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-rw-r--r--sound/soc/s3c24xx/Kconfig35
-rw-r--r--sound/soc/s3c24xx/Makefile9
-rw-r--r--sound/soc/s3c24xx/neo1973_gta02_wm8753.c498
-rw-r--r--sound/soc/s3c24xx/s3c-i2s-v2.c17
-rw-r--r--sound/soc/s3c24xx/s3c2443-ac97.c20
-rw-r--r--sound/soc/s3c24xx/s3c24xx-i2s.c5
-rw-r--r--sound/soc/s3c24xx/s3c24xx-pcm.c2
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec.c394
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec.h22
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec_hermes.c153
-rw-r--r--sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c137
11 files changed, 1280 insertions, 12 deletions
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index df494d1e346f..923428fc1adb 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,6 +1,7 @@
1config SND_S3C24XX_SOC 1config SND_S3C24XX_SOC
2 tristate "SoC Audio for the Samsung S3CXXXX chips" 2 tristate "SoC Audio for the Samsung S3CXXXX chips"
3 depends on ARCH_S3C2410 3 depends on ARCH_S3C2410 || ARCH_S3C64XX
4 select S3C64XX_DMA if ARCH_S3C64XX
4 help 5 help
5 Say Y or M if you want to add support for codecs attached to 6 Say Y or M if you want to add support for codecs attached to
6 the S3C24XX AC97 or I2S interfaces. You will also need to 7 the S3C24XX AC97 or I2S interfaces. You will also need to
@@ -38,6 +39,15 @@ config SND_S3C24XX_SOC_NEO1973_WM8753
38 Say Y if you want to add support for SoC audio on smdk2440 39 Say Y if you want to add support for SoC audio on smdk2440
39 with the WM8753. 40 with the WM8753.
40 41
42config SND_S3C24XX_SOC_NEO1973_GTA02_WM8753
43 tristate "Audio support for the Openmoko Neo FreeRunner (GTA02)"
44 depends on SND_S3C24XX_SOC && MACH_NEO1973_GTA02
45 select SND_S3C24XX_SOC_I2S
46 select SND_SOC_WM8753
47 help
48 This driver provides audio support for the Openmoko Neo FreeRunner
49 smartphone.
50
41config SND_S3C24XX_SOC_JIVE_WM8750 51config SND_S3C24XX_SOC_JIVE_WM8750
42 tristate "SoC I2S Audio support for Jive" 52 tristate "SoC I2S Audio support for Jive"
43 depends on SND_S3C24XX_SOC && MACH_JIVE 53 depends on SND_S3C24XX_SOC && MACH_JIVE
@@ -57,7 +67,7 @@ config SND_S3C24XX_SOC_SMDK2443_WM9710
57 67
58config SND_S3C24XX_SOC_LN2440SBC_ALC650 68config SND_S3C24XX_SOC_LN2440SBC_ALC650
59 tristate "SoC AC97 Audio support for LN2440SBC - ALC650" 69 tristate "SoC AC97 Audio support for LN2440SBC - ALC650"
60 depends on SND_S3C24XX_SOC 70 depends on SND_S3C24XX_SOC && ARCH_S3C2410
61 select SND_S3C2443_SOC_AC97 71 select SND_S3C2443_SOC_AC97
62 select SND_SOC_AC97_CODEC 72 select SND_SOC_AC97_CODEC
63 help 73 help
@@ -66,7 +76,26 @@ config SND_S3C24XX_SOC_LN2440SBC_ALC650
66 76
67config SND_S3C24XX_SOC_S3C24XX_UDA134X 77config SND_S3C24XX_SOC_S3C24XX_UDA134X
68 tristate "SoC I2S Audio support UDA134X wired to a S3C24XX" 78 tristate "SoC I2S Audio support UDA134X wired to a S3C24XX"
69 depends on SND_S3C24XX_SOC 79 depends on SND_S3C24XX_SOC && ARCH_S3C2410
70 select SND_S3C24XX_SOC_I2S 80 select SND_S3C24XX_SOC_I2S
71 select SND_SOC_L3 81 select SND_SOC_L3
72 select SND_SOC_UDA134X 82 select SND_SOC_UDA134X
83
84config SND_S3C24XX_SOC_SIMTEC
85 tristate
86 help
87 Internal node for common S3C24XX/Simtec suppor
88
89config SND_S3C24XX_SOC_SIMTEC_TLV320AIC23
90 tristate "SoC I2S Audio support for TLV320AIC23 on Simtec boards"
91 depends on SND_S3C24XX_SOC && ARCH_S3C2410
92 select SND_S3C24XX_SOC_I2S
93 select SND_SOC_TLV320AIC23
94 select SND_S3C24XX_SOC_SIMTEC
95
96config SND_S3C24XX_SOC_SIMTEC_HERMES
97 tristate "SoC I2S Audio support for Simtec Hermes board"
98 depends on SND_S3C24XX_SOC && ARCH_S3C2410
99 select SND_S3C24XX_SOC_I2S
100 select SND_SOC_TLV320AIC3X
101 select SND_S3C24XX_SOC_SIMTEC
diff --git a/sound/soc/s3c24xx/Makefile b/sound/soc/s3c24xx/Makefile
index 07a93a2ebe5f..99f5a7dd3fc6 100644
--- a/sound/soc/s3c24xx/Makefile
+++ b/sound/soc/s3c24xx/Makefile
@@ -16,12 +16,21 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o
16# S3C24XX Machine Support 16# S3C24XX Machine Support
17snd-soc-jive-wm8750-objs := jive_wm8750.o 17snd-soc-jive-wm8750-objs := jive_wm8750.o
18snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o 18snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o
19snd-soc-neo1973-gta02-wm8753-objs := neo1973_gta02_wm8753.o
19snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o 20snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o
20snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o 21snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o
21snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o 22snd-soc-s3c24xx-uda134x-objs := s3c24xx_uda134x.o
23snd-soc-s3c24xx-simtec-objs := s3c24xx_simtec.o
24snd-soc-s3c24xx-simtec-hermes-objs := s3c24xx_simtec_hermes.o
25snd-soc-s3c24xx-simtec-tlv320aic23-objs := s3c24xx_simtec_tlv320aic23.o
22 26
23obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o 27obj-$(CONFIG_SND_S3C24XX_SOC_JIVE_WM8750) += snd-soc-jive-wm8750.o
24obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o 28obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o
29obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_GTA02_WM8753) += snd-soc-neo1973-gta02-wm8753.o
25obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o 30obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o
26obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o 31obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o
27obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o 32obj-$(CONFIG_SND_S3C24XX_SOC_S3C24XX_UDA134X) += snd-soc-s3c24xx-uda134x.o
33obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC) += snd-soc-s3c24xx-simtec.o
34obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_HERMES) += snd-soc-s3c24xx-simtec-hermes.o
35obj-$(CONFIG_SND_S3C24XX_SOC_SIMTEC_TLV320AIC23) += snd-soc-s3c24xx-simtec-tlv320aic23.o
36
diff --git a/sound/soc/s3c24xx/neo1973_gta02_wm8753.c b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
new file mode 100644
index 000000000000..0c52e36ddd87
--- /dev/null
+++ b/sound/soc/s3c24xx/neo1973_gta02_wm8753.c
@@ -0,0 +1,498 @@
1/*
2 * neo1973_gta02_wm8753.c -- SoC audio for Openmoko Freerunner(GTA02)
3 *
4 * Copyright 2007 Openmoko Inc
5 * Author: Graeme Gregory <graeme@openmoko.org>
6 * Copyright 2007 Wolfson Microelectronics PLC.
7 * Author: Graeme Gregory <linux@wolfsonmicro.com>
8 * Copyright 2009 Wolfson Microelectronics
9 *
10 * This program is free software; you can redistribute it and/or modify it
11 * under the terms of the GNU General Public License as published by the
12 * Free Software Foundation; either version 2 of the License, or (at your
13 * option) any later version.
14 */
15
16#include <linux/module.h>
17#include <linux/moduleparam.h>
18#include <linux/timer.h>
19#include <linux/interrupt.h>
20#include <linux/platform_device.h>
21#include <linux/gpio.h>
22#include <sound/core.h>
23#include <sound/pcm.h>
24#include <sound/soc.h>
25#include <sound/soc-dapm.h>
26
27#include <asm/mach-types.h>
28
29#include <plat/regs-iis.h>
30
31#include <mach/regs-clock.h>
32#include <asm/io.h>
33#include <mach/gta02.h>
34#include "../codecs/wm8753.h"
35#include "s3c24xx-pcm.h"
36#include "s3c24xx-i2s.h"
37
38static struct snd_soc_card neo1973_gta02;
39
40static int neo1973_gta02_hifi_hw_params(struct snd_pcm_substream *substream,
41 struct snd_pcm_hw_params *params)
42{
43 struct snd_soc_pcm_runtime *rtd = substream->private_data;
44 struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
45 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
46 unsigned int pll_out = 0, bclk = 0;
47 int ret = 0;
48 unsigned long iis_clkrate;
49
50 iis_clkrate = s3c24xx_i2s_get_clockrate();
51
52 switch (params_rate(params)) {
53 case 8000:
54 case 16000:
55 pll_out = 12288000;
56 break;
57 case 48000:
58 bclk = WM8753_BCLK_DIV_4;
59 pll_out = 12288000;
60 break;
61 case 96000:
62 bclk = WM8753_BCLK_DIV_2;
63 pll_out = 12288000;
64 break;
65 case 11025:
66 bclk = WM8753_BCLK_DIV_16;
67 pll_out = 11289600;
68 break;
69 case 22050:
70 bclk = WM8753_BCLK_DIV_8;
71 pll_out = 11289600;
72 break;
73 case 44100:
74 bclk = WM8753_BCLK_DIV_4;
75 pll_out = 11289600;
76 break;
77 case 88200:
78 bclk = WM8753_BCLK_DIV_2;
79 pll_out = 11289600;
80 break;
81 }
82
83 /* set codec DAI configuration */
84 ret = snd_soc_dai_set_fmt(codec_dai,
85 SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
86 SND_SOC_DAIFMT_CBM_CFM);
87 if (ret < 0)
88 return ret;
89
90 /* set cpu DAI configuration */
91 ret = snd_soc_dai_set_fmt(cpu_dai,
92 SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
93 SND_SOC_DAIFMT_CBM_CFM);
94 if (ret < 0)
95 return ret;
96
97 /* set the codec system clock for DAC and ADC */
98 ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_MCLK, pll_out,
99 SND_SOC_CLOCK_IN);
100 if (ret < 0)
101 return ret;
102
103 /* set MCLK division for sample rate */
104 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_MCLK,
105 S3C2410_IISMOD_32FS);
106 if (ret < 0)
107 return ret;
108
109 /* set codec BCLK division for sample rate */
110 ret = snd_soc_dai_set_clkdiv(codec_dai,
111 WM8753_BCLKDIV, bclk);
112 if (ret < 0)
113 return ret;
114
115 /* set prescaler division for sample rate */
116 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
117 S3C24XX_PRESCALE(4, 4));
118 if (ret < 0)
119 return ret;
120
121 /* codec PLL input is PCLK/4 */
122 ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL1,
123 iis_clkrate / 4, pll_out);
124 if (ret < 0)
125 return ret;
126
127 return 0;
128}
129
130static int neo1973_gta02_hifi_hw_free(struct snd_pcm_substream *substream)
131{
132 struct snd_soc_pcm_runtime *rtd = substream->private_data;
133 struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
134
135 /* disable the PLL */
136 return snd_soc_dai_set_pll(codec_dai, WM8753_PLL1, 0, 0);
137}
138
139/*
140 * Neo1973 WM8753 HiFi DAI opserations.
141 */
142static struct snd_soc_ops neo1973_gta02_hifi_ops = {
143 .hw_params = neo1973_gta02_hifi_hw_params,
144 .hw_free = neo1973_gta02_hifi_hw_free,
145};
146
147static int neo1973_gta02_voice_hw_params(
148 struct snd_pcm_substream *substream,
149 struct snd_pcm_hw_params *params)
150{
151 struct snd_soc_pcm_runtime *rtd = substream->private_data;
152 struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
153 unsigned int pcmdiv = 0;
154 int ret = 0;
155 unsigned long iis_clkrate;
156
157 iis_clkrate = s3c24xx_i2s_get_clockrate();
158
159 if (params_rate(params) != 8000)
160 return -EINVAL;
161 if (params_channels(params) != 1)
162 return -EINVAL;
163
164 pcmdiv = WM8753_PCM_DIV_6; /* 2.048 MHz */
165
166 /* todo: gg check mode (DSP_B) against CSR datasheet */
167 /* set codec DAI configuration */
168 ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B |
169 SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
170 if (ret < 0)
171 return ret;
172
173 /* set the codec system clock for DAC and ADC */
174 ret = snd_soc_dai_set_sysclk(codec_dai, WM8753_PCMCLK,
175 12288000, SND_SOC_CLOCK_IN);
176 if (ret < 0)
177 return ret;
178
179 /* set codec PCM division for sample rate */
180 ret = snd_soc_dai_set_clkdiv(codec_dai, WM8753_PCMDIV,
181 pcmdiv);
182 if (ret < 0)
183 return ret;
184
185 /* configue and enable PLL for 12.288MHz output */
186 ret = snd_soc_dai_set_pll(codec_dai, WM8753_PLL2,
187 iis_clkrate / 4, 12288000);
188 if (ret < 0)
189 return ret;
190
191 return 0;
192}
193
194static int neo1973_gta02_voice_hw_free(struct snd_pcm_substream *substream)
195{
196 struct snd_soc_pcm_runtime *rtd = substream->private_data;
197 struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
198
199 /* disable the PLL */
200 return snd_soc_dai_set_pll(codec_dai, WM8753_PLL2, 0, 0);
201}
202
203static struct snd_soc_ops neo1973_gta02_voice_ops = {
204 .hw_params = neo1973_gta02_voice_hw_params,
205 .hw_free = neo1973_gta02_voice_hw_free,
206};
207
208#define LM4853_AMP 1
209#define LM4853_SPK 2
210
211static u8 lm4853_state;
212
213/* This has no effect, it exists only to maintain compatibility with
214 * existing ALSA state files.
215 */
216static int lm4853_set_state(struct snd_kcontrol *kcontrol,
217 struct snd_ctl_elem_value *ucontrol)
218{
219 int val = ucontrol->value.integer.value[0];
220
221 if (val)
222 lm4853_state |= LM4853_AMP;
223 else
224 lm4853_state &= ~LM4853_AMP;
225
226 return 0;
227}
228
229static int lm4853_get_state(struct snd_kcontrol *kcontrol,
230 struct snd_ctl_elem_value *ucontrol)
231{
232 ucontrol->value.integer.value[0] = lm4853_state & LM4853_AMP;
233
234 return 0;
235}
236
237static int lm4853_set_spk(struct snd_kcontrol *kcontrol,
238 struct snd_ctl_elem_value *ucontrol)
239{
240 int val = ucontrol->value.integer.value[0];
241
242 if (val) {
243 lm4853_state |= LM4853_SPK;
244 gpio_set_value(GTA02_GPIO_HP_IN, 0);
245 } else {
246 lm4853_state &= ~LM4853_SPK;
247 gpio_set_value(GTA02_GPIO_HP_IN, 1);
248 }
249
250 return 0;
251}
252
253static int lm4853_get_spk(struct snd_kcontrol *kcontrol,
254 struct snd_ctl_elem_value *ucontrol)
255{
256 ucontrol->value.integer.value[0] = (lm4853_state & LM4853_SPK) >> 1;
257
258 return 0;
259}
260
261static int lm4853_event(struct snd_soc_dapm_widget *w,
262 struct snd_kcontrol *k,
263 int event)
264{
265 gpio_set_value(GTA02_GPIO_AMP_SHUT, SND_SOC_DAPM_EVENT_OFF(value));
266
267 return 0;
268}
269
270static const struct snd_soc_dapm_widget wm8753_dapm_widgets[] = {
271 SND_SOC_DAPM_SPK("Stereo Out", lm4853_event),
272 SND_SOC_DAPM_LINE("GSM Line Out", NULL),
273 SND_SOC_DAPM_LINE("GSM Line In", NULL),
274 SND_SOC_DAPM_MIC("Headset Mic", NULL),
275 SND_SOC_DAPM_MIC("Handset Mic", NULL),
276 SND_SOC_DAPM_SPK("Handset Spk", NULL),
277};
278
279
280/* example machine audio_mapnections */
281static const struct snd_soc_dapm_route audio_map[] = {
282
283 /* Connections to the lm4853 amp */
284 {"Stereo Out", NULL, "LOUT1"},
285 {"Stereo Out", NULL, "ROUT1"},
286
287 /* Connections to the GSM Module */
288 {"GSM Line Out", NULL, "MONO1"},
289 {"GSM Line Out", NULL, "MONO2"},
290 {"RXP", NULL, "GSM Line In"},
291 {"RXN", NULL, "GSM Line In"},
292
293 /* Connections to Headset */
294 {"MIC1", NULL, "Mic Bias"},
295 {"Mic Bias", NULL, "Headset Mic"},
296
297 /* Call Mic */
298 {"MIC2", NULL, "Mic Bias"},
299 {"MIC2N", NULL, "Mic Bias"},
300 {"Mic Bias", NULL, "Handset Mic"},
301
302 /* Call Speaker */
303 {"Handset Spk", NULL, "LOUT2"},
304 {"Handset Spk", NULL, "ROUT2"},
305
306 /* Connect the ALC pins */
307 {"ACIN", NULL, "ACOP"},
308};
309
310static const struct snd_kcontrol_new wm8753_neo1973_gta02_controls[] = {
311 SOC_DAPM_PIN_SWITCH("Stereo Out"),
312 SOC_DAPM_PIN_SWITCH("GSM Line Out"),
313 SOC_DAPM_PIN_SWITCH("GSM Line In"),
314 SOC_DAPM_PIN_SWITCH("Headset Mic"),
315 SOC_DAPM_PIN_SWITCH("Handset Mic"),
316 SOC_DAPM_PIN_SWITCH("Handset Spk"),
317
318 /* This has no effect, it exists only to maintain compatibility with
319 * existing ALSA state files.
320 */
321 SOC_SINGLE_EXT("Amp State Switch", 6, 0, 1, 0,
322 lm4853_get_state,
323 lm4853_set_state),
324 SOC_SINGLE_EXT("Amp Spk Switch", 7, 0, 1, 0,
325 lm4853_get_spk,
326 lm4853_set_spk),
327};
328
329/*
330 * This is an example machine initialisation for a wm8753 connected to a
331 * neo1973 GTA02.
332 */
333static int neo1973_gta02_wm8753_init(struct snd_soc_codec *codec)
334{
335 int err;
336
337 /* set up NC codec pins */
338 snd_soc_dapm_nc_pin(codec, "OUT3");
339 snd_soc_dapm_nc_pin(codec, "OUT4");
340 snd_soc_dapm_nc_pin(codec, "LINE1");
341 snd_soc_dapm_nc_pin(codec, "LINE2");
342
343 /* Add neo1973 gta02 specific widgets */
344 snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
345 ARRAY_SIZE(wm8753_dapm_widgets));
346
347 /* add neo1973 gta02 specific controls */
348 err = snd_soc_add_controls(codec, wm8753_neo1973_gta02_controls,
349 ARRAY_SIZE(wm8753_neo1973_gta02_controls));
350
351 if (err < 0)
352 return err;
353
354 /* set up neo1973 gta02 specific audio path audio_map */
355 snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
356
357 /* set endpoints to default off mode */
358 snd_soc_dapm_disable_pin(codec, "Stereo Out");
359 snd_soc_dapm_disable_pin(codec, "GSM Line Out");
360 snd_soc_dapm_disable_pin(codec, "GSM Line In");
361 snd_soc_dapm_disable_pin(codec, "Headset Mic");
362 snd_soc_dapm_disable_pin(codec, "Handset Mic");
363 snd_soc_dapm_disable_pin(codec, "Handset Spk");
364
365 snd_soc_dapm_sync(codec);
366
367 return 0;
368}
369
370/*
371 * BT Codec DAI
372 */
373static struct snd_soc_dai bt_dai = {
374 .name = "Bluetooth",
375 .id = 0,
376 .playback = {
377 .channels_min = 1,
378 .channels_max = 1,
379 .rates = SNDRV_PCM_RATE_8000,
380 .formats = SNDRV_PCM_FMTBIT_S16_LE,},
381 .capture = {
382 .channels_min = 1,
383 .channels_max = 1,
384 .rates = SNDRV_PCM_RATE_8000,
385 .formats = SNDRV_PCM_FMTBIT_S16_LE,},
386};
387
388static struct snd_soc_dai_link neo1973_gta02_dai[] = {
389{ /* Hifi Playback - for similatious use with voice below */
390 .name = "WM8753",
391 .stream_name = "WM8753 HiFi",
392 .cpu_dai = &s3c24xx_i2s_dai,
393 .codec_dai = &wm8753_dai[WM8753_DAI_HIFI],
394 .init = neo1973_gta02_wm8753_init,
395 .ops = &neo1973_gta02_hifi_ops,
396},
397{ /* Voice via BT */
398 .name = "Bluetooth",
399 .stream_name = "Voice",
400 .cpu_dai = &bt_dai,
401 .codec_dai = &wm8753_dai[WM8753_DAI_VOICE],
402 .ops = &neo1973_gta02_voice_ops,
403},
404};
405
406static struct snd_soc_card neo1973_gta02 = {
407 .name = "neo1973-gta02",
408 .platform = &s3c24xx_soc_platform,
409 .dai_link = neo1973_gta02_dai,
410 .num_links = ARRAY_SIZE(neo1973_gta02_dai),
411};
412
413static struct snd_soc_device neo1973_gta02_snd_devdata = {
414 .card = &neo1973_gta02,
415 .codec_dev = &soc_codec_dev_wm8753,
416};
417
418static struct platform_device *neo1973_gta02_snd_device;
419
420static int __init neo1973_gta02_init(void)
421{
422 int ret;
423
424 if (!machine_is_neo1973_gta02()) {
425 printk(KERN_INFO
426 "Only GTA02 is supported by this ASoC driver\n");
427 return -ENODEV;
428 }
429
430 /* register bluetooth DAI here */
431 ret = snd_soc_register_dai(&bt_dai);
432 if (ret)
433 return ret;
434
435 neo1973_gta02_snd_device = platform_device_alloc("soc-audio", -1);
436 if (!neo1973_gta02_snd_device)
437 return -ENOMEM;
438
439 platform_set_drvdata(neo1973_gta02_snd_device,
440 &neo1973_gta02_snd_devdata);
441 neo1973_gta02_snd_devdata.dev = &neo1973_gta02_snd_device->dev;
442 ret = platform_device_add(neo1973_gta02_snd_device);
443
444 if (ret) {
445 platform_device_put(neo1973_gta02_snd_device);
446 return ret;
447 }
448
449 /* Initialise GPIOs used by amp */
450 ret = gpio_request(GTA02_GPIO_HP_IN, "GTA02_HP_IN");
451 if (ret) {
452 pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_HP_IN);
453 goto err_unregister_device;
454 }
455
456 ret = gpio_direction_output(GTA02_GPIO_AMP_HP_IN, 1);
457 if (ret) {
458 pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_HP_IN);
459 goto err_free_gpio_hp_in;
460 }
461
462 ret = gpio_request(GTA02_GPIO_AMP_SHUT, "GTA02_AMP_SHUT");
463 if (ret) {
464 pr_err("gta02_wm8753: Failed to register GPIO %d\n", GTA02_GPIO_AMP_SHUT);
465 goto err_free_gpio_hp_in;
466 }
467
468 ret = gpio_direction_output(GTA02_GPIO_AMP_SHUT, 1);
469 if (ret) {
470 pr_err("gta02_wm8753: Failed to configure GPIO %d\n", GTA02_GPIO_AMP_SHUT);
471 goto err_free_gpio_amp_shut;
472 }
473
474 return 0;
475
476err_free_gpio_amp_shut:
477 gpio_free(GTA02_GPIO_AMP_SHUT);
478err_free_gpio_hp_in:
479 gpio_free(GTA02_GPIO_HP_IN);
480err_unregister_device:
481 platform_device_unregister(neo1973_gta02_snd_device);
482 return ret;
483}
484module_init(neo1973_gta02_init);
485
486static void __exit neo1973_gta02_exit(void)
487{
488 snd_soc_unregister_dai(&bt_dai);
489 platform_device_unregister(neo1973_gta02_snd_device);
490 gpio_free(GTA02_GPIO_HP_IN);
491 gpio_free(GTA02_GPIO_AMP_SHUT);
492}
493module_exit(neo1973_gta02_exit);
494
495/* Module information */
496MODULE_AUTHOR("Graeme Gregory, graeme@openmoko.org");
497MODULE_DESCRIPTION("ALSA SoC WM8753 Neo1973 GTA02");
498MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 1a283170ca92..aa7af0b8d421 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -36,6 +36,7 @@
36#include <mach/dma.h> 36#include <mach/dma.h>
37 37
38#include "s3c-i2s-v2.h" 38#include "s3c-i2s-v2.h"
39#include "s3c24xx-pcm.h"
39 40
40#undef S3C_IIS_V2_SUPPORTED 41#undef S3C_IIS_V2_SUPPORTED
41 42
@@ -357,19 +358,19 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream,
357#endif 358#endif
358 359
359#ifdef CONFIG_PLAT_S3C64XX 360#ifdef CONFIG_PLAT_S3C64XX
360 iismod &= ~0x606; 361 iismod &= ~(S3C64XX_IISMOD_BLC_MASK | S3C2412_IISMOD_BCLK_MASK);
361 /* Sample size */ 362 /* Sample size */
362 switch (params_format(params)) { 363 switch (params_format(params)) {
363 case SNDRV_PCM_FORMAT_S8: 364 case SNDRV_PCM_FORMAT_S8:
364 /* 8 bit sample, 16fs BCLK */ 365 /* 8 bit sample, 16fs BCLK */
365 iismod |= 0x2004; 366 iismod |= (S3C64XX_IISMOD_BLC_8BIT | S3C2412_IISMOD_BCLK_16FS);
366 break; 367 break;
367 case SNDRV_PCM_FORMAT_S16_LE: 368 case SNDRV_PCM_FORMAT_S16_LE:
368 /* 16 bit sample, 32fs BCLK */ 369 /* 16 bit sample, 32fs BCLK */
369 break; 370 break;
370 case SNDRV_PCM_FORMAT_S24_LE: 371 case SNDRV_PCM_FORMAT_S24_LE:
371 /* 24 bit sample, 48fs BCLK */ 372 /* 24 bit sample, 48fs BCLK */
372 iismod |= 0x4002; 373 iismod |= (S3C64XX_IISMOD_BLC_24BIT | S3C2412_IISMOD_BCLK_48FS);
373 break; 374 break;
374 } 375 }
375#endif 376#endif
@@ -387,6 +388,8 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
387 int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); 388 int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE);
388 unsigned long irqs; 389 unsigned long irqs;
389 int ret = 0; 390 int ret = 0;
391 int channel = ((struct s3c24xx_pcm_dma_params *)
392 rtd->dai->cpu_dai->dma_data)->channel;
390 393
391 pr_debug("Entered %s\n", __func__); 394 pr_debug("Entered %s\n", __func__);
392 395
@@ -416,6 +419,14 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
416 s3c2412_snd_txctrl(i2s, 1); 419 s3c2412_snd_txctrl(i2s, 1);
417 420
418 local_irq_restore(irqs); 421 local_irq_restore(irqs);
422
423 /*
424 * Load the next buffer to DMA to meet the reqirement
425 * of the auto reload mechanism of S3C24XX.
426 * This call won't bother S3C64XX.
427 */
428 s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
429
419 break; 430 break;
420 431
421 case SNDRV_PCM_TRIGGER_STOP: 432 case SNDRV_PCM_TRIGGER_STOP:
diff --git a/sound/soc/s3c24xx/s3c2443-ac97.c b/sound/soc/s3c24xx/s3c2443-ac97.c
index 3f03d5ddfacd..fc1beb0930b9 100644
--- a/sound/soc/s3c24xx/s3c2443-ac97.c
+++ b/sound/soc/s3c24xx/s3c2443-ac97.c
@@ -47,7 +47,7 @@ static struct s3c24xx_ac97_info s3c24xx_ac97;
47 47
48static DECLARE_COMPLETION(ac97_completion); 48static DECLARE_COMPLETION(ac97_completion);
49static u32 codec_ready; 49static u32 codec_ready;
50static DECLARE_MUTEX(ac97_mutex); 50static DEFINE_MUTEX(ac97_mutex);
51 51
52static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97, 52static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
53 unsigned short reg) 53 unsigned short reg)
@@ -56,7 +56,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
56 u32 ac_codec_cmd; 56 u32 ac_codec_cmd;
57 u32 stat, addr, data; 57 u32 stat, addr, data;
58 58
59 down(&ac97_mutex); 59 mutex_lock(&ac97_mutex);
60 60
61 codec_ready = S3C_AC97_GLBSTAT_CODECREADY; 61 codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
62 ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); 62 ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
@@ -79,7 +79,7 @@ static unsigned short s3c2443_ac97_read(struct snd_ac97 *ac97,
79 printk(KERN_ERR "s3c24xx-ac97: req addr = %02x," 79 printk(KERN_ERR "s3c24xx-ac97: req addr = %02x,"
80 " rep addr = %02x\n", reg, addr); 80 " rep addr = %02x\n", reg, addr);
81 81
82 up(&ac97_mutex); 82 mutex_unlock(&ac97_mutex);
83 83
84 return (unsigned short)data; 84 return (unsigned short)data;
85} 85}
@@ -90,7 +90,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
90 u32 ac_glbctrl; 90 u32 ac_glbctrl;
91 u32 ac_codec_cmd; 91 u32 ac_codec_cmd;
92 92
93 down(&ac97_mutex); 93 mutex_lock(&ac97_mutex);
94 94
95 codec_ready = S3C_AC97_GLBSTAT_CODECREADY; 95 codec_ready = S3C_AC97_GLBSTAT_CODECREADY;
96 ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); 96 ac_codec_cmd = readl(s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
@@ -109,7 +109,7 @@ static void s3c2443_ac97_write(struct snd_ac97 *ac97, unsigned short reg,
109 ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ; 109 ac_codec_cmd |= S3C_AC97_CODEC_CMD_READ;
110 writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD); 110 writel(ac_codec_cmd, s3c24xx_ac97.regs + S3C_AC97_CODEC_CMD);
111 111
112 up(&ac97_mutex); 112 mutex_unlock(&ac97_mutex);
113 113
114} 114}
115 115
@@ -290,6 +290,9 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
290 struct snd_soc_dai *dai) 290 struct snd_soc_dai *dai)
291{ 291{
292 u32 ac_glbctrl; 292 u32 ac_glbctrl;
293 struct snd_soc_pcm_runtime *rtd = substream->private_data;
294 int channel = ((struct s3c24xx_pcm_dma_params *)
295 rtd->dai->cpu_dai->dma_data)->channel;
293 296
294 ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); 297 ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
295 switch (cmd) { 298 switch (cmd) {
@@ -312,6 +315,8 @@ static int s3c2443_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
312 } 315 }
313 writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); 316 writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
314 317
318 s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
319
315 return 0; 320 return 0;
316} 321}
317 322
@@ -334,6 +339,9 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
334 int cmd, struct snd_soc_dai *dai) 339 int cmd, struct snd_soc_dai *dai)
335{ 340{
336 u32 ac_glbctrl; 341 u32 ac_glbctrl;
342 struct snd_soc_pcm_runtime *rtd = substream->private_data;
343 int channel = ((struct s3c24xx_pcm_dma_params *)
344 rtd->dai->cpu_dai->dma_data)->channel;
337 345
338 ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); 346 ac_glbctrl = readl(s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
339 switch (cmd) { 347 switch (cmd) {
@@ -349,6 +357,8 @@ static int s3c2443_ac97_mic_trigger(struct snd_pcm_substream *substream,
349 } 357 }
350 writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); 358 writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL);
351 359
360 s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
361
352 return 0; 362 return 0;
353} 363}
354 364
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c
index 556e35f0ab73..40e2c4790f0d 100644
--- a/sound/soc/s3c24xx/s3c24xx-i2s.c
+++ b/sound/soc/s3c24xx/s3c24xx-i2s.c
@@ -279,6 +279,9 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
279 struct snd_soc_dai *dai) 279 struct snd_soc_dai *dai)
280{ 280{
281 int ret = 0; 281 int ret = 0;
282 struct snd_soc_pcm_runtime *rtd = substream->private_data;
283 int channel = ((struct s3c24xx_pcm_dma_params *)
284 rtd->dai->cpu_dai->dma_data)->channel;
282 285
283 pr_debug("Entered %s\n", __func__); 286 pr_debug("Entered %s\n", __func__);
284 287
@@ -296,6 +299,8 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
296 s3c24xx_snd_rxctrl(1); 299 s3c24xx_snd_rxctrl(1);
297 else 300 else
298 s3c24xx_snd_txctrl(1); 301 s3c24xx_snd_txctrl(1);
302
303 s3c2410_dma_ctrl(channel, S3C2410_DMAOP_STARTED);
299 break; 304 break;
300 case SNDRV_PCM_TRIGGER_STOP: 305 case SNDRV_PCM_TRIGGER_STOP:
301 case SNDRV_PCM_TRIGGER_SUSPEND: 306 case SNDRV_PCM_TRIGGER_SUSPEND:
diff --git a/sound/soc/s3c24xx/s3c24xx-pcm.c b/sound/soc/s3c24xx/s3c24xx-pcm.c
index eecfa5eba06b..5cbbdc80fde3 100644
--- a/sound/soc/s3c24xx/s3c24xx-pcm.c
+++ b/sound/soc/s3c24xx/s3c24xx-pcm.c
@@ -255,7 +255,6 @@ static int s3c24xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
255 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: 255 case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
256 prtd->state |= ST_RUNNING; 256 prtd->state |= ST_RUNNING;
257 s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START); 257 s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_START);
258 s3c2410_dma_ctrl(prtd->params->channel, S3C2410_DMAOP_STARTED);
259 break; 258 break;
260 259
261 case SNDRV_PCM_TRIGGER_STOP: 260 case SNDRV_PCM_TRIGGER_STOP:
@@ -318,6 +317,7 @@ static int s3c24xx_pcm_open(struct snd_pcm_substream *substream)
318 317
319 pr_debug("Entered %s\n", __func__); 318 pr_debug("Entered %s\n", __func__);
320 319
320 snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS);
321 snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware); 321 snd_soc_set_runtime_hwparams(substream, &s3c24xx_pcm_hardware);
322 322
323 prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL); 323 prtd = kzalloc(sizeof(struct s3c24xx_runtime_data), GFP_KERNEL);
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.c b/sound/soc/s3c24xx/s3c24xx_simtec.c
new file mode 100644
index 000000000000..1966e0d5652d
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.c
@@ -0,0 +1,394 @@
1/* sound/soc/s3c24xx/s3c24xx_simtec.c
2 *
3 * Copyright 2009 Simtec Electronics
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License version 2 as
7 * published by the Free Software Foundation.
8*/
9
10#include <linux/module.h>
11#include <linux/moduleparam.h>
12#include <linux/platform_device.h>
13#include <linux/gpio.h>
14#include <linux/clk.h>
15#include <linux/i2c.h>
16
17#include <sound/core.h>
18#include <sound/pcm.h>
19#include <sound/soc.h>
20#include <sound/soc-dapm.h>
21
22#include <plat/audio-simtec.h>
23
24#include "s3c24xx-pcm.h"
25#include "s3c24xx-i2s.h"
26#include "s3c24xx_simtec.h"
27
28static struct s3c24xx_audio_simtec_pdata *pdata;
29static struct clk *xtal_clk;
30
31static int spk_gain;
32static int spk_unmute;
33
34/**
35 * speaker_gain_get - read the speaker gain setting.
36 * @kcontrol: The control for the speaker gain.
37 * @ucontrol: The value that needs to be updated.
38 *
39 * Read the value for the AMP gain control.
40 */
41static int speaker_gain_get(struct snd_kcontrol *kcontrol,
42 struct snd_ctl_elem_value *ucontrol)
43{
44 ucontrol->value.integer.value[0] = spk_gain;
45 return 0;
46}
47
48/**
49 * speaker_gain_set - set the value of the speaker amp gain
50 * @value: The value to write.
51 */
52static void speaker_gain_set(int value)
53{
54 gpio_set_value_cansleep(pdata->amp_gain[0], value & 1);
55 gpio_set_value_cansleep(pdata->amp_gain[1], value >> 1);
56}
57
58/**
59 * speaker_gain_put - set the speaker gain setting.
60 * @kcontrol: The control for the speaker gain.
61 * @ucontrol: The value that needs to be set.
62 *
63 * Set the value of the speaker gain from the specified
64 * @ucontrol setting.
65 *
66 * Note, if the speaker amp is muted, then we do not set a gain value
67 * as at-least one of the ICs that is fitted will try and power up even
68 * if the main control is set to off.
69 */
70static int speaker_gain_put(struct snd_kcontrol *kcontrol,
71 struct snd_ctl_elem_value *ucontrol)
72{
73 int value = ucontrol->value.integer.value[0];
74
75 spk_gain = value;
76
77 if (!spk_unmute)
78 speaker_gain_set(value);
79
80 return 0;
81}
82
83static const struct snd_kcontrol_new amp_gain_controls[] = {
84 SOC_SINGLE_EXT("Speaker Gain", 0, 0, 3, 0,
85 speaker_gain_get, speaker_gain_put),
86};
87
88/**
89 * spk_unmute_state - set the unmute state of the speaker
90 * @to: zero to unmute, non-zero to ununmute.
91 */
92static void spk_unmute_state(int to)
93{
94 pr_debug("%s: to=%d\n", __func__, to);
95
96 spk_unmute = to;
97 gpio_set_value(pdata->amp_gpio, to);
98
99 /* if we're umuting, also re-set the gain */
100 if (to && pdata->amp_gain[0] > 0)
101 speaker_gain_set(spk_gain);
102}
103
104/**
105 * speaker_unmute_get - read the speaker unmute setting.
106 * @kcontrol: The control for the speaker gain.
107 * @ucontrol: The value that needs to be updated.
108 *
109 * Read the value for the AMP gain control.
110 */
111static int speaker_unmute_get(struct snd_kcontrol *kcontrol,
112 struct snd_ctl_elem_value *ucontrol)
113{
114 ucontrol->value.integer.value[0] = spk_unmute;
115 return 0;
116}
117
118/**
119 * speaker_unmute_put - set the speaker unmute setting.
120 * @kcontrol: The control for the speaker gain.
121 * @ucontrol: The value that needs to be set.
122 *
123 * Set the value of the speaker gain from the specified
124 * @ucontrol setting.
125 */
126static int speaker_unmute_put(struct snd_kcontrol *kcontrol,
127 struct snd_ctl_elem_value *ucontrol)
128{
129 spk_unmute_state(ucontrol->value.integer.value[0]);
130 return 0;
131}
132
133/* This is added as a manual control as the speaker amps create clicks
134 * when their power state is changed, which are far more noticeable than
135 * anything produced by the CODEC itself.
136 */
137static const struct snd_kcontrol_new amp_unmute_controls[] = {
138 SOC_SINGLE_EXT("Speaker Switch", 0, 0, 1, 0,
139 speaker_unmute_get, speaker_unmute_put),
140};
141
142void simtec_audio_init(struct snd_soc_codec *codec)
143{
144 if (pdata->amp_gpio > 0) {
145 pr_debug("%s: adding amp routes\n", __func__);
146
147 snd_soc_add_controls(codec, amp_unmute_controls,
148 ARRAY_SIZE(amp_unmute_controls));
149 }
150
151 if (pdata->amp_gain[0] > 0) {
152 pr_debug("%s: adding amp controls\n", __func__);
153 snd_soc_add_controls(codec, amp_gain_controls,
154 ARRAY_SIZE(amp_gain_controls));
155 }
156}
157EXPORT_SYMBOL_GPL(simtec_audio_init);
158
159#define CODEC_CLOCK 12000000
160
161/**
162 * simtec_hw_params - update hardware parameters
163 * @substream: The audio substream instance.
164 * @params: The parameters requested.
165 *
166 * Update the codec data routing and configuration settings
167 * from the supplied data.
168 */
169static int simtec_hw_params(struct snd_pcm_substream *substream,
170 struct snd_pcm_hw_params *params)
171{
172 struct snd_soc_pcm_runtime *rtd = substream->private_data;
173 struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
174 struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
175 int ret;
176
177 /* Set the CODEC as the bus clock master, I2S */
178 ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
179 SND_SOC_DAIFMT_NB_NF |
180 SND_SOC_DAIFMT_CBM_CFM);
181 if (ret) {
182 pr_err("%s: failed set cpu dai format\n", __func__);
183 return ret;
184 }
185
186 /* Set the CODEC as the bus clock master */
187 ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
188 SND_SOC_DAIFMT_NB_NF |
189 SND_SOC_DAIFMT_CBM_CFM);
190 if (ret) {
191 pr_err("%s: failed set codec dai format\n", __func__);
192 return ret;
193 }
194
195 ret = snd_soc_dai_set_sysclk(codec_dai, 0,
196 CODEC_CLOCK, SND_SOC_CLOCK_IN);
197 if (ret) {
198 pr_err( "%s: failed setting codec sysclk\n", __func__);
199 return ret;
200 }
201
202 if (pdata->use_mpllin) {
203 ret = snd_soc_dai_set_sysclk(cpu_dai, S3C24XX_CLKSRC_MPLL,
204 0, SND_SOC_CLOCK_OUT);
205
206 if (ret) {
207 pr_err("%s: failed to set MPLLin as clksrc\n",
208 __func__);
209 return ret;
210 }
211 }
212
213 if (pdata->output_cdclk) {
214 int cdclk_scale;
215
216 cdclk_scale = clk_get_rate(xtal_clk) / CODEC_CLOCK;
217 cdclk_scale--;
218
219 ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C24XX_DIV_PRESCALER,
220 cdclk_scale);
221 }
222
223 return 0;
224}
225
226static int simtec_call_startup(struct s3c24xx_audio_simtec_pdata *pd)
227{
228 /* call any board supplied startup code, this currently only
229 * covers the bast/vr1000 which have a CPLD in the way of the
230 * LRCLK */
231 if (pd->startup)
232 pd->startup();
233
234 return 0;
235}
236
237static struct snd_soc_ops simtec_snd_ops = {
238 .hw_params = simtec_hw_params,
239};
240
241/**
242 * attach_gpio_amp - get and configure the necessary gpios
243 * @dev: The device we're probing.
244 * @pd: The platform data supplied by the board.
245 *
246 * If there is a GPIO based amplifier attached to the board, claim
247 * the necessary GPIO lines for it, and set default values.
248 */
249static int attach_gpio_amp(struct device *dev,
250 struct s3c24xx_audio_simtec_pdata *pd)
251{
252 int ret;
253
254 /* attach gpio amp gain (if any) */
255 if (pdata->amp_gain[0] > 0) {
256 ret = gpio_request(pd->amp_gain[0], "gpio-amp-gain0");
257 if (ret) {
258 dev_err(dev, "cannot get amp gpio gain0\n");
259 return ret;
260 }
261
262 ret = gpio_request(pd->amp_gain[1], "gpio-amp-gain1");
263 if (ret) {
264 dev_err(dev, "cannot get amp gpio gain1\n");
265 gpio_free(pdata->amp_gain[0]);
266 return ret;
267 }
268
269 gpio_direction_output(pd->amp_gain[0], 0);
270 gpio_direction_output(pd->amp_gain[1], 0);
271 }
272
273 /* note, curently we assume GPA0 isn't valid amp */
274 if (pdata->amp_gpio > 0) {
275 ret = gpio_request(pd->amp_gpio, "gpio-amp");
276 if (ret) {
277 dev_err(dev, "cannot get amp gpio %d (%d)\n",
278 pd->amp_gpio, ret);
279 goto err_amp;
280 }
281
282 /* set the amp off at startup */
283 spk_unmute_state(0);
284 }
285
286 return 0;
287
288err_amp:
289 if (pd->amp_gain[0] > 0) {
290 gpio_free(pd->amp_gain[0]);
291 gpio_free(pd->amp_gain[1]);
292 }
293
294 return ret;
295}
296
297static void detach_gpio_amp(struct s3c24xx_audio_simtec_pdata *pd)
298{
299 if (pd->amp_gain[0] > 0) {
300 gpio_free(pd->amp_gain[0]);
301 gpio_free(pd->amp_gain[1]);
302 }
303
304 if (pd->amp_gpio > 0)
305 gpio_free(pd->amp_gpio);
306}
307
308#ifdef CONFIG_PM
309int simtec_audio_resume(struct device *dev)
310{
311 simtec_call_startup(pdata);
312 return 0;
313}
314
315struct dev_pm_ops simtec_audio_pmops = {
316 .resume = simtec_audio_resume,
317};
318EXPORT_SYMBOL_GPL(simtec_audio_pmops);
319#endif
320
321int __devinit simtec_audio_core_probe(struct platform_device *pdev,
322 struct snd_soc_device *socdev)
323{
324 struct platform_device *snd_dev;
325 int ret;
326
327 socdev->card->dai_link->ops = &simtec_snd_ops;
328
329 pdata = pdev->dev.platform_data;
330 if (!pdata) {
331 dev_err(&pdev->dev, "no platform data supplied\n");
332 return -EINVAL;
333 }
334
335 simtec_call_startup(pdata);
336
337 xtal_clk = clk_get(&pdev->dev, "xtal");
338 if (IS_ERR(xtal_clk)) {
339 dev_err(&pdev->dev, "could not get clkout0\n");
340 return -EINVAL;
341 }
342
343 dev_info(&pdev->dev, "xtal rate is %ld\n", clk_get_rate(xtal_clk));
344
345 ret = attach_gpio_amp(&pdev->dev, pdata);
346 if (ret)
347 goto err_clk;
348
349 snd_dev = platform_device_alloc("soc-audio", -1);
350 if (!snd_dev) {
351 dev_err(&pdev->dev, "failed to alloc soc-audio devicec\n");
352 ret = -ENOMEM;
353 goto err_gpio;
354 }
355
356 platform_set_drvdata(snd_dev, socdev);
357 socdev->dev = &snd_dev->dev;
358
359 ret = platform_device_add(snd_dev);
360 if (ret) {
361 dev_err(&pdev->dev, "failed to add soc-audio dev\n");
362 goto err_pdev;
363 }
364
365 platform_set_drvdata(pdev, snd_dev);
366 return 0;
367
368err_pdev:
369 platform_device_put(snd_dev);
370
371err_gpio:
372 detach_gpio_amp(pdata);
373
374err_clk:
375 clk_put(xtal_clk);
376 return ret;
377}
378EXPORT_SYMBOL_GPL(simtec_audio_core_probe);
379
380int __devexit simtec_audio_remove(struct platform_device *pdev)
381{
382 struct platform_device *snd_dev = platform_get_drvdata(pdev);
383
384 platform_device_unregister(snd_dev);
385
386 detach_gpio_amp(pdata);
387 clk_put(xtal_clk);
388 return 0;
389}
390EXPORT_SYMBOL_GPL(simtec_audio_remove);
391
392MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
393MODULE_DESCRIPTION("ALSA SoC Simtec Audio common support");
394MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec.h b/sound/soc/s3c24xx/s3c24xx_simtec.h
new file mode 100644
index 000000000000..2714203af161
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec.h
@@ -0,0 +1,22 @@
1/* sound/soc/s3c24xx/s3c24xx_simtec.h
2 *
3 * Copyright 2009 Simtec Electronics
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License version 2 as
7 * published by the Free Software Foundation.
8*/
9
10extern void simtec_audio_init(struct snd_soc_codec *codec);
11
12extern int simtec_audio_core_probe(struct platform_device *pdev,
13 struct snd_soc_device *socdev);
14
15extern int simtec_audio_remove(struct platform_device *pdev);
16
17#ifdef CONFIG_PM
18extern struct dev_pm_ops simtec_audio_pmops;
19#define simtec_audio_pm &simtec_audio_pmops
20#else
21#define simtec_audio_pm NULL
22#endif
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
new file mode 100644
index 000000000000..8346bd96eaf5
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
@@ -0,0 +1,153 @@
1/* sound/soc/s3c24xx/s3c24xx_simtec_hermes.c
2 *
3 * Copyright 2009 Simtec Electronics
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License version 2 as
7 * published by the Free Software Foundation.
8*/
9
10#include <linux/module.h>
11#include <linux/clk.h>
12#include <linux/platform_device.h>
13
14#include <sound/core.h>
15#include <sound/pcm.h>
16#include <sound/soc.h>
17#include <sound/soc-dapm.h>
18
19#include <plat/audio-simtec.h>
20
21#include "s3c24xx-pcm.h"
22#include "s3c24xx-i2s.h"
23#include "s3c24xx_simtec.h"
24
25#include "../codecs/tlv320aic3x.h"
26
27static const struct snd_soc_dapm_widget dapm_widgets[] = {
28 SND_SOC_DAPM_LINE("GSM Out", NULL),
29 SND_SOC_DAPM_LINE("GSM In", NULL),
30 SND_SOC_DAPM_LINE("Line In", NULL),
31 SND_SOC_DAPM_LINE("Line Out", NULL),
32 SND_SOC_DAPM_LINE("ZV", NULL),
33 SND_SOC_DAPM_MIC("Mic Jack", NULL),
34 SND_SOC_DAPM_HP("Headphone Jack", NULL),
35};
36
37static const struct snd_soc_dapm_route base_map[] = {
38 /* Headphone connected to HP{L,R}OUT and HP{L,R}COM */
39
40 { "Headphone Jack", NULL, "HPLOUT" },
41 { "Headphone Jack", NULL, "HPLCOM" },
42 { "Headphone Jack", NULL, "HPROUT" },
43 { "Headphone Jack", NULL, "HPRCOM" },
44
45 /* ZV connected to Line1 */
46
47 { "LINE1L", NULL, "ZV" },
48 { "LINE1R", NULL, "ZV" },
49
50 /* Line In connected to Line2 */
51
52 { "LINE2L", NULL, "Line In" },
53 { "LINE2R", NULL, "Line In" },
54
55 /* Microphone connected to MIC3R and MIC_BIAS */
56
57 { "MIC3L", NULL, "Mic Jack" },
58
59 /* GSM connected to MONO_LOUT and MIC3L (in) */
60
61 { "GSM Out", NULL, "MONO_LOUT" },
62 { "MIC3L", NULL, "GSM In" },
63
64 /* Speaker is connected to LINEOUT{LN,LP,RN,RP}, however we are
65 * not using the DAPM to power it up and down as there it makes
66 * a click when powering up. */
67};
68
69/**
70 * simtec_hermes_init - initialise and add controls
71 * @codec; The codec instance to attach to.
72 *
73 * Attach our controls and configure the necessary codec
74 * mappings for our sound card instance.
75*/
76static int simtec_hermes_init(struct snd_soc_codec *codec)
77{
78 snd_soc_dapm_new_controls(codec, dapm_widgets,
79 ARRAY_SIZE(dapm_widgets));
80
81 snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
82
83 snd_soc_dapm_enable_pin(codec, "Headphone Jack");
84 snd_soc_dapm_enable_pin(codec, "Line In");
85 snd_soc_dapm_enable_pin(codec, "Line Out");
86 snd_soc_dapm_enable_pin(codec, "Mic Jack");
87
88 simtec_audio_init(codec);
89 snd_soc_dapm_sync(codec);
90
91 return 0;
92}
93
94static struct aic3x_setup_data codec_setup = {
95};
96
97static struct snd_soc_dai_link simtec_dai_aic33 = {
98 .name = "tlv320aic33",
99 .stream_name = "TLV320AIC33",
100 .cpu_dai = &s3c24xx_i2s_dai,
101 .codec_dai = &aic3x_dai,
102 .init = simtec_hermes_init,
103};
104
105/* simtec audio machine driver */
106static struct snd_soc_card snd_soc_machine_simtec_aic33 = {
107 .name = "Simtec-Hermes",
108 .platform = &s3c24xx_soc_platform,
109 .dai_link = &simtec_dai_aic33,
110 .num_links = 1,
111};
112
113/* simtec audio subsystem */
114static struct snd_soc_device simtec_snd_devdata_aic33 = {
115 .card = &snd_soc_machine_simtec_aic33,
116 .codec_dev = &soc_codec_dev_aic3x,
117 .codec_data = &codec_setup,
118};
119
120static int __devinit simtec_audio_hermes_probe(struct platform_device *pd)
121{
122 dev_info(&pd->dev, "probing....\n");
123 return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic33);
124}
125
126static struct platform_driver simtec_audio_hermes_platdrv = {
127 .driver = {
128 .owner = THIS_MODULE,
129 .name = "s3c24xx-simtec-hermes-snd",
130 .pm = simtec_audio_pm,
131 },
132 .probe = simtec_audio_hermes_probe,
133 .remove = __devexit_p(simtec_audio_remove),
134};
135
136MODULE_ALIAS("platform:s3c24xx-simtec-hermes-snd");
137
138static int __init simtec_hermes_modinit(void)
139{
140 return platform_driver_register(&simtec_audio_hermes_platdrv);
141}
142
143static void __exit simtec_hermes_modexit(void)
144{
145 platform_driver_unregister(&simtec_audio_hermes_platdrv);
146}
147
148module_init(simtec_hermes_modinit);
149module_exit(simtec_hermes_modexit);
150
151MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
152MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
153MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
new file mode 100644
index 000000000000..25797e096175
--- /dev/null
+++ b/sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
@@ -0,0 +1,137 @@
1/* sound/soc/s3c24xx/s3c24xx_simtec_tlv320aic23.c
2 *
3 * Copyright 2009 Simtec Electronics
4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License version 2 as
7 * published by the Free Software Foundation.
8*/
9
10#include <linux/module.h>
11#include <linux/clk.h>
12#include <linux/platform_device.h>
13
14#include <sound/core.h>
15#include <sound/pcm.h>
16#include <sound/soc.h>
17#include <sound/soc-dapm.h>
18
19#include <plat/audio-simtec.h>
20
21#include "s3c24xx-pcm.h"
22#include "s3c24xx-i2s.h"
23#include "s3c24xx_simtec.h"
24
25#include "../codecs/tlv320aic23.h"
26
27/* supported machines:
28 *
29 * Machine Connections AMP
30 * ------- ----------- ---
31 * BAST MIC, HPOUT, LOUT, LIN TPA2001D1 (HPOUTL,R) (gain hardwired)
32 * VR1000 HPOUT, LIN None
33 * VR2000 LIN, LOUT, MIC, HP LM4871 (HPOUTL,R)
34 * DePicture LIN, LOUT, MIC, HP LM4871 (HPOUTL,R)
35 * Anubis LIN, LOUT, MIC, HP TPA2001D1 (HPOUTL,R)
36 */
37
38static const struct snd_soc_dapm_widget dapm_widgets[] = {
39 SND_SOC_DAPM_HP("Headphone Jack", NULL),
40 SND_SOC_DAPM_LINE("Line In", NULL),
41 SND_SOC_DAPM_LINE("Line Out", NULL),
42 SND_SOC_DAPM_MIC("Mic Jack", NULL),
43};
44
45static const struct snd_soc_dapm_route base_map[] = {
46 { "Headphone Jack", NULL, "LHPOUT"},
47 { "Headphone Jack", NULL, "RHPOUT"},
48
49 { "Line Out", NULL, "LOUT" },
50 { "Line Out", NULL, "ROUT" },
51
52 { "LLINEIN", NULL, "Line In"},
53 { "RLINEIN", NULL, "Line In"},
54
55 { "MICIN", NULL, "Mic Jack"},
56};
57
58/**
59 * simtec_tlv320aic23_init - initialise and add controls
60 * @codec; The codec instance to attach to.
61 *
62 * Attach our controls and configure the necessary codec
63 * mappings for our sound card instance.
64*/
65static int simtec_tlv320aic23_init(struct snd_soc_codec *codec)
66{
67 snd_soc_dapm_new_controls(codec, dapm_widgets,
68 ARRAY_SIZE(dapm_widgets));
69
70 snd_soc_dapm_add_routes(codec, base_map, ARRAY_SIZE(base_map));
71
72 snd_soc_dapm_enable_pin(codec, "Headphone Jack");
73 snd_soc_dapm_enable_pin(codec, "Line In");
74 snd_soc_dapm_enable_pin(codec, "Line Out");
75 snd_soc_dapm_enable_pin(codec, "Mic Jack");
76
77 simtec_audio_init(codec);
78 snd_soc_dapm_sync(codec);
79
80 return 0;
81}
82
83static struct snd_soc_dai_link simtec_dai_aic23 = {
84 .name = "tlv320aic23",
85 .stream_name = "TLV320AIC23",
86 .cpu_dai = &s3c24xx_i2s_dai,
87 .codec_dai = &tlv320aic23_dai,
88 .init = simtec_tlv320aic23_init,
89};
90
91/* simtec audio machine driver */
92static struct snd_soc_card snd_soc_machine_simtec_aic23 = {
93 .name = "Simtec",
94 .platform = &s3c24xx_soc_platform,
95 .dai_link = &simtec_dai_aic23,
96 .num_links = 1,
97};
98
99/* simtec audio subsystem */
100static struct snd_soc_device simtec_snd_devdata_aic23 = {
101 .card = &snd_soc_machine_simtec_aic23,
102 .codec_dev = &soc_codec_dev_tlv320aic23,
103};
104
105static int __devinit simtec_audio_tlv320aic23_probe(struct platform_device *pd)
106{
107 return simtec_audio_core_probe(pd, &simtec_snd_devdata_aic23);
108}
109
110static struct platform_driver simtec_audio_tlv320aic23_platdrv = {
111 .driver = {
112 .owner = THIS_MODULE,
113 .name = "s3c24xx-simtec-tlv320aic23",
114 .pm = simtec_audio_pm,
115 },
116 .probe = simtec_audio_tlv320aic23_probe,
117 .remove = __devexit_p(simtec_audio_remove),
118};
119
120MODULE_ALIAS("platform:s3c24xx-simtec-tlv320aic23");
121
122static int __init simtec_tlv320aic23_modinit(void)
123{
124 return platform_driver_register(&simtec_audio_tlv320aic23_platdrv);
125}
126
127static void __exit simtec_tlv320aic23_modexit(void)
128{
129 platform_driver_unregister(&simtec_audio_tlv320aic23_platdrv);
130}
131
132module_init(simtec_tlv320aic23_modinit);
133module_exit(simtec_tlv320aic23_modexit);
134
135MODULE_AUTHOR("Ben Dooks <ben@simtec.co.uk>");
136MODULE_DESCRIPTION("ALSA SoC Simtec Audio support");
137MODULE_LICENSE("GPL");
generated by cgit v1.2.2 (git 2.25.0) at 2025-01-06 12:07:00 -0500