapm);
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/* Translate selection to bitmap */
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pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
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/* Setup pins after corresponding bits if changed */
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pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
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if (pin != snd_soc_dapm_get_pin_status(dapm, "Mouthpiece")) {
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin_unlocked(dapm, "Mouthpiece");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
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}
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pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
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if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "Earpiece");
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}
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pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
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if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "Microphone");
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}
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pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
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if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
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}
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pin = !!(pins & (1 << AMS_DELTA_AGC));
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if (pin != ams_delta_audio_agc) {
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ams_delta_audio_agc = pin;
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changed = 1;
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if (pin)
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snd_soc_dapm_enable_pin_unlocked(dapm, "AGCIN");
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else
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snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
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}
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if (changed)
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snd_soc_dapm_sync_unlocked(dapm);
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snd_soc_dapm_mutex_unlock(dapm);
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return changed;
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}
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static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
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struct snd_ctl_elem_value *ucontrol)
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{
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struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
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struct snd_soc_dapm_context *dapm = &card->dapm;
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unsigned short pins, mode;
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pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
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AMS_DELTA_MOUTHPIECE) |
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(snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
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AMS_DELTA_EARPIECE));
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if (pins)
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pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
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AMS_DELTA_MICROPHONE);
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else
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pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
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AMS_DELTA_MICROPHONE) |
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(snd_soc_dapm_get_pin_status(dapm, "Speaker") <<
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AMS_DELTA_SPEAKER) |
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(ams_delta_audio_agc << AMS_DELTA_AGC));
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for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
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if (pins == ams_delta_audio_mode_pins[mode])
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break;
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if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
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return -EINVAL;
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ucontrol->value.enumerated.item[0] = mode;
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return 0;
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}
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static SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum,
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ams_delta_audio_mode);
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static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
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SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum,
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ams_delta_get_audio_mode, ams_delta_set_audio_mode),
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};
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/* Hook switch */
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static struct snd_soc_jack ams_delta_hook_switch;
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static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
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{
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.name = "hook_switch",
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.report = SND_JACK_HEADSET,
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.invert = 1,
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.debounce_time = 150,
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}
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};
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/* After we are able to control the codec over the modem,
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* the hook switch can be used for dynamic DAPM reconfiguration. */
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static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
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/* Handset */
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{
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.pin = "Mouthpiece",
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.mask = SND_JACK_MICROPHONE,
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},
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{
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.pin = "Earpiece",
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.mask = SND_JACK_HEADPHONE,
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},
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/* Handsfree */
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{
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.pin = "Microphone",
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.mask = SND_JACK_MICROPHONE,
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.invert = 1,
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},
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{
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.pin = "Speaker",
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.mask = SND_JACK_HEADPHONE,
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.invert = 1,
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},
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};
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/*
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* Modem line discipline, required for making above controls functional.
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* Activated from userspace with ldattach, possibly invoked from udev rule.
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*/
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/* To actually apply any modem controlled configuration changes to the codec,
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* we must connect codec DAI pins to the modem for a moment. Be careful not
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* to interfere with our digital mute function that shares the same hardware. */
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static struct timer_list cx81801_timer;
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static bool cx81801_cmd_pending;
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static bool ams_delta_muted;
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static DEFINE_SPINLOCK(ams_delta_lock);
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static struct gpio_desc *gpiod_modem_codec;
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static void cx81801_timeout(struct timer_list *unused)
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{
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int muted;
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spin_lock(&ams_delta_lock);
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cx81801_cmd_pending = 0;
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muted = ams_delta_muted;
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spin_unlock(&ams_delta_lock);
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/* Reconnect the codec DAI back from the modem to the CPU DAI
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* only if digital mute still off */
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if (!muted)
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gpiod_set_value(gpiod_modem_codec, 0);
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}
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/* Line discipline .open() */
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static int cx81801_open(struct tty_struct *tty)
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{
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int ret;
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if (!cx20442_codec)
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return -ENODEV;
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/*
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* Pass the codec structure pointer for use by other ldisc callbacks,
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* both the card and the codec specific parts.
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*/
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tty->disc_data = cx20442_codec;
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ret = v253_ops.open(tty);
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if (ret < 0)
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tty->disc_data = NULL;
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return ret;
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}
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/* Line discipline .close() */
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static void cx81801_close(struct tty_struct *tty)
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{
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struct snd_soc_component *component = tty->disc_data;
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struct snd_soc_dapm_context *dapm = &component->card->dapm;
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del_timer_sync(&cx81801_timer);
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/* Prevent the hook switch from further changing the DAPM pins */
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INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
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if (!component)
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return;
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v253_ops.close(tty);
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/* Revert back to default audio input/output constellation */
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snd_soc_dapm_mutex_lock(dapm);
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snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
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snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
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snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
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snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
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snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
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snd_soc_dapm_sync_unlocked(dapm);
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snd_soc_dapm_mutex_unlock(dapm);
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}
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/* Line discipline .hangup() */
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static int cx81801_hangup(struct tty_struct *tty)
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{
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cx81801_close(tty);
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return 0;
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}
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/* Line discipline .receive_buf() */
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static void cx81801_receive(struct tty_struct *tty,
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const unsigned char *cp, char *fp, int count)
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{
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struct snd_soc_component *component = tty->disc_data;
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const unsigned char *c;
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int apply, ret;
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if (!component)
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return;
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if (!component->card->pop_time) {
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/* First modem response, complete setup procedure */
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/* Initialize timer used for config pulse generation */
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timer_setup(&cx81801_timer, cx81801_timeout, 0);
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v253_ops.receive_buf(tty, cp, fp, count);
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/* Link hook switch to DAPM pins */
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ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
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ARRAY_SIZE(ams_delta_hook_switch_pins),
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ams_delta_hook_switch_pins);
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if (ret)
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dev_warn(component->dev,
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"Failed to link hook switch to DAPM pins, "
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"will continue with hook switch unlinked.\n");
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return;
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}
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v253_ops.receive_buf(tty, cp, fp, count);
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for (c = &cp[count - 1]; c >= cp; c--) {
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if (*c != '\r')
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continue;
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/* Complete modem response received, apply config to codec */
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spin_lock_bh(&ams_delta_lock);
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mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
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apply = !ams_delta_muted && !cx81801_cmd_pending;
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cx81801_cmd_pending = 1;
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spin_unlock_bh(&ams_delta_lock);
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/* Apply config pulse by connecting the codec to the modem
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* if not already done */
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if (apply)
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|
gpiod_set_value(gpiod_modem_codec, 1);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* Line discipline .write_wakeup() */
|
|
static void cx81801_wakeup(struct tty_struct *tty)
|
|
{
|
|
v253_ops.write_wakeup(tty);
|
|
}
|
|
|
|
static struct tty_ldisc_ops cx81801_ops = {
|
|
.name = "cx81801",
|
|
.owner = THIS_MODULE,
|
|
.open = cx81801_open,
|
|
.close = cx81801_close,
|
|
.hangup = cx81801_hangup,
|
|
.receive_buf = cx81801_receive,
|
|
.write_wakeup = cx81801_wakeup,
|
|
};
|
|
|
|
|
|
/*
|
|
* Even if not very useful, the sound card can still work without any of the
|
|
* above functionality activated. You can still control its audio input/output
|
|
* constellation and speakerphone gain from userspace by issuing AT commands
|
|
* over the modem port.
|
|
*/
|
|
|
|
static struct snd_soc_ops ams_delta_ops;
|
|
|
|
|
|
/* Digital mute implemented using modem/CPU multiplexer.
|
|
* Shares hardware with codec config pulse generation */
|
|
static bool ams_delta_muted = 1;
|
|
|
|
static int ams_delta_mute(struct snd_soc_dai *dai, int mute, int direction)
|
|
{
|
|
int apply;
|
|
|
|
if (ams_delta_muted == mute)
|
|
return 0;
|
|
|
|
spin_lock_bh(&ams_delta_lock);
|
|
ams_delta_muted = mute;
|
|
apply = !cx81801_cmd_pending;
|
|
spin_unlock_bh(&ams_delta_lock);
|
|
|
|
if (apply)
|
|
gpiod_set_value(gpiod_modem_codec, !!mute);
|
|
return 0;
|
|
}
|
|
|
|
/* Our codec DAI probably doesn't have its own .ops structure */
|
|
static const struct snd_soc_dai_ops ams_delta_dai_ops = {
|
|
.mute_stream = ams_delta_mute,
|
|
.no_capture_mute = 1,
|
|
};
|
|
|
|
/* Will be used if the codec ever has its own digital_mute function */
|
|
static int ams_delta_startup(struct snd_pcm_substream *substream)
|
|
{
|
|
return ams_delta_mute(NULL, 0, substream->stream);
|
|
}
|
|
|
|
static void ams_delta_shutdown(struct snd_pcm_substream *substream)
|
|
{
|
|
ams_delta_mute(NULL, 1, substream->stream);
|
|
}
|
|
|
|
|
|
/*
|
|
* Card initialization
|
|
*/
|
|
|
|
static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
|
|
{
|
|
struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
|
|
struct snd_soc_card *card = rtd->card;
|
|
struct snd_soc_dapm_context *dapm = &card->dapm;
|
|
int ret;
|
|
/* Codec is ready, now add/activate board specific controls */
|
|
|
|
/* Store a pointer to the codec structure for tty ldisc use */
|
|
cx20442_codec = asoc_rtd_to_codec(rtd, 0)->component;
|
|
|
|
/* Add hook switch - can be used to control the codec from userspace
|
|
* even if line discipline fails */
|
|
ret = snd_soc_card_jack_new(card, "hook_switch", SND_JACK_HEADSET,
|
|
&ams_delta_hook_switch, NULL, 0);
|
|
if (ret)
|
|
dev_warn(card->dev,
|
|
"Failed to allocate resources for hook switch, "
|
|
"will continue without one.\n");
|
|
else {
|
|
ret = snd_soc_jack_add_gpiods(card->dev, &ams_delta_hook_switch,
|
|
ARRAY_SIZE(ams_delta_hook_switch_gpios),
|
|
ams_delta_hook_switch_gpios);
|
|
if (ret)
|
|
dev_warn(card->dev,
|
|
"Failed to set up hook switch GPIO line, "
|
|
"will continue with hook switch inactive.\n");
|
|
}
|
|
|
|
gpiod_modem_codec = devm_gpiod_get(card->dev, "modem_codec",
|
|
GPIOD_OUT_HIGH);
|
|
if (IS_ERR(gpiod_modem_codec)) {
|
|
dev_warn(card->dev, "Failed to obtain modem_codec GPIO\n");
|
|
return 0;
|
|
}
|
|
|
|
/* Set up digital mute if not provided by the codec */
|
|
if (!codec_dai->driver->ops) {
|
|
codec_dai->driver->ops = &ams_delta_dai_ops;
|
|
} else {
|
|
ams_delta_ops.startup = ams_delta_startup;
|
|
ams_delta_ops.shutdown = ams_delta_shutdown;
|
|
}
|
|
|
|
/* Register optional line discipline for over the modem control */
|
|
ret = tty_register_ldisc(N_V253, &cx81801_ops);
|
|
if (ret) {
|
|
dev_warn(card->dev,
|
|
"Failed to register line discipline, "
|
|
"will continue without any controls.\n");
|
|
return 0;
|
|
}
|
|
|
|
/* Set up initial pin constellation */
|
|
snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
|
|
snd_soc_dapm_disable_pin(dapm, "Speaker");
|
|
snd_soc_dapm_disable_pin(dapm, "AGCIN");
|
|
snd_soc_dapm_disable_pin(dapm, "AGCOUT");
|
|
|
|
return 0;
|
|
}
|
|
|
|
/* DAI glue - connects codec <--> CPU */
|
|
SND_SOC_DAILINK_DEFS(cx20442,
|
|
DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.1")),
|
|
DAILINK_COMP_ARRAY(COMP_CODEC("cx20442-codec", "cx20442-voice")),
|
|
DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.1")));
|
|
|
|
static struct snd_soc_dai_link ams_delta_dai_link = {
|
|
.name = "CX20442",
|
|
.stream_name = "CX20442",
|
|
.init = ams_delta_cx20442_init,
|
|
.ops = &ams_delta_ops,
|
|
.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF |
|
|
SND_SOC_DAIFMT_CBM_CFM,
|
|
SND_SOC_DAILINK_REG(cx20442),
|
|
};
|
|
|
|
/* Audio card driver */
|
|
static struct snd_soc_card ams_delta_audio_card = {
|
|
.name = "AMS_DELTA",
|
|
.owner = THIS_MODULE,
|
|
.dai_link = &ams_delta_dai_link,
|
|
.num_links = 1,
|
|
|
|
.controls = ams_delta_audio_controls,
|
|
.num_controls = ARRAY_SIZE(ams_delta_audio_controls),
|
|
.dapm_widgets = ams_delta_dapm_widgets,
|
|
.num_dapm_widgets = ARRAY_SIZE(ams_delta_dapm_widgets),
|
|
.dapm_routes = ams_delta_audio_map,
|
|
.num_dapm_routes = ARRAY_SIZE(ams_delta_audio_map),
|
|
};
|
|
|
|
/* Module init/exit */
|
|
static int ams_delta_probe(struct platform_device *pdev)
|
|
{
|
|
struct snd_soc_card *card = &ams_delta_audio_card;
|
|
int ret;
|
|
|
|
card->dev = &pdev->dev;
|
|
|
|
handset_mute = devm_gpiod_get(card->dev, "handset_mute",
|
|
GPIOD_OUT_HIGH);
|
|
if (IS_ERR(handset_mute))
|
|
return PTR_ERR(handset_mute);
|
|
|
|
handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute",
|
|
GPIOD_OUT_HIGH);
|
|
if (IS_ERR(handsfree_mute))
|
|
return PTR_ERR(handsfree_mute);
|
|
|
|
ret = snd_soc_register_card(card);
|
|
if (ret) {
|
|
dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
|
|
card->dev = NULL;
|
|
return ret;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static int ams_delta_remove(struct platform_device *pdev)
|
|
{
|
|
struct snd_soc_card *card = platform_get_drvdata(pdev);
|
|
|
|
if (tty_unregister_ldisc(N_V253) != 0)
|
|
dev_warn(&pdev->dev,
|
|
"failed to unregister V253 line discipline\n");
|
|
|
|
snd_soc_unregister_card(card);
|
|
card->dev = NULL;
|
|
return 0;
|
|
}
|
|
|
|
#define DRV_NAME "ams-delta-audio"
|
|
|
|
static struct platform_driver ams_delta_driver = {
|
|
.driver = {
|
|
.name = DRV_NAME,
|
|
},
|
|
.probe = ams_delta_probe,
|
|
.remove = ams_delta_remove,
|
|
};
|
|
|
|
module_platform_driver(ams_delta_driver);
|
|
|
|
MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
|
|
MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
|
|
MODULE_LICENSE("GPL");
|
|
MODULE_ALIAS("platform:" DRV_NAME);
|