From 36aeff6146925025033e2bcd45fa1e9725bc4599 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Wed, 12 May 2010 10:35:36 +0300 Subject: ASoC: TWL4030: Add control for digimic Left Right swap The codec has support for swapping the left and right channels in the digimic interface. New kcontrol to handle this bit. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/twl4030.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 0fe74d1e2c5f..6a34f562b563 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1102,6 +1102,16 @@ static const struct soc_enum twl4030_vibradir_enum = ARRAY_SIZE(twl4030_vibradir_texts), twl4030_vibradir_texts); +/* Digimic Left and right swapping */ +static const char *twl4030_digimicswap_texts[] = { + "Not swapped", "Swapped", +}; + +static const struct soc_enum twl4030_digimicswap_enum = + SOC_ENUM_SINGLE(TWL4030_REG_MISC_SET_1, 0, + ARRAY_SIZE(twl4030_digimicswap_texts), + twl4030_digimicswap_texts); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Codec operation mode control */ SOC_ENUM_EXT("Codec Operation Mode", twl4030_op_modes_enum, @@ -1178,6 +1188,8 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_ENUM("Vibra H-bridge mode", twl4030_vibradirmode_enum), SOC_ENUM("Vibra H-bridge direction", twl4030_vibradir_enum), + + SOC_ENUM("Digimic LR Swap", twl4030_digimicswap_enum), }; static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { -- cgit v1.2.2 From d98508a121e8f4b1ccf876fea463fa0afddc4e19 Mon Sep 17 00:00:00 2001 From: Sergey Lapin Date: Thu, 13 May 2010 19:48:16 +0400 Subject: OMAP: McBSP: Add 32-bit mode support This patchs should allow to use 32-bit samples on e.g. TLV320AIC3x codec, or others. Signed-off-by: Sergey Lapin Acked-by: Mark Brown Acked-by: Peter Ujfalusi Signed-off-by: Liam Girdwood --- sound/soc/omap/omap-mcbsp.c | 28 ++++++++++++++++++++++++---- 1 file changed, 24 insertions(+), 4 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 2d33a89f147a..6f44cb4d30b8 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -320,8 +320,18 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma; omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port; omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode; - omap_mcbsp_dai_dma_params[id][substream->stream].data_type = - OMAP_DMA_DATA_TYPE_S16; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S16; + break; + case SNDRV_PCM_FORMAT_S32_LE: + omap_mcbsp_dai_dma_params[id][substream->stream].data_type = + OMAP_DMA_DATA_TYPE_S32; + break; + default: + return -EINVAL; + } snd_soc_dai_set_dma_data(cpu_dai, substream, &omap_mcbsp_dai_dma_params[id][substream->stream]); @@ -356,6 +366,14 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); break; + case SNDRV_PCM_FORMAT_S32_LE: + /* Set word lengths */ + wlen = 32; + regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_32); + regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_32); + regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_32); + regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_32); + break; default: /* Unsupported PCM format */ return -EINVAL; @@ -659,13 +677,15 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = { .channels_min = 1, \ .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ }, \ .capture = { \ .channels_min = 1, \ .channels_max = 16, \ .rates = OMAP_MCBSP_RATES, \ - .formats = SNDRV_PCM_FMTBIT_S16_LE, \ + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE, \ }, \ .ops = &omap_mcbsp_dai_ops, \ .private_data = &mcbsp_data[(link_id)].bus_id, \ -- cgit v1.2.2 From 7fd1d74bfc0ecf3dfa139b47daa7941841724886 Mon Sep 17 00:00:00 2001 From: Felipe Balbi Date: Mon, 17 May 2010 14:21:45 +0300 Subject: ASoC: tlv320dac33: Use dev_dbg in dac33_hard_power function Since the cases when the same power state would be set again handled gracefully, we do not need to use dev_warn. Signed-off-by: Felipe Balbi Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 007fe830be46..ad1795a83acb 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -352,7 +352,7 @@ static int dac33_hard_power(struct snd_soc_codec *codec, int power) /* Safety check */ if (unlikely(power == dac33->chip_power)) { - dev_warn(codec->dev, "Trying to set the same power state: %s\n", + dev_dbg(codec->dev, "Trying to set the same power state: %s\n", power ? "ON" : "OFF"); goto exit; } -- cgit v1.2.2 From 2d4cdd6fc974716555fcbaf7ec1d4dda22784c1b Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Mon, 17 May 2010 14:21:46 +0300 Subject: ASoC: tlv320dac33: Avoid powering off while in BIAS_OFF Avoid calling the dac33_hard_power when the codec was already in BIAS_OFF state. This could happen in device suspend and module removal time. Signed-off-by: Peter Ujfalusi Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/tlv320dac33.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index ad1795a83acb..bcf6d934499a 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -589,6 +589,9 @@ static int dac33_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: + /* Do not power off, when the codec is already off */ + if (codec->bias_level == SND_SOC_BIAS_OFF) + return 0; ret = dac33_hard_power(codec, 0); if (ret != 0) return ret; -- cgit v1.2.2 From 5e64d6aadd2b8d5d480c3ff402f5dbbd009f58c5 Mon Sep 17 00:00:00 2001 From: Misael Lopez Cruz Date: Mon, 17 May 2010 19:53:10 -0500 Subject: ASoC: SDP4430: Add sdp4430 machine driver Add ASoC support for TI SDP4430. Signed-off-by: Misael Lopez Cruz Signed-off-by: Margarita Olaya Cabrera Signed-off-by: Jorge Eduardo Candelaria Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/Kconfig | 9 ++ sound/soc/omap/Makefile | 2 + sound/soc/omap/sdp4430.c | 228 +++++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 239 insertions(+) create mode 100644 sound/soc/omap/sdp4430.c (limited to 'sound/soc') diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index 83be4a76d2bb..d542ea2ff6be 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -98,6 +98,15 @@ config SND_OMAP_SOC_SDP3430 Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. +config SND_OMAP_SOC_SDP4430 + tristate "SoC Audio support for Texas Instruments SDP4430" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_4430SDP + select SND_OMAP_SOC_MCPDM + select SND_SOC_TWL6040 + help + Say Y if you want to add support for SoC audio on Texas Instruments + SDP4430. + config SND_OMAP_SOC_OMAP3_PANDORA tristate "SoC Audio support for OMAP3 Pandora" depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 3a75755f25e4..ba9fc650db28 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -17,6 +17,7 @@ snd-soc-omap2evm-objs := omap2evm.o snd-soc-omap3evm-objs := omap3evm.o snd-soc-am3517evm-objs := am3517evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-sdp4430-objs := sdp4430.o snd-soc-omap3pandora-objs := omap3pandora.o snd-soc-omap3beagle-objs := omap3beagle.o snd-soc-zoom2-objs := zoom2.o @@ -31,6 +32,7 @@ obj-$(CONFIG_SND_OMAP_SOC_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3EVM) += snd-soc-omap3evm.o obj-$(CONFIG_SND_OMAP_SOC_AM3517EVM) += snd-soc-am3517evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_SDP4430) += snd-soc-sdp4430.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c new file mode 100644 index 000000000000..83831ee35d07 --- /dev/null +++ b/sound/soc/omap/sdp4430.c @@ -0,0 +1,228 @@ +/* + * sdp4430.c -- SoC audio for TI OMAP4430 SDP + * + * Author: Misael Lopez Cruz + * + * 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. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include +#include +#include +#include +#include +#include + +#include +#include +#include + +#include "mcpdm.h" +#include "omap-mcpdm.h" +#include "omap-pcm.h" +#include "../codecs/twl6040.h" + +static int twl6040_power_mode; + +static int sdp4430_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_dai *codec_dai = rtd->dai->codec_dai; + int clk_id, freq; + int ret; + + if (twl6040_power_mode) { + clk_id = TWL6040_SYSCLK_SEL_HPPLL; + freq = 38400000; + } else { + clk_id = TWL6040_SYSCLK_SEL_LPPLL; + freq = 32768; + } + + /* set the codec mclk */ + ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq, + SND_SOC_CLOCK_IN); + if (ret) { + printk(KERN_ERR "can't set codec system clock\n"); + return ret; + } +} + +static struct snd_soc_ops sdp4430_ops = { + .hw_params = sdp4430_hw_params, +}; + +static int sdp4430_get_power_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = twl6040_power_mode; + return 0; +} + +static int sdp4430_set_power_mode(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (twl6040_power_mode == ucontrol->value.integer.value[0]) + return 0; + + twl6040_power_mode = ucontrol->value.integer.value[0]; + + return 1; +} + +static const char *power_texts[] = {"Low-Power", "High-Performance"}; + +static const struct soc_enum sdp4430_enum[] = { + SOC_ENUM_SINGLE_EXT(2, power_texts), +}; + +static const struct snd_kcontrol_new sdp4430_controls[] = { + SOC_ENUM_EXT("TWL6040 Power Mode", sdp4430_enum[0], + sdp4430_get_power_mode, sdp4430_set_power_mode), +}; + +/* SDP4430 machine DAPM */ +static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Ext Mic", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_HP("Headset Stereophone", NULL), +}; + +static const struct snd_soc_dapm_route audio_map[] = { + /* External Mics: MAINMIC, SUBMIC with bias*/ + {"MAINMIC", NULL, "Main Mic Bias"}, + {"SUBMIC", NULL, "Main Mic Bias"}, + {"Main Mic Bias", NULL, "Ext Mic"}, + + /* External Speakers: HFL, HFR */ + {"Ext Spk", NULL, "HFL"}, + {"Ext Spk", NULL, "HFR"}, + + /* Headset Mic: HSMIC with bias */ + {"HSMIC", NULL, "Headset Mic Bias"}, + {"Headset Mic Bias", NULL, "Headset Mic"}, + + /* Headset Stereophone (Headphone): HSOL, HSOR */ + {"Headset Stereophone", NULL, "HSOL"}, + {"Headset Stereophone", NULL, "HSOR"}, +}; + +static int sdp4430_twl6040_init(struct snd_soc_codec *codec) +{ + int ret; + + /* Add SDP4430 specific controls */ + ret = snd_soc_add_controls(codec, sdp4430_controls, + ARRAY_SIZE(sdp4430_controls)); + if (ret) + return ret; + + /* Add SDP4430 specific widgets */ + ret = snd_soc_dapm_new_controls(codec, sdp4430_twl6040_dapm_widgets, + ARRAY_SIZE(sdp4430_twl6040_dapm_widgets)); + if (ret) + return ret; + + /* Set up SDP4430 specific audio path audio_map */ + snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); + + /* SDP4430 connected pins */ + snd_soc_dapm_enable_pin(codec, "Ext Mic"); + snd_soc_dapm_enable_pin(codec, "Ext Spk"); + snd_soc_dapm_enable_pin(codec, "Headset Mic"); + snd_soc_dapm_enable_pin(codec, "Headset Stereophone"); + + /* TWL6040 not connected pins */ + snd_soc_dapm_nc_pin(codec, "AFML"); + snd_soc_dapm_nc_pin(codec, "AFMR"); + + ret = snd_soc_dapm_sync(codec); + + return ret; +} + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link sdp4430_dai = { + .name = "TWL6040", + .stream_name = "TWL6040", + .cpu_dai = &omap_mcpdm_dai, + .codec_dai = &twl6040_dai, + .init = sdp4430_twl6040_init, +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_sdp4430 = { + .name = "SDP4430", + .platform = &omap_soc_platform, + .dai_link = &sdp4430_dai, + .num_links = 1, +}; + +/* Audio subsystem */ +static struct snd_soc_device sdp4430_snd_devdata = { + .card = &snd_soc_sdp4430, + .codec_dev = &soc_codec_dev_twl6040, +}; + +static struct platform_device *sdp4430_snd_device; + +static int __init sdp4430_soc_init(void) +{ + int ret; + + if (!machine_is_omap_4430sdp()) { + pr_debug("Not SDP4430!\n"); + return -ENODEV; + } + printk(KERN_INFO "SDP4430 SoC init\n"); + + sdp4430_snd_device = platform_device_alloc("soc-audio", -1); + if (!sdp4430_snd_device) { + printk(KERN_ERR "Platform device allocation failed\n"); + return -ENOMEM; + } + + platform_set_drvdata(sdp4430_snd_device, &sdp4430_snd_devdata); + sdp4430_snd_devdata.dev = &sdp4430_snd_device->dev; + + ret = platform_device_add(sdp4430_snd_device); + if (ret) + goto err; + + /* Codec starts in HP mode */ + twl6040_power_mode = 1; + + return 0; + +err: + printk(KERN_ERR "Unable to add platform device\n"); + platform_device_put(sdp4430_snd_device); + return ret; +} +module_init(sdp4430_soc_init); + +static void __exit sdp4430_soc_exit(void) +{ + platform_device_unregister(sdp4430_snd_device); +} +module_exit(sdp4430_soc_exit); + +MODULE_AUTHOR("Misael Lopez Cruz "); +MODULE_DESCRIPTION("ALSA SoC SDP4430"); +MODULE_LICENSE("GPL"); + -- cgit v1.2.2 From 7254e2bddc96c47cb5edd30cfd3e1f12a2df9149 Mon Sep 17 00:00:00 2001 From: Jorge Eduardo Candelaria Date: Tue, 18 May 2010 12:44:17 -0500 Subject: ASoC: SDP4430: Add support for Earphone speaker Enable earphone speaker in sdp4430 machine driver. Signed-off-by: Jorge Eduardo Candelaria Signed-off-by: Margarita Olaya Cabrera Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/omap/sdp4430.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 83831ee35d07..3a0c19866121 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -101,6 +101,7 @@ static const struct snd_soc_dapm_widget sdp4430_twl6040_dapm_widgets[] = { SND_SOC_DAPM_SPK("Ext Spk", NULL), SND_SOC_DAPM_MIC("Headset Mic", NULL), SND_SOC_DAPM_HP("Headset Stereophone", NULL), + SND_SOC_DAPM_SPK("Earphone Spk", NULL), }; static const struct snd_soc_dapm_route audio_map[] = { @@ -120,6 +121,9 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Headset Stereophone (Headphone): HSOL, HSOR */ {"Headset Stereophone", NULL, "HSOL"}, {"Headset Stereophone", NULL, "HSOR"}, + + /* Earphone speaker */ + {"Earphone Spk", NULL, "EP"}, }; static int sdp4430_twl6040_init(struct snd_soc_codec *codec) -- cgit v1.2.2 From 871a05a78b4879d768bc2fde4b75439f517e2839 Mon Sep 17 00:00:00 2001 From: Jorge Eduardo Candelaria Date: Tue, 18 May 2010 12:44:18 -0500 Subject: ASoC: TWL6040: Enable earphone path in codec Add control to enable earphone driver in TWL6040 codec. This driver is connected to HSDAC Left. Signed-off-by: Jorge Eduardo Candelaria Signed-off-by: Margarita Olaya Cabrera Acked-by: Mark Brown Signed-off-by: Liam Girdwood --- sound/soc/codecs/twl6040.c | 21 ++++++++++++++++++++- 1 file changed, 20 insertions(+), 1 deletion(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 2ae442edeb9a..af36346ff336 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -432,6 +432,12 @@ static DECLARE_TLV_DB_SCALE(hs_tlv, -3000, 200, 0); */ static DECLARE_TLV_DB_SCALE(hf_tlv, -5200, 200, 0); +/* + * EPGAIN volume control: + * from -24 to 6 dB in 2 dB steps + */ +static DECLARE_TLV_DB_SCALE(ep_tlv, -2400, 200, 0); + /* Left analog microphone selection */ static const char *twl6040_amicl_texts[] = {"Headset Mic", "Main Mic", "Aux/FM Left", "Off"}; @@ -479,6 +485,9 @@ static const struct snd_kcontrol_new hfl_driver_switch_controls = static const struct snd_kcontrol_new hfr_driver_switch_controls = SOC_DAPM_SINGLE("Switch", TWL6040_REG_HFRCTL, 4, 1, 0); +static const struct snd_kcontrol_new ep_driver_switch_controls = + SOC_DAPM_SINGLE("Switch", TWL6040_REG_EARCTL, 0, 1, 0); + static const struct snd_kcontrol_new twl6040_snd_controls[] = { /* Capture gains */ SOC_DOUBLE_TLV("Capture Preamplifier Volume", @@ -491,7 +500,8 @@ static const struct snd_kcontrol_new twl6040_snd_controls[] = { TWL6040_REG_HSGAIN, 0, 4, 0xF, 1, hs_tlv), SOC_DOUBLE_R_TLV("Handsfree Playback Volume", TWL6040_REG_HFLGAIN, TWL6040_REG_HFRGAIN, 0, 0x1D, 1, hf_tlv), - + SOC_SINGLE_TLV("Earphone Playback Volume", + TWL6040_REG_EARCTL, 1, 0xF, 1, ep_tlv), }; static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { @@ -507,6 +517,7 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HSOR"), SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), + SND_SOC_DAPM_OUTPUT("EP"), /* Analog input muxes for the capture amplifiers */ SND_SOC_DAPM_MUX("Analog Left Capture Route", @@ -572,6 +583,10 @@ static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = { SND_SOC_NOPM, 0, 0, &hfr_driver_switch_controls, twl6040_power_mode_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("Earphone Driver", + SND_SOC_NOPM, 0, 0, &ep_driver_switch_controls, + twl6040_power_mode_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* Analog playback PGAs */ SND_SOC_DAPM_PGA("HFDAC Left PGA", @@ -607,6 +622,10 @@ static const struct snd_soc_dapm_route intercon[] = { {"HSOL", NULL, "Headset Left Driver"}, {"HSOR", NULL, "Headset Right Driver"}, + /* Earphone playback path */ + {"Earphone Driver", "Switch", "HSDAC Left"}, + {"EP", NULL, "Earphone Driver"}, + /* Handsfree playback path */ {"HFDAC Left Playback", "Switch", "HFDAC Left"}, {"HFDAC Right Playback", "Switch", "HFDAC Right"}, -- cgit v1.2.2 From d8b55d2cd00df4d599985440fd75b38d153bffcb Mon Sep 17 00:00:00 2001 From: Liam Girdwood Date: Wed, 19 May 2010 14:14:51 +0100 Subject: ASoC: sdp4430 - add sdp4430 pcm ops to DAI. Fix build warning about unused ops and add ops to the sdp4430 DAI link. Signed-off-by: Liam Girdwood --- sound/soc/omap/sdp4430.c | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc') diff --git a/sound/soc/omap/sdp4430.c b/sound/soc/omap/sdp4430.c index 3a0c19866121..4ebbde6b565f 100644 --- a/sound/soc/omap/sdp4430.c +++ b/sound/soc/omap/sdp4430.c @@ -167,6 +167,7 @@ static struct snd_soc_dai_link sdp4430_dai = { .cpu_dai = &omap_mcpdm_dai, .codec_dai = &twl6040_dai, .init = sdp4430_twl6040_init, + .ops = &sdp4430_ops, }; /* Audio machine driver */ -- cgit v1.2.2 From 266d38c8e3d7f62152b1448fd9a7265f32f32d87 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 19 May 2010 13:55:26 +0300 Subject: ASoC: tpa6130a2: Define output pins with SND_SOC_DAPM_OUTPUT Codec output pin should be defined with SND_SOC_DAPM_OUTPUT as otherwise external widgets doesn't alter the output state. Signed-off-by: Jarkko Nikula Cc: Peter Ujfalusi Signed-off-by: Liam Girdwood --- sound/soc/codecs/tpa6130a2.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index fa4fa33a51aa..89788921280e 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -371,8 +371,8 @@ static const struct snd_soc_dapm_widget tpa6130a2_dapm_widgets[] = { 0, 0, tpa6130a2_supply_event, SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), /* Outputs */ - SND_SOC_DAPM_HP("TPA6130A2 Headphone Left", NULL), - SND_SOC_DAPM_HP("TPA6130A2 Headphone Right", NULL), + SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Left"), + SND_SOC_DAPM_OUTPUT("TPA6130A2 Headphone Right"), }; static const struct snd_soc_dapm_route audio_map[] = { -- cgit v1.2.2 From ad8332c1302bcb4f80d593fd3eb477be9d7f5604 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula Date: Wed, 19 May 2010 10:52:28 +0300 Subject: ASoC: tpa6130a2: Remove CPVSS and HPVdd supplies These pins are for decoupling capacitors for the internal charge pumps in TPA6130A2 and TPA6140A2 and not for connecting external supply. Thanks to Eduardo Valentin for pointing out the issue with TPA6130A2 and Ilkka Koskinen with TPA6140A2. Signed-off-by: Jarkko Nikula Acked-by: Peter Ujfalusi Reviewed-by: Ilkka Koskinen Signed-off-by: Liam Girdwood --- sound/soc/codecs/tpa6130a2.c | 52 +++++++++++++++----------------------------- 1 file changed, 18 insertions(+), 34 deletions(-) (limited to 'sound/soc') diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 89788921280e..20ac67700395 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -35,22 +35,11 @@ static struct i2c_client *tpa6130a2_client; -#define TPA6130A2_NUM_SUPPLIES 2 -static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { - "CPVSS", - "Vdd", -}; - -static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { - "HPVdd", - "AVdd", -}; - /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; unsigned char regs[TPA6130A2_CACHEREGNUM]; - struct regulator_bulk_data supplies[TPA6130A2_NUM_SUPPLIES]; + struct regulator *supply; int power_gpio; unsigned char power_state; enum tpa_model id; @@ -135,11 +124,10 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 1); - ret = regulator_bulk_enable(ARRAY_SIZE(data->supplies), - data->supplies); + ret = regulator_enable(data->supply); if (ret != 0) { dev_err(&tpa6130a2_client->dev, - "Failed to enable supplies: %d\n", ret); + "Failed to enable supply: %d\n", ret); goto exit; } @@ -160,11 +148,10 @@ static int tpa6130a2_power(int power) if (data->power_gpio >= 0) gpio_set_value(data->power_gpio, 0); - ret = regulator_bulk_disable(ARRAY_SIZE(data->supplies), - data->supplies); + ret = regulator_disable(data->supply); if (ret != 0) { dev_err(&tpa6130a2_client->dev, - "Failed to disable supplies: %d\n", ret); + "Failed to disable supply: %d\n", ret); goto exit; } @@ -411,7 +398,8 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, struct device *dev; struct tpa6130a2_data *data; struct tpa6130a2_platform_data *pdata; - int i, ret; + const char *regulator; + int ret; dev = &client->dev; @@ -453,25 +441,21 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, } switch (data->id) { + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); case TPA6130A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + regulator = "Vdd"; break; case TPA6140A2: - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6140a2_supply_names[i];; + regulator = "AVdd"; break; - default: - dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", - pdata->id); - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; } - ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), - data->supplies); - if (ret != 0) { - dev_err(dev, "Failed to request supplies: %d\n", ret); + data->supply = regulator_get(dev, regulator); + if (IS_ERR(data->supply)) { + ret = PTR_ERR(data->supply); + dev_err(dev, "Failed to request supply: %d\n", ret); goto err_regulator; } @@ -494,7 +478,7 @@ static int __devinit tpa6130a2_probe(struct i2c_client *client, return 0; err_power: - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_put(data->supply); err_regulator: if (data->power_gpio >= 0) gpio_free(data->power_gpio); @@ -515,7 +499,7 @@ static int __devexit tpa6130a2_remove(struct i2c_client *client) if (data->power_gpio >= 0) gpio_free(data->power_gpio); - regulator_bulk_free(ARRAY_SIZE(data->supplies), data->supplies); + regulator_put(data->supply); kfree(data); tpa6130a2_client = NULL; -- cgit v1.2.2