Merge remote-tracking branches 'asoc/topic/dpcm', 'asoc/topic/es8328', 'asoc/topic/extcon' and 'asoc/topic/fsl' into asoc-next
Mark Brown
commit
fba3040105
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@ -274,9 +274,10 @@ static void arizona_extcon_pulse_micbias(struct arizona_extcon_info *info)
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struct arizona *arizona = info->arizona;
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const char *widget = arizona_extcon_get_micbias(info);
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struct snd_soc_dapm_context *dapm = arizona->dapm;
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struct snd_soc_component *component = snd_soc_dapm_to_component(dapm);
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int ret;
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ret = snd_soc_dapm_force_enable_pin(dapm, widget);
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ret = snd_soc_component_force_enable_pin(component, widget);
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if (ret != 0)
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dev_warn(arizona->dev, "Failed to enable %s: %d\n",
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widget, ret);
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@ -284,7 +285,7 @@ static void arizona_extcon_pulse_micbias(struct arizona_extcon_info *info)
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snd_soc_dapm_sync(dapm);
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if (!arizona->pdata.micd_force_micbias) {
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ret = snd_soc_dapm_disable_pin(arizona->dapm, widget);
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ret = snd_soc_component_disable_pin(component, widget);
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if (ret != 0)
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dev_warn(arizona->dev, "Failed to disable %s: %d\n",
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widget, ret);
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@ -349,6 +350,7 @@ static void arizona_stop_mic(struct arizona_extcon_info *info)
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struct arizona *arizona = info->arizona;
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const char *widget = arizona_extcon_get_micbias(info);
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struct snd_soc_dapm_context *dapm = arizona->dapm;
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struct snd_soc_component *component = snd_soc_dapm_to_component(dapm);
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bool change;
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int ret;
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@ -356,7 +358,7 @@ static void arizona_stop_mic(struct arizona_extcon_info *info)
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ARIZONA_MICD_ENA, 0,
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&change);
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ret = snd_soc_dapm_disable_pin(dapm, widget);
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ret = snd_soc_component_disable_pin(component, widget);
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if (ret != 0)
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dev_warn(arizona->dev,
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"Failed to disable %s: %d\n",
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@ -27,7 +27,27 @@
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#include <sound/tlv.h>
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#include <sound/ak4641.h>
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#include "ak4641.h"
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/* AK4641 register space */
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#define AK4641_PM1 0x00
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#define AK4641_PM2 0x01
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#define AK4641_SIG1 0x02
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#define AK4641_SIG2 0x03
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#define AK4641_MODE1 0x04
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#define AK4641_MODE2 0x05
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#define AK4641_DAC 0x06
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#define AK4641_MIC 0x07
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#define AK4641_TIMER 0x08
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#define AK4641_ALC1 0x09
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#define AK4641_ALC2 0x0a
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#define AK4641_PGA 0x0b
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#define AK4641_LATT 0x0c
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#define AK4641_RATT 0x0d
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#define AK4641_VOL 0x0e
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#define AK4641_STATUS 0x0f
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#define AK4641_EQLO 0x10
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#define AK4641_EQMID 0x11
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#define AK4641_EQHI 0x12
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#define AK4641_BTIF 0x13
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/* codec private data */
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struct ak4641_priv {
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@ -1,47 +0,0 @@
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/*
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* ak4641.h -- AK4641 SoC Audio driver
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*
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* Copyright 2008 Harald Welte <laforge@gnufiish.org>
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*
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* Based on ak4535.h
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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 version 2 as
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* published by the Free Software Foundation.
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*/
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#ifndef _AK4641_H
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#define _AK4641_H
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/* AK4641 register space */
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#define AK4641_PM1 0x00
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#define AK4641_PM2 0x01
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#define AK4641_SIG1 0x02
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#define AK4641_SIG2 0x03
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#define AK4641_MODE1 0x04
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#define AK4641_MODE2 0x05
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#define AK4641_DAC 0x06
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#define AK4641_MIC 0x07
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#define AK4641_TIMER 0x08
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#define AK4641_ALC1 0x09
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#define AK4641_ALC2 0x0a
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#define AK4641_PGA 0x0b
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#define AK4641_LATT 0x0c
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#define AK4641_RATT 0x0d
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#define AK4641_VOL 0x0e
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#define AK4641_STATUS 0x0f
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#define AK4641_EQLO 0x10
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#define AK4641_EQMID 0x11
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#define AK4641_EQHI 0x12
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#define AK4641_BTIF 0x13
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#define AK4641_CACHEREGNUM 0x14
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#define AK4641_DAI_HIFI 0
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#define AK4641_DAI_VOICE 1
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#endif
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@ -278,43 +278,6 @@ int es8328_probe(struct device *dev, struct regmap *regmap);
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#define ES8328_REG_MAX 0x35
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#define ES8328_PLL1 0
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#define ES8328_PLL2 1
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/* clock inputs */
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#define ES8328_MCLK 0
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#define ES8328_PCMCLK 1
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/* clock divider id's */
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#define ES8328_PCMDIV 0
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#define ES8328_BCLKDIV 1
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#define ES8328_VXCLKDIV 2
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/* PCM clock dividers */
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#define ES8328_PCM_DIV_1 (0 << 6)
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#define ES8328_PCM_DIV_3 (2 << 6)
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#define ES8328_PCM_DIV_5_5 (3 << 6)
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#define ES8328_PCM_DIV_2 (4 << 6)
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#define ES8328_PCM_DIV_4 (5 << 6)
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#define ES8328_PCM_DIV_6 (6 << 6)
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#define ES8328_PCM_DIV_8 (7 << 6)
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/* BCLK clock dividers */
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#define ES8328_BCLK_DIV_1 (0 << 7)
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#define ES8328_BCLK_DIV_2 (1 << 7)
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#define ES8328_BCLK_DIV_4 (2 << 7)
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#define ES8328_BCLK_DIV_8 (3 << 7)
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/* VXCLK clock dividers */
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#define ES8328_VXCLK_DIV_1 (0 << 6)
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#define ES8328_VXCLK_DIV_2 (1 << 6)
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#define ES8328_VXCLK_DIV_4 (2 << 6)
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#define ES8328_VXCLK_DIV_8 (3 << 6)
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#define ES8328_VXCLK_DIV_16 (4 << 6)
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#define ES8328_DAI_HIFI 0
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#define ES8328_DAI_VOICE 1
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#define ES8328_1536FS 1536
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#define ES8328_1024FS 1024
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#define ES8328_768FS 768
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@ -40,6 +40,7 @@ config SND_SOC_FSL_SPDIF
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select REGMAP_MMIO
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select SND_SOC_IMX_PCM_DMA if SND_IMX_SOC != n
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select SND_SOC_IMX_PCM_FIQ if SND_IMX_SOC != n && (MXC_TZIC || MXC_AVIC)
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select BITREVERSE
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help
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Say Y if you want to add Sony/Philips Digital Interface (SPDIF)
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support for the Freescale CPUs.
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@ -735,6 +735,11 @@ static int mxs_saif_probe(struct platform_device *pdev)
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else
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saif->id = ret;
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if (saif->id >= ARRAY_SIZE(mxs_saif)) {
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dev_err(&pdev->dev, "get wrong saif id\n");
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return -EINVAL;
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}
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/*
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* If there is no "fsl,saif-master" phandle, it's a saif
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* master. Otherwise, it's a slave and its phandle points
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@ -749,11 +754,11 @@ static int mxs_saif_probe(struct platform_device *pdev)
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return ret;
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else
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saif->master_id = ret;
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}
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if (saif->master_id >= ARRAY_SIZE(mxs_saif)) {
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dev_err(&pdev->dev, "get wrong master id\n");
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return -EINVAL;
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if (saif->master_id >= ARRAY_SIZE(mxs_saif)) {
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dev_err(&pdev->dev, "get wrong master id\n");
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return -EINVAL;
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}
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}
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mxs_saif[saif->id] = saif;
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@ -27,8 +27,6 @@
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#include <asm/mach-types.h>
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#include "pxa2xx-i2s.h"
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#include "../codecs/ak4641.h"
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static struct snd_soc_jack hs_jack;
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/* Headphones jack detection DAPM pin */
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@ -2882,7 +2882,7 @@ int snd_soc_platform_trigger(struct snd_pcm_substream *substream,
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EXPORT_SYMBOL_GPL(snd_soc_platform_trigger);
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#ifdef CONFIG_DEBUG_FS
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static char *dpcm_state_string(enum snd_soc_dpcm_state state)
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static const char *dpcm_state_string(enum snd_soc_dpcm_state state)
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{
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switch (state) {
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case SND_SOC_DPCM_STATE_NEW:
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