forked from Minki/linux
f4c80d5a16
This time was again a relatively calm development cycle; most of updates are about drivers, and no radical changes are seen in any core code. Here are some highlights: ALSA core: - Continued hardening of ALSA hrtimer - A few leak fixes in timer interface - Fix poll error handling in PCM and compress - Add error propagation in compress API - Removal of dead rtctimer driver HD-audio: - Native ELD notify support for i915 HDMI - Realtek ALC234 & co support - Code refactoring to standardize chmap support - Continued development for SKL HDMI core support Firewire: - Apply delayed card registration to all drivers - Improved / stabilized the handling of PCM stream start / stop - Add tracepoints to dump a part of isochronous packet data - Fixed incoming/outgoing packet parameter usages - Add support for M-Audio profire series USB-audio: - Fixes for UAC2 clock source - SS+ support - Workaround for oft-seen repeated sample rate read errors ASoC: - Further slow progress on the topology code - Substantial updates and improvements for the da7219, es8328, fsl-ssi, Intel and rcar drivers. - Compress error handling in WM ADSP driver -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAABCAAGBQJXPYgvAAoJEGwxgFQ9KSmka3IQAJfXxKYyL0mqOgUpFav2QprE j4nQFSQf2KMAHgod1iF4Pv5glRZ3T8CbWllu/+GT87ny4wwJH76D07VCZSnrA+cv NMxRMN8QiGWS+eNPDNqRbcpzQvgwRK17VAmvpIfZtdntq3IryPLyCnY+FJ6Xt5v7 CjgGjlKJQ8i6AJVtoKVlrCOTBPS8YezQ7o67v8+BNrHDyOr0pwLERhvqJBRjaCbj fKj+JNDsWyu4kX0nInKNGah+5Qiib68+UNK5M+/PnoWv9tEOBPNXeWqRkcRpwnrF t1BQLnKGdlcSIufXcvxHDdxLftJZ38w+EbnQ/2r+SYHYIwPqTWdvVeXZUiq70wW/ WBUEOHybaHTNc52nMpjo/PU72CHa29zvKq+QHMXMRmFfVrLepIgEpBRBUjENtCjM 3OUn1IhYiNI4FOfgLm5duuYSBVdS4C2qstBDMtGpP64l7AmBZMFtbGUP8pKhvpzF FR2VoQpBFLPo805lQBKYbxdpzUGqfR7M/O73WRMzB/ZPZa95VNCDoRDQBbYF4Wzy SByVcE56znxoS9AmbhU6LzCXxdyVp6YAXZNR0pHp+8QdrRoFQZwRhfNVN3FIeNub COV+0pCQ2GTYvVdfLjdh6VT4shXeg5ZrUVnE3akL+8OzXow9lKyhknvLHn71aTZi HT0vSirSdrEYf4zg6wtB =QsAc -----END PGP SIGNATURE----- Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound updates from Takashi Iwai: "This time was again a relatively calm development cycle; most of updates are about drivers, and no radical changes are seen in any core code. Here are some highlights: ALSA core: - Continued hardening of ALSA hrtimer - A few leak fixes in timer interface - Fix poll error handling in PCM and compress - Add error propagation in compress API - Removal of dead rtctimer driver HD-audio: - Native ELD notify support for i915 HDMI - Realtek ALC234 & co support - Code refactoring to standardize chmap support - Continued development for SKL HDMI core support Firewire: - Apply delayed card registration to all drivers - Improved / stabilized the handling of PCM stream start / stop - Add tracepoints to dump a part of isochronous packet data - Fixed incoming/outgoing packet parameter usages - Add support for M-Audio profire series USB-audio: - Fixes for UAC2 clock source - SS+ support - Workaround for oft-seen repeated sample rate read errors ASoC: - Further slow progress on the topology code - Substantial updates and improvements for the da7219, es8328, fsl-ssi, Intel and rcar drivers. - Compress error handling in WM ADSP driver" * tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (177 commits) ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type sound: oss: Use setup_timer and mod_timer. ASoC: hdac_hdmi: Remove the unused 'timeout' variable ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex. ASoC: fsl_ssi: Fix channel slipping in Playback at startup ASoC: fsl_ssi: Fix samples being dropped at Playback startup ASoC: fsl_ssi: Save a dev reference for dev_err() purpose. ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk. ASoC: fsl_ssi: Real hardware channels max number is 32 ASoC: pcm5102a: Add support for PCM5102A codec ASoC: hdac_hdmi: add link management ASoC: Intel: Skylake: add link management ALSA: hdac: add link pm and ref counting ALSA: au88x0: Fix zero clear of stream->resources ASoC: rt298: Add DMI match for Broxton-P reference platform ASoC: rt298: fix null deref on acpi driver data ASoC: dapm: deprecate MICBIAS widget type ALSA: firewire-lib: drop skip argument from helper functions to queue a packet ALSA: firewire-lib: add context information to tracepoints ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity ...
332 lines
8.9 KiB
C
332 lines
8.9 KiB
C
/*
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* cht-bsw-max98090.c - ASoc Machine driver for Intel Cherryview-based
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* platforms Cherrytrail and Braswell, with max98090 & TI codec.
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*
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* Copyright (C) 2015 Intel Corp
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* Author: Fang, Yang A <yang.a.fang@intel.com>
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* This file is modified from cht_bsw_rt5645.c
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* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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*
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* This program is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; version 2 of the License.
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*
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* This program is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* General Public License for more details.
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*
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* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
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*/
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#include <linux/module.h>
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#include <linux/platform_device.h>
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#include <linux/slab.h>
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#include <linux/acpi.h>
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#include <sound/pcm.h>
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#include <sound/pcm_params.h>
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#include <sound/soc.h>
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#include <sound/jack.h>
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#include "../../codecs/max98090.h"
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#include "../atom/sst-atom-controls.h"
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#include "../../codecs/ts3a227e.h"
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#define CHT_PLAT_CLK_3_HZ 19200000
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#define CHT_CODEC_DAI "HiFi"
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struct cht_mc_private {
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struct snd_soc_jack jack;
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bool ts3a227e_present;
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};
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static inline struct snd_soc_dai *cht_get_codec_dai(struct snd_soc_card *card)
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{
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struct snd_soc_pcm_runtime *rtd;
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list_for_each_entry(rtd, &card->rtd_list, list) {
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if (!strncmp(rtd->codec_dai->name, CHT_CODEC_DAI,
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strlen(CHT_CODEC_DAI)))
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return rtd->codec_dai;
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}
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return NULL;
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}
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static const struct snd_soc_dapm_widget cht_dapm_widgets[] = {
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SND_SOC_DAPM_HP("Headphone", NULL),
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SND_SOC_DAPM_MIC("Headset Mic", NULL),
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SND_SOC_DAPM_MIC("Int Mic", NULL),
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SND_SOC_DAPM_SPK("Ext Spk", NULL),
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};
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static const struct snd_soc_dapm_route cht_audio_map[] = {
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{"IN34", NULL, "Headset Mic"},
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{"Headset Mic", NULL, "MICBIAS"},
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{"DMICL", NULL, "Int Mic"},
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{"Headphone", NULL, "HPL"},
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{"Headphone", NULL, "HPR"},
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{"Ext Spk", NULL, "SPKL"},
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{"Ext Spk", NULL, "SPKR"},
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{"HiFi Playback", NULL, "ssp2 Tx"},
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{"ssp2 Tx", NULL, "codec_out0"},
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{"ssp2 Tx", NULL, "codec_out1"},
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{"codec_in0", NULL, "ssp2 Rx" },
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{"codec_in1", NULL, "ssp2 Rx" },
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{"ssp2 Rx", NULL, "HiFi Capture"},
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};
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static const struct snd_kcontrol_new cht_mc_controls[] = {
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SOC_DAPM_PIN_SWITCH("Headphone"),
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SOC_DAPM_PIN_SWITCH("Headset Mic"),
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SOC_DAPM_PIN_SWITCH("Int Mic"),
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SOC_DAPM_PIN_SWITCH("Ext Spk"),
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};
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static int cht_aif1_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params)
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{
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struct snd_soc_pcm_runtime *rtd = substream->private_data;
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struct snd_soc_dai *codec_dai = rtd->codec_dai;
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int ret;
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ret = snd_soc_dai_set_sysclk(codec_dai, M98090_REG_SYSTEM_CLOCK,
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CHT_PLAT_CLK_3_HZ, SND_SOC_CLOCK_IN);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret);
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return ret;
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}
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return 0;
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}
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static int cht_ti_jack_event(struct notifier_block *nb,
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unsigned long event, void *data)
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{
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struct snd_soc_jack *jack = (struct snd_soc_jack *)data;
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struct snd_soc_dapm_context *dapm = &jack->card->dapm;
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if (event & SND_JACK_MICROPHONE) {
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snd_soc_dapm_force_enable_pin(dapm, "SHDN");
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snd_soc_dapm_force_enable_pin(dapm, "MICBIAS");
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snd_soc_dapm_sync(dapm);
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} else {
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snd_soc_dapm_disable_pin(dapm, "MICBIAS");
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snd_soc_dapm_disable_pin(dapm, "SHDN");
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snd_soc_dapm_sync(dapm);
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}
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return 0;
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}
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static struct notifier_block cht_jack_nb = {
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.notifier_call = cht_ti_jack_event,
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};
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static int cht_codec_init(struct snd_soc_pcm_runtime *runtime)
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{
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int ret;
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int jack_type;
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struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card);
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struct snd_soc_jack *jack = &ctx->jack;
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/**
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* TI supports 4 butons headset detection
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* KEY_MEDIA
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* KEY_VOICECOMMAND
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* KEY_VOLUMEUP
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* KEY_VOLUMEDOWN
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*/
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if (ctx->ts3a227e_present)
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jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE |
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SND_JACK_BTN_0 | SND_JACK_BTN_1 |
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SND_JACK_BTN_2 | SND_JACK_BTN_3;
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else
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jack_type = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE;
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ret = snd_soc_card_jack_new(runtime->card, "Headset Jack",
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jack_type, jack, NULL, 0);
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if (ret) {
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dev_err(runtime->dev, "Headset Jack creation failed %d\n", ret);
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return ret;
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}
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if (ctx->ts3a227e_present)
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snd_soc_jack_notifier_register(jack, &cht_jack_nb);
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return ret;
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}
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static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd,
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struct snd_pcm_hw_params *params)
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{
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struct snd_interval *rate = hw_param_interval(params,
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SNDRV_PCM_HW_PARAM_RATE);
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struct snd_interval *channels = hw_param_interval(params,
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SNDRV_PCM_HW_PARAM_CHANNELS);
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int ret = 0;
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unsigned int fmt = 0;
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ret = snd_soc_dai_set_tdm_slot(rtd->cpu_dai, 0x3, 0x3, 2, 16);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set cpu_dai slot fmt: %d\n", ret);
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return ret;
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}
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fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF
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| SND_SOC_DAIFMT_CBS_CFS;
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ret = snd_soc_dai_set_fmt(rtd->cpu_dai, fmt);
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if (ret < 0) {
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dev_err(rtd->dev, "can't set cpu_dai set fmt: %d\n", ret);
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return ret;
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}
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/* The DSP will covert the FE rate to 48k, stereo, 24bits */
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rate->min = rate->max = 48000;
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channels->min = channels->max = 2;
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/* set SSP2 to 24-bit */
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params_set_format(params, SNDRV_PCM_FORMAT_S24_LE);
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return 0;
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}
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static int cht_aif1_startup(struct snd_pcm_substream *substream)
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{
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return snd_pcm_hw_constraint_single(substream->runtime,
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SNDRV_PCM_HW_PARAM_RATE, 48000);
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}
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static int cht_max98090_headset_init(struct snd_soc_component *component)
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{
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struct snd_soc_card *card = component->card;
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struct cht_mc_private *ctx = snd_soc_card_get_drvdata(card);
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return ts3a227e_enable_jack_detect(component, &ctx->jack);
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}
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static struct snd_soc_ops cht_aif1_ops = {
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.startup = cht_aif1_startup,
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};
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static struct snd_soc_ops cht_be_ssp2_ops = {
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.hw_params = cht_aif1_hw_params,
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};
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static struct snd_soc_aux_dev cht_max98090_headset_dev = {
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.name = "Headset Chip",
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.init = cht_max98090_headset_init,
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.codec_name = "i2c-104C227E:00",
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};
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static struct snd_soc_dai_link cht_dailink[] = {
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[MERR_DPCM_AUDIO] = {
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.name = "Audio Port",
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.stream_name = "Audio",
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.cpu_dai_name = "media-cpu-dai",
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.codec_dai_name = "snd-soc-dummy-dai",
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.codec_name = "snd-soc-dummy",
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.platform_name = "sst-mfld-platform",
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.nonatomic = true,
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.dynamic = 1,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.ops = &cht_aif1_ops,
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},
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[MERR_DPCM_DEEP_BUFFER] = {
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.name = "Deep-Buffer Audio Port",
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.stream_name = "Deep-Buffer Audio",
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.cpu_dai_name = "deepbuffer-cpu-dai",
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.codec_dai_name = "snd-soc-dummy-dai",
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.codec_name = "snd-soc-dummy",
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.platform_name = "sst-mfld-platform",
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.nonatomic = true,
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.dynamic = 1,
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.dpcm_playback = 1,
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.ops = &cht_aif1_ops,
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},
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[MERR_DPCM_COMPR] = {
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.name = "Compressed Port",
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.stream_name = "Compress",
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.cpu_dai_name = "compress-cpu-dai",
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.codec_dai_name = "snd-soc-dummy-dai",
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.codec_name = "snd-soc-dummy",
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.platform_name = "sst-mfld-platform",
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},
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/* back ends */
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{
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.name = "SSP2-Codec",
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.id = 1,
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.cpu_dai_name = "ssp2-port",
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.platform_name = "sst-mfld-platform",
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.no_pcm = 1,
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.codec_dai_name = "HiFi",
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.codec_name = "i2c-193C9890:00",
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.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF
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| SND_SOC_DAIFMT_CBS_CFS,
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.init = cht_codec_init,
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.be_hw_params_fixup = cht_codec_fixup,
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.dpcm_playback = 1,
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.dpcm_capture = 1,
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.ops = &cht_be_ssp2_ops,
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},
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};
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/* SoC card */
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static struct snd_soc_card snd_soc_card_cht = {
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.name = "chtmax98090",
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.owner = THIS_MODULE,
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.dai_link = cht_dailink,
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.num_links = ARRAY_SIZE(cht_dailink),
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.aux_dev = &cht_max98090_headset_dev,
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.num_aux_devs = 1,
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.dapm_widgets = cht_dapm_widgets,
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.num_dapm_widgets = ARRAY_SIZE(cht_dapm_widgets),
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.dapm_routes = cht_audio_map,
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.num_dapm_routes = ARRAY_SIZE(cht_audio_map),
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.controls = cht_mc_controls,
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.num_controls = ARRAY_SIZE(cht_mc_controls),
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};
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static int snd_cht_mc_probe(struct platform_device *pdev)
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{
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int ret_val = 0;
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struct cht_mc_private *drv;
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drv = devm_kzalloc(&pdev->dev, sizeof(*drv), GFP_ATOMIC);
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if (!drv)
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return -ENOMEM;
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drv->ts3a227e_present = acpi_dev_found("104C227E");
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if (!drv->ts3a227e_present) {
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/* no need probe TI jack detection chip */
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snd_soc_card_cht.aux_dev = NULL;
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snd_soc_card_cht.num_aux_devs = 0;
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}
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/* register the soc card */
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snd_soc_card_cht.dev = &pdev->dev;
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snd_soc_card_set_drvdata(&snd_soc_card_cht, drv);
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ret_val = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cht);
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if (ret_val) {
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dev_err(&pdev->dev,
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"snd_soc_register_card failed %d\n", ret_val);
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return ret_val;
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}
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platform_set_drvdata(pdev, &snd_soc_card_cht);
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return ret_val;
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}
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static struct platform_driver snd_cht_mc_driver = {
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.driver = {
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.name = "cht-bsw-max98090",
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},
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.probe = snd_cht_mc_probe,
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};
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module_platform_driver(snd_cht_mc_driver)
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MODULE_DESCRIPTION("ASoC Intel(R) Braswell Machine driver");
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MODULE_AUTHOR("Fang, Yang A <yang.a.fang@intel.com>");
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MODULE_LICENSE("GPL v2");
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MODULE_ALIAS("platform:cht-bsw-max98090");
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