From 1894c59fdb63692f5ba2576875cc558b856935ca Mon Sep 17 00:00:00 2001 From: Tim Niemeyer Date: Mon, 5 May 2008 14:16:12 +0200 Subject: [ALSA] soc - Patch to add debug messages to the neo1973_wm8753 (GTA01) sound driver Signed-off-by: Tim Niemeyer Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 46 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 46 insertions(+) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 0e9d1c5f2484..e469186a407d 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -43,6 +43,14 @@ #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" +/* Debugging stuff */ +#define S3C24XX_SOC_NEO1973_WM8753_DEBUG 0 +#if S3C24XX_SOC_NEO1973_WM8753_DEBUG +#define DBG(x...) printk(KERN_DEBUG "s3c24xx-soc-neo1973-wm8753: " x) +#else +#define DBG(x...) +#endif + /* define the scenarios */ #define NEO_AUDIO_OFF 0 #define NEO_GSM_CALL_AUDIO_HANDSET 1 @@ -67,6 +75,8 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; + DBG("Entered %s\n", __func__); + iis_clkrate = s3c24xx_i2s_get_clockrate(); switch (params_rate(params)) { @@ -151,6 +161,8 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + DBG("Entered %s\n", __func__); + /* disable the PLL */ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); } @@ -172,6 +184,8 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, int ret = 0; unsigned long iis_clkrate; + DBG("Entered %s\n", __func__); + iis_clkrate = s3c24xx_i2s_get_clockrate(); if (params_rate(params) != 8000) @@ -213,6 +227,8 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + DBG("Entered %s\n", __func__); + /* disable the PLL */ return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); } @@ -233,6 +249,8 @@ static int neo1973_get_scenario(struct snd_kcontrol *kcontrol, static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) { + DBG("Entered %s\n", __func__); + switch (neo1973_scenario) { case NEO_AUDIO_OFF: snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); @@ -315,6 +333,8 @@ static int neo1973_set_scenario(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + DBG("Entered %s\n", __func__); + if (neo1973_scenario == ucontrol->value.integer.value[0]) return 0; @@ -327,6 +347,8 @@ static u8 lm4857_regs[4] = {0x00, 0x40, 0x80, 0xC0}; static void lm4857_write_regs(void) { + DBG("Entered %s\n", __func__); + if (i2c_master_send(i2c, lm4857_regs, 4) != 4) printk(KERN_ERR "lm4857: i2c write failed\n"); } @@ -338,6 +360,8 @@ static int lm4857_get_reg(struct snd_kcontrol *kcontrol, int shift = (kcontrol->private_value >> 8) & 0x0F; int mask = (kcontrol->private_value >> 16) & 0xFF; + DBG("Entered %s\n", __func__); + ucontrol->value.integer.value[0] = (lm4857_regs[reg] >> shift) & mask; return 0; } @@ -364,6 +388,8 @@ static int lm4857_get_mode(struct snd_kcontrol *kcontrol, { u8 value = lm4857_regs[LM4857_CTRL] & 0x0F; + DBG("Entered %s\n", __func__); + if (value) value -= 5; @@ -376,6 +402,8 @@ static int lm4857_set_mode(struct snd_kcontrol *kcontrol, { u8 value = ucontrol->value.integer.value[0]; + DBG("Entered %s\n", __func__); + if (value) value += 5; @@ -483,6 +511,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) { int i, err; + DBG("Entered %s\n", __func__); + /* set up NC codec pins */ snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); @@ -583,6 +613,8 @@ static int lm4857_amp_probe(struct i2c_adapter *adap, int addr, int kind) { int ret; + DBG("Entered %s\n", __func__); + client_template.adapter = adap; client_template.addr = addr; @@ -606,6 +638,8 @@ exit_err: static int lm4857_i2c_detach(struct i2c_client *client) { + DBG("Entered %s\n", __func__); + i2c_detach_client(client); kfree(client); return 0; @@ -613,6 +647,8 @@ static int lm4857_i2c_detach(struct i2c_client *client) static int lm4857_i2c_attach(struct i2c_adapter *adap) { + DBG("Entered %s\n", __func__); + return i2c_probe(adap, &addr_data, lm4857_amp_probe); } @@ -620,6 +656,8 @@ static u8 lm4857_state; static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) { + DBG("Entered %s\n", __func__); + dev_dbg(&dev->dev, "lm4857_suspend\n"); lm4857_state = lm4857_regs[LM4857_CTRL] & 0xf; if (lm4857_state) { @@ -631,6 +669,8 @@ static int lm4857_suspend(struct i2c_client *dev, pm_message_t state) static int lm4857_resume(struct i2c_client *dev) { + DBG("Entered %s\n", __func__); + if (lm4857_state) { lm4857_regs[LM4857_CTRL] |= (lm4857_state & 0x0f); lm4857_write_regs(); @@ -640,6 +680,8 @@ static int lm4857_resume(struct i2c_client *dev) static void lm4857_shutdown(struct i2c_client *dev) { + DBG("Entered %s\n", __func__); + dev_dbg(&dev->dev, "lm4857_shutdown\n"); lm4857_regs[LM4857_CTRL] &= 0xf0; lm4857_write_regs(); @@ -671,6 +713,8 @@ static int __init neo1973_init(void) { int ret; + DBG("Entered %s\n", __func__); + neo1973_snd_device = platform_device_alloc("soc-audio", -1); if (!neo1973_snd_device) return -ENOMEM; @@ -691,6 +735,8 @@ static int __init neo1973_init(void) static void __exit neo1973_exit(void) { + DBG("Entered %s\n", __func__); + i2c_del_driver(&lm4857_i2c_driver); platform_device_unregister(neo1973_snd_device); } -- cgit v1.2.1 From 8f3112d7a847c2933a42ce29f17899f585d09106 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 13 May 2008 14:58:03 +0200 Subject: [ALSA] soc - neo1973_wm8753 - Convert to bulk DAPM registration APIs Signed-off-by: Mark Brown Cc: Graeme Gregory Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 17 ++++++----------- 1 file changed, 6 insertions(+), 11 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index e469186a407d..79c5027273cb 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -425,8 +425,7 @@ static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = { }; -/* example machine audio_mapnections */ -static const char *audio_map[][3] = { +static const struct snd_soc_dapm_route dapm_routes[] = { /* Connections to the lm4857 amp */ {"Audio Out", NULL, "LOUT1"}, @@ -449,8 +448,6 @@ static const char *audio_map[][3] = { /* Connect the ALC pins */ {"ACIN", NULL, "ACOP"}, - - {NULL, NULL, NULL}, }; static const char *lm4857_mode[] = { @@ -526,8 +523,8 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) set_scenario_endpoints(codec, NEO_AUDIO_OFF); /* Add neo1973 specific widgets */ - for (i = 0; i < ARRAY_SIZE(wm8753_dapm_widgets); i++) - snd_soc_dapm_new_control(codec, &wm8753_dapm_widgets[i]); + snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets, + ARRAY_SIZE(wm8753_dapm_widgets)); /* add neo1973 specific controls */ for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) { @@ -538,11 +535,9 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) return err; } - /* set up neo1973 specific audio path audio_mapnects */ - for (i = 0; audio_map[i][0] != NULL; i++) { - snd_soc_dapm_connect_input(codec, audio_map[i][0], - audio_map[i][1], audio_map[i][2]); - } + /* set up neo1973 specific audio routes */ + err = snd_soc_dapm_add_routes(codec, dapm_routes, + ARRAY_SIZE(dapm_routes)); snd_soc_dapm_sync_endpoints(codec); return 0; -- cgit v1.2.1 From b2efbbfba24efc8456d68d5af42d082ab1c2febc Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 19 May 2008 12:30:58 +0200 Subject: [ALSA] ASoC: Remove in-code changelogs The overwhelming majority just say 'initial version' anyway. Signed-off-by: Mark Brown Acked-by: Ben Dooks Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 4 ---- sound/soc/s3c24xx/s3c2443-ac97.c | 3 --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 ----- sound/soc/s3c24xx/s3c24xx-pcm.c | 4 ---- sound/soc/s3c24xx/smdk2443_wm9710.c | 3 --- 5 files changed, 19 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 79c5027273cb..c1a0161bc72e 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -10,10 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 20th Jan 2007 Initial version. - * 05th Feb 2007 Rename all to Neo1973 - * */ #include diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index e81d9a6c83da..0eed140dcd9b 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -10,9 +10,6 @@ * 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. - * - * Revision history - * 21st Mar 2007 Initial Version */ #include diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 1ed6afd45459..ddf87246c77b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -12,11 +12,6 @@ * 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. - * - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index 7806ae614617..ef599745159c 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -12,10 +12,6 @@ * 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. - * - * Revision history - * 11th Dec 2006 Merged with Simtec driver - * 10th Nov 2006 Initial version. */ #include diff --git a/sound/soc/s3c24xx/smdk2443_wm9710.c b/sound/soc/s3c24xx/smdk2443_wm9710.c index b4a56302b9ab..8515d6ff03f2 100644 --- a/sound/soc/s3c24xx/smdk2443_wm9710.c +++ b/sound/soc/s3c24xx/smdk2443_wm9710.c @@ -10,9 +10,6 @@ * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * - * Revision history - * 8th Mar 2007 Initial version. - * */ #include -- cgit v1.2.1 From 33e5b22285f63ede858c00456f3ffbc2ea79d6cf Mon Sep 17 00:00:00 2001 From: Werner Almesberger Date: Mon, 14 Apr 2008 14:26:44 +0200 Subject: [ALSA] soc - Fix s3c24xx-i2s LR sync while timer ticks are disabled When timer ticks are disabled when calling sound/soc/s3c24xx/s3c24xx-i2s.c:s3c24xx_snd_lrsync and the LR signal never happens, we loop forever. This has been observed in the following call chain: snd_pcm_common_ioctl1 -> snd_pcm_action_lock_irq -> snd_pcm_action_single -> snd_pcm_do_resume -> soc_pcm_trigger -> s3c24xx_i2s_trigger The patch below changes the timeout mechanism to use udelay, which doesn't need timer ticks. Signed-off-by: Werner Almesberger Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/s3c24xx-i2s.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ddf87246c77b..4c52f7946d9e 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -175,7 +175,7 @@ static void s3c24xx_snd_rxctrl(int on) static int s3c24xx_snd_lrsync(void) { u32 iiscon; - unsigned long timeout = jiffies + msecs_to_jiffies(5); + int timeout = 50; /* 5ms */ DBG("Entered %s\n", __func__); @@ -184,8 +184,9 @@ static int s3c24xx_snd_lrsync(void) if (iiscon & S3C2410_IISCON_LRINDEX) break; - if (time_after(jiffies, timeout)) + if (!timeout--) return -ETIMEDOUT; + udelay(100); } return 0; -- cgit v1.2.1 From 89fe5117928b2c1272c9376362131ded561c91ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Fri, 23 May 2008 16:10:37 +0200 Subject: sound: Convert to menuconfig Convert menu in sound Kconfig files to menuconfig and if. Signed-off-by: Takashi Iwai --- sound/soc/s3c24xx/Kconfig | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig index 1f6dbfc4caa8..b9f2353effeb 100644 --- a/sound/soc/s3c24xx/Kconfig +++ b/sound/soc/s3c24xx/Kconfig @@ -1,7 +1,6 @@ config SND_S3C24XX_SOC tristate "SoC Audio for the Samsung S3C24XX chips" - depends on ARCH_S3C2410 && SND_SOC - select SND_PCM + depends on ARCH_S3C2410 help Say Y or M if you want to add support for codecs attached to the S3C24XX AC97, I2S or SSP interface. You will also need @@ -16,7 +15,6 @@ config SND_S3C2412_SOC_I2S config SND_S3C2443_SOC_AC97 tristate select AC97_BUS - select SND_AC97_CODEC select SND_SOC_AC97_BUS config SND_S3C24XX_SOC_NEO1973_WM8753 -- cgit v1.2.1 From bdb92876f0a9d2b431199e385732ede89ff0b97d Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 11 Jun 2008 13:47:10 +0100 Subject: ALSA: ASoC: Pass the DAI being configured into CPU DAI probe and remove This allows per-DAI initialisation to be done by the CPU DAI drivers. Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/s3c2412-i2s.c | 3 ++- sound/soc/s3c24xx/s3c2443-ac97.c | 6 ++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 3 ++- 3 files changed, 8 insertions(+), 4 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c4a46dd589b3..c463a82dec3a 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -601,7 +601,8 @@ struct clk *s3c2412_get_iisclk(void) EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); -static int s3c2412_i2s_probe(struct platform_device *pdev) +static int s3c2412_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__); diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 0eed140dcd9b..533565b61b2f 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -209,7 +209,8 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { .dma_size = 4, }; -static int s3c2443_ac97_probe(struct platform_device *pdev) +static int s3c2443_ac97_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { int ret; u32 ac_glbctrl; @@ -260,7 +261,8 @@ static int s3c2443_ac97_probe(struct platform_device *pdev) return ret; } -static void s3c2443_ac97_remove(struct platform_device *pdev) +static void s3c2443_ac97_remove(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 4c52f7946d9e..42e96b5ff825 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -377,7 +377,8 @@ u32 s3c24xx_i2s_get_clockrate(void) } EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); -static int s3c24xx_i2s_probe(struct platform_device *pdev) +static int s3c24xx_i2s_probe(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) { DBG("Entered %s\n", __func__); -- cgit v1.2.1 From a2e31a595ea74120a5e9de91fce56dd835edff94 Mon Sep 17 00:00:00 2001 From: Mike Montour Date: Wed, 11 Jun 2008 13:47:14 +0100 Subject: ALSA: ASoC: Add TLV information to the LM4857 controls on the GTA01 Signed-off-by: Mike Montour Signed-off-by: Mark Brown Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 16 ++++++++++------ 1 file changed, 10 insertions(+), 6 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index c1a0161bc72e..34851238dea0 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -22,6 +22,7 @@ #include #include #include +#include #include #include @@ -474,13 +475,16 @@ static const struct soc_enum neo_scenario_enum[] = { SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(neo_scenarios), neo_scenarios), }; +static const DECLARE_TLV_DB_SCALE(stereo_tlv, -4050, 150, 0); +static const DECLARE_TLV_DB_SCALE(mono_tlv, -3450, 150, 0); + static const struct snd_kcontrol_new wm8753_neo1973_controls[] = { - SOC_SINGLE_EXT("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), - SOC_SINGLE_EXT("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, - lm4857_get_reg, lm4857_set_reg), + SOC_SINGLE_EXT_TLV("Amp Left Playback Volume", LM4857_LVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Right Playback Volume", LM4857_RVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, stereo_tlv), + SOC_SINGLE_EXT_TLV("Amp Mono Playback Volume", LM4857_MVOL, 0, 31, 0, + lm4857_get_reg, lm4857_set_reg, mono_tlv), SOC_ENUM_EXT("Amp Mode", lm4857_mode_enum[0], lm4857_get_mode, lm4857_set_mode), SOC_ENUM_EXT("Neo Mode", neo_scenario_enum[0], -- cgit v1.2.1 From a5302181e5321664047f75715242aac4e0bbd17c Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 13:35:17 +0100 Subject: ALSA: asoc: core - refactored DAPM pin control API. Refactored snd_soc_dapm_set_endpoint() to snd_soc_dapm_enable_pin() and snd_soc_dapm_disable_pin(). Renamed snd_soc_dapm_sync_endpoints() to snd_soc_dapm_sync(). Renamed snd_soc_dapm_get_endpoint_status() to snd_soc_dapm_get_pin_status(). Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 116 ++++++++++++++++++------------------- 1 file changed, 58 insertions(+), 58 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 34851238dea0..f053e85ff608 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -250,77 +250,77 @@ static int set_scenario_endpoints(struct snd_soc_codec *codec, int scenario) switch (neo1973_scenario) { case NEO_AUDIO_OFF: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_GSM_CALL_AUDIO_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 1); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line Out"); + snd_soc_dapm_enable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_SPEAKERS: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_STEREO_TO_HEADPHONES: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 1); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_enable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HANDSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 1); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_HEADSET: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 1); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; case NEO_CAPTURE_BLUETOOTH: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); break; default: - snd_soc_dapm_set_endpoint(codec, "Audio Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line Out", 0); - snd_soc_dapm_set_endpoint(codec, "GSM Line In", 0); - snd_soc_dapm_set_endpoint(codec, "Headset Mic", 0); - snd_soc_dapm_set_endpoint(codec, "Call Mic", 0); + snd_soc_dapm_disable_pin(codec, "Audio Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line Out"); + snd_soc_dapm_disable_pin(codec, "GSM Line In"); + snd_soc_dapm_disable_pin(codec, "Headset Mic"); + snd_soc_dapm_disable_pin(codec, "Call Mic"); } - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } @@ -511,12 +511,12 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) DBG("Entered %s\n", __func__); /* set up NC codec pins */ - snd_soc_dapm_set_endpoint(codec, "LOUT2", 0); - snd_soc_dapm_set_endpoint(codec, "ROUT2", 0); - snd_soc_dapm_set_endpoint(codec, "OUT3", 0); - snd_soc_dapm_set_endpoint(codec, "OUT4", 0); - snd_soc_dapm_set_endpoint(codec, "LINE1", 0); - snd_soc_dapm_set_endpoint(codec, "LINE2", 0); + snd_soc_dapm_disable_pin(codec, "LOUT2"); + snd_soc_dapm_disable_pin(codec, "ROUT2"); + snd_soc_dapm_disable_pin(codec, "OUT3"); + snd_soc_dapm_disable_pin(codec, "OUT4"); + snd_soc_dapm_disable_pin(codec, "LINE1"); + snd_soc_dapm_disable_pin(codec, "LINE2"); /* set endpoints to default mode */ @@ -539,7 +539,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) err = snd_soc_dapm_add_routes(codec, dapm_routes, ARRAY_SIZE(dapm_routes)); - snd_soc_dapm_sync_endpoints(codec); + snd_soc_dapm_sync(codec); return 0; } -- cgit v1.2.1 From 1992a6fbd929196aebe95e0e7b04c4da66c3bfec Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Mon, 7 Jul 2008 16:08:24 +0100 Subject: ALSA: asoc: s3c24xx - merge structs snd_soc_codec_dai and snd_soc_cpu_dai. This patch merges struct snd_soc_codec_dai and struct snd_soc_cpu_dai into struct snd_soc_dai for the S3C24xx platform. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 12 ++++++------ sound/soc/s3c24xx/s3c2412-i2s.c | 14 +++++++------- sound/soc/s3c24xx/s3c2412-i2s.h | 2 +- sound/soc/s3c24xx/s3c2443-ac97.c | 10 +++++----- sound/soc/s3c24xx/s3c24xx-ac97.h | 2 +- sound/soc/s3c24xx/s3c24xx-i2s.c | 14 +++++++------- sound/soc/s3c24xx/s3c24xx-i2s.h | 2 +- sound/soc/s3c24xx/s3c24xx-pcm.c | 2 +- 8 files changed, 29 insertions(+), 29 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index f053e85ff608..51a4ce3dbd19 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -66,8 +66,8 @@ static int neo1973_hifi_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_codec_dai *codec_dai = rtd->dai->codec_dai; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; unsigned int pll_out = 0, bclk = 0; int ret = 0; unsigned long iis_clkrate; @@ -156,7 +156,7 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; DBG("Entered %s\n", __func__); @@ -176,7 +176,7 @@ static int neo1973_voice_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_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; unsigned int pcmdiv = 0; int ret = 0; unsigned long iis_clkrate; @@ -222,7 +222,7 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; DBG("Entered %s\n", __func__); @@ -546,7 +546,7 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec) /* * BT Codec DAI */ -static struct snd_soc_cpu_dai bt_dai = { +static struct snd_soc_dai bt_dai = { .name = "Bluetooth", .id = 0, .type = SND_SOC_DAI_PCM, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c index c463a82dec3a..ee4676ed1283 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.c +++ b/sound/soc/s3c24xx/s3c2412-i2s.c @@ -295,7 +295,7 @@ static inline int s3c2412_snd_is_clkmaster(void) /* * Set S3C2412 I2S DAI format */ -static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -500,7 +500,7 @@ EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); /* * Set S3C2412 Clock source */ -static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); @@ -528,7 +528,7 @@ static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C2412 Clock dividers */ -static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -602,7 +602,7 @@ EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); static int s3c2412_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -648,7 +648,7 @@ static int s3c2412_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c2412_i2s_suspend(struct platform_device *dev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; u32 iismod; @@ -676,7 +676,7 @@ static int s3c2412_i2s_suspend(struct platform_device *dev, } static int s3c2412_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { struct s3c2412_i2s_info *i2s = &s3c2412_i2s; @@ -708,7 +708,7 @@ static int s3c2412_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c2412_i2s_dai = { +struct snd_soc_dai s3c2412_i2s_dai = { .name = "s3c2412-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c2412-i2s.h b/sound/soc/s3c24xx/s3c2412-i2s.h index 27f48e1ffa86..aac08a25e541 100644 --- a/sound/soc/s3c24xx/s3c2412-i2s.h +++ b/sound/soc/s3c24xx/s3c2412-i2s.h @@ -24,7 +24,7 @@ extern struct clk *s3c2412_get_iisclk(void); -extern struct snd_soc_cpu_dai s3c2412_i2s_dai; +extern struct snd_soc_dai s3c2412_i2s_dai; struct s3c2412_rate_calc { unsigned int clk_div; /* for prescaler */ diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c index 533565b61b2f..783349b7fede 100644 --- a/sound/soc/s3c24xx/s3c2443-ac97.c +++ b/sound/soc/s3c24xx/s3c2443-ac97.c @@ -210,7 +210,7 @@ static struct s3c24xx_pcm_dma_params s3c2443_ac97_mic_mono_in = { }; static int s3c2443_ac97_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { int ret; u32 ac_glbctrl; @@ -262,7 +262,7 @@ static int s3c2443_ac97_probe(struct platform_device *pdev, } static void s3c2443_ac97_remove(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); @@ -274,7 +274,7 @@ static int s3c2443_ac97_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_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) cpu_dai->dma_data = &s3c2443_ac97_pcm_stereo_out; @@ -316,7 +316,7 @@ static int s3c2443_ac97_hw_mic_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_cpu_dai *cpu_dai = rtd->dai->cpu_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) return -ENODEV; @@ -352,7 +352,7 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream, SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) -struct snd_soc_cpu_dai s3c2443_ac97_dai[] = { +struct snd_soc_dai s3c2443_ac97_dai[] = { { .name = "s3c2443-ac97", .id = 0, diff --git a/sound/soc/s3c24xx/s3c24xx-ac97.h b/sound/soc/s3c24xx/s3c24xx-ac97.h index bf03e8ed16c3..a96dcadf28b4 100644 --- a/sound/soc/s3c24xx/s3c24xx-ac97.h +++ b/sound/soc/s3c24xx/s3c24xx-ac97.h @@ -26,6 +26,6 @@ #define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 #endif -extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; +extern struct snd_soc_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 42e96b5ff825..397524282b57 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -205,7 +205,7 @@ static inline int s3c24xx_snd_is_clkmaster(void) /* * Set S3C24xx I2S DAI format */ -static int s3c24xx_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { u32 iismod; @@ -313,7 +313,7 @@ exit_err: /* * Set S3C24xx Clock source */ -static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { u32 iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -339,7 +339,7 @@ static int s3c24xx_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, /* * Set S3C24xx Clock dividers */ -static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, +static int s3c24xx_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { u32 reg; @@ -378,7 +378,7 @@ u32 s3c24xx_i2s_get_clockrate(void) EXPORT_SYMBOL_GPL(s3c24xx_i2s_get_clockrate); static int s3c24xx_i2s_probe(struct platform_device *pdev, - struct snd_soc_cpu_dai *dai) + struct snd_soc_dai *dai) { DBG("Entered %s\n", __func__); @@ -411,7 +411,7 @@ static int s3c24xx_i2s_probe(struct platform_device *pdev, #ifdef CONFIG_PM static int s3c24xx_i2s_suspend(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); @@ -426,7 +426,7 @@ static int s3c24xx_i2s_suspend(struct platform_device *pdev, } static int s3c24xx_i2s_resume(struct platform_device *pdev, - struct snd_soc_cpu_dai *cpu_dai) + struct snd_soc_dai *cpu_dai) { DBG("Entered %s\n", __func__); clk_enable(s3c24xx_i2s.iis_clk); @@ -449,7 +449,7 @@ static int s3c24xx_i2s_resume(struct platform_device *pdev, SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) -struct snd_soc_cpu_dai s3c24xx_i2s_dai = { +struct snd_soc_dai s3c24xx_i2s_dai = { .name = "s3c24xx-i2s", .id = 0, .type = SND_SOC_DAI_I2S, diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.h b/sound/soc/s3c24xx/s3c24xx-i2s.h index 537b4ecce8a3..726d91cf4e1c 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.h +++ b/sound/soc/s3c24xx/s3c24xx-i2s.h @@ -32,6 +32,6 @@ u32 s3c24xx_i2s_get_clockrate(void); -extern struct snd_soc_cpu_dai s3c24xx_i2s_dai; +extern struct snd_soc_dai s3c24xx_i2s_dai; #endif /*S3C24XXI2S_H_*/ diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c index ef599745159c..cef79b34dc6f 100644 --- a/sound/soc/s3c24xx/s3c24xx-pcm.c +++ b/sound/soc/s3c24xx/s3c24xx-pcm.c @@ -429,7 +429,7 @@ static void s3c24xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 s3c24xx_pcm_dmamask = DMA_32BIT_MASK; static int s3c24xx_pcm_new(struct snd_card *card, - struct snd_soc_codec_dai *dai, struct snd_pcm *pcm) + struct snd_soc_dai *dai, struct snd_pcm *pcm) { int ret = 0; -- cgit v1.2.1 From 64105cfd65df74fdf82c1d053b2c9953304a94ea Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Tue, 8 Jul 2008 13:19:18 +0100 Subject: ALSA: asoc: machines - add Digital Audio Interface (DAI) control functions. This patch adds several functions for DAI control and config and replaces the current method of calling function pointers within the DAI struct within the machine drivers. Signed-off-by: Liam Girdwood Signed-off-by: Takashi Iwai Signed-off-by: Jaroslav Kysela --- sound/soc/s3c24xx/neo1973_wm8753.c | 26 +++++++++++++------------- 1 file changed, 13 insertions(+), 13 deletions(-) (limited to 'sound/soc/s3c24xx') diff --git a/sound/soc/s3c24xx/neo1973_wm8753.c b/sound/soc/s3c24xx/neo1973_wm8753.c index 51a4ce3dbd19..4d7a9aa15f1a 100644 --- a/sound/soc/s3c24xx/neo1973_wm8753.c +++ b/sound/soc/s3c24xx/neo1973_wm8753.c @@ -108,44 +108,44 @@ static int neo1973_hifi_hw_params(struct snd_pcm_substream *substream, } /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set cpu DAI configuration */ - ret = cpu_dai->dai_ops.set_fmt(cpu_dai, + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_MCLK, pll_out, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set MCLK division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK, S3C2410_IISMOD_32FS); if (ret < 0) return ret; /* set codec BCLK division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_BCLKDIV, bclk); if (ret < 0) return ret; /* set prescaler division for sample rate */ - ret = cpu_dai->dai_ops.set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER, S3C24XX_PRESCALE(4, 4)); if (ret < 0) return ret; /* codec PLL input is PCLK/4 */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, iis_clkrate / 4, pll_out); if (ret < 0) return ret; @@ -161,7 +161,7 @@ static int neo1973_hifi_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL1, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0); } /* @@ -194,24 +194,24 @@ static int neo1973_voice_hw_params(struct snd_pcm_substream *substream, /* todo: gg check mode (DSP_B) against CSR datasheet */ /* set codec DAI configuration */ - ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS); if (ret < 0) return ret; /* set the codec system clock for DAC and ADC */ - ret = codec_dai->dai_ops.set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, + ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK, 12288000, SND_SOC_CLOCK_IN); if (ret < 0) return ret; /* set codec PCM division for sample rate */ - ret = codec_dai->dai_ops.set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); + ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV, pcmdiv); if (ret < 0) return ret; /* configue and enable PLL for 12.288MHz output */ - ret = codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, + ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, iis_clkrate / 4, 12288000); if (ret < 0) return ret; @@ -227,7 +227,7 @@ static int neo1973_voice_hw_free(struct snd_pcm_substream *substream) DBG("Entered %s\n", __func__); /* disable the PLL */ - return codec_dai->dai_ops.set_pll(codec_dai, WM8753_PLL2, 0, 0); + return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0); } static struct snd_soc_ops neo1973_voice_ops = { -- cgit v1.2.1