November 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2 RESEND] ASoC: wm9081: Use snd_soc_update_bits for read-modify-write
by Axel Lin 05 Nov '11

05 Nov '11

Use snd_soc_update_bits for read-modify-write register access instead of open-coding it using snd_soc_read and snd_soc_write Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- RESEND: I just found I doesn't set my mail client properly to show correct sender, so here is a resend. v2: For Initial cold start ( change bias level from SND_SOC_BIAS_OFF to SND_SOC_BIAS_STANDBY ), the v1 does not set VMID_SEL[1:0] bits correctly for Normal bias enable & soft start off. In v2 we also clear VMID_SEL[1:0] bits for Normal bias enable & soft start off. We did clear VMID_SEL[1:0] bits in SND_SOC_BIAS_OFF, thus we know the VMID_SEL[1:0] bits status is 0b00 in this case. sound/soc/codecs/wm9081.c | 82 +++++++++++++++++++------------------------- 1 files changed, 35 insertions(+), 47 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 66092ef..9c38ddb 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -766,84 +766,73 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { static int wm9081_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { - u16 reg; - switch (level) { case SND_SOC_BIAS_ON: break; case SND_SOC_BIAS_PREPARE: /* VMID=2*40k */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x2; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_SEL_MASK, 0x2); /* Normal bias current */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, 0); break; case SND_SOC_BIAS_STANDBY: /* Initial cold start */ if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { /* Disable LINEOUT discharge */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg &= ~WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, 0); /* Select startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC | WM9081_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC | WM9081_BIAS_ENA, + WM9081_BIAS_SRC | WM9081_BIAS_ENA); /* VMID 2*4k; Soft VMID ramp enable */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg |= WM9081_VMID_RAMP | 0x6; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); - + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | 0x6, + WM9081_VMID_RAMP | 0x6); mdelay(100); /* Normal bias enable & soft start off */ - reg |= WM9081_BIAS_ENA; - reg &= ~WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_BIAS_ENA | + WM9081_VMID_SEL_MASK, + WM9081_BIAS_ENA); /* Standard bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, 0); } /* VMID 2*240k */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg &= ~WM9081_VMID_SEL_MASK; - reg |= 0x04; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_VMID_SEL_MASK, 0x04); /* Standby bias current on */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_STBY_BIAS_ENA; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_STBY_BIAS_ENA, + WM9081_STBY_BIAS_ENA); break; case SND_SOC_BIAS_OFF: /* Startup bias source */ - reg = snd_soc_read(codec, WM9081_BIAS_CONTROL_1); - reg |= WM9081_BIAS_SRC; - snd_soc_write(codec, WM9081_BIAS_CONTROL_1, reg); + snd_soc_update_bits(codec, WM9081_BIAS_CONTROL_1, + WM9081_BIAS_SRC, WM9081_BIAS_SRC); /* Disable VMID and biases with soft ramping */ - reg = snd_soc_read(codec, WM9081_VMID_CONTROL); - reg &= ~(WM9081_VMID_SEL_MASK | WM9081_BIAS_ENA); - reg |= WM9081_VMID_RAMP; - snd_soc_write(codec, WM9081_VMID_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_VMID_CONTROL, + WM9081_VMID_RAMP | WM9081_VMID_SEL_MASK | + WM9081_BIAS_ENA, WM9081_VMID_RAMP); /* Actively discharge LINEOUT */ - reg = snd_soc_read(codec, WM9081_ANTI_POP_CONTROL); - reg |= WM9081_LINEOUT_DISCH; - snd_soc_write(codec, WM9081_ANTI_POP_CONTROL, reg); + snd_soc_update_bits(codec, WM9081_ANTI_POP_CONTROL, + WM9081_LINEOUT_DISCH, + WM9081_LINEOUT_DISCH); break; } @@ -1242,11 +1231,10 @@ static int wm9081_probe(struct snd_soc_codec *codec) wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Enable zero cross by default */ - reg = snd_soc_read(codec, WM9081_ANALOGUE_LINEOUT); - snd_soc_write(codec, WM9081_ANALOGUE_LINEOUT, reg | WM9081_LINEOUTZC); - reg = snd_soc_read(codec, WM9081_ANALOGUE_SPEAKER_PGA); - snd_soc_write(codec, WM9081_ANALOGUE_SPEAKER_PGA, - reg | WM9081_SPKPGAZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, + WM9081_LINEOUTZC, WM9081_LINEOUTZC); + snd_soc_update_bits(codec, WM9081_ANALOGUE_SPEAKER_PGA, + WM9081_SPKPGAZC, WM9081_SPKPGAZC); snd_soc_add_controls(codec, wm9081_snd_controls, ARRAY_SIZE(wm9081_snd_controls)); -- 1.7.5.4

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[alsa-devel] [PATCH] ASoC: Disable thermal shutdown when not using speakers in wm_hubs
by Mark Brown 05 Nov '11

05 Nov '11

The thermal shutdown support in wm_hubs devices is tied to the speaker drivers (which are the only high power subsystems within the device). Ensure minimal current usage when the thermal shutdown support is not required by disabling the circuit when the speaker drivers are powered down. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 84f33d4..f98170e 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -654,6 +654,7 @@ SND_SOC_DAPM_MIXER("SPKL Boost", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_MIXER("SPKR Boost", SND_SOC_NOPM, 0, 0, right_speaker_boost, ARRAY_SIZE(right_speaker_boost)), +SND_SOC_DAPM_SUPPLY("TSHUT", WM8993_POWER_MANAGEMENT_2, 14, 0, NULL, 0), SND_SOC_DAPM_PGA("SPKL Driver", WM8993_POWER_MANAGEMENT_1, 12, 0, NULL, 0), SND_SOC_DAPM_PGA("SPKR Driver", WM8993_POWER_MANAGEMENT_1, 13, 0, @@ -789,10 +790,12 @@ static const struct snd_soc_dapm_route analogue_routes[] = { { "SPKL Driver", NULL, "VMID" }, { "SPKL Driver", NULL, "SPKL Boost" }, { "SPKL Driver", NULL, "CLK_SYS" }, + { "SPKL Driver", NULL, "TSHUT" }, { "SPKR Driver", NULL, "VMID" }, { "SPKR Driver", NULL, "SPKR Boost" }, { "SPKR Driver", NULL, "CLK_SYS" }, + { "SPKR Driver", NULL, "TSHUT" }, { "SPKOUTLP", NULL, "SPKL Driver" }, { "SPKOUTLN", NULL, "SPKL Driver" }, -- 1.7.7.1

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[alsa-devel] [PATCH] ASoC: Only enable thermal shutdown when required on WM9081
by Mark Brown 04 Nov '11

04 Nov '11

The WM9081 thermal shutdown is only effective when the speaker output is enabled so disable it when that is not in use for a small current saving. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 550bf92..a4570e7 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -796,6 +796,7 @@ SND_SOC_DAPM_SUPPLY("CLK_SYS", WM9081_CLOCK_CONTROL_3, 0, 0, clk_sys_event, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SUPPLY("CLK_DSP", WM9081_CLOCK_CONTROL_3, 1, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("TOCLK", WM9081_CLOCK_CONTROL_3, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TSENSE", WM9081_POWER_MANAGEMENT, 7, 0, NULL, 0), }; @@ -818,6 +819,7 @@ static const struct snd_soc_dapm_route wm9081_audio_paths[] = { { "Speaker PGA", NULL, "CLK_SYS" }, { "Speaker", NULL, "Speaker PGA" }, + { "Speaker", NULL, "TSENSE" }, { "SPKN", NULL, "Speaker" }, { "SPKP", NULL, "Speaker" }, -- 1.7.7.1

1 0

[alsa-devel] [PATCH] ASoC: Convert WM9081 to direct regmap API usage
by Mark Brown 04 Nov '11

04 Nov '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 280 +++++++++++++++++++++++++++++---------------- 1 files changed, 182 insertions(+), 98 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 4e0489f..09d6da0 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -19,6 +19,7 @@ #include <linux/pm.h> #include <linux/i2c.h> #include <linux/platform_device.h> +#include <linux/regmap.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -30,69 +31,63 @@ #include <sound/wm9081.h> #include "wm9081.h" -static u16 wm9081_reg_defaults[] = { - 0x0000, /* R0 - Software Reset */ - 0x0000, /* R1 */ - 0x00B9, /* R2 - Analogue Lineout */ - 0x00B9, /* R3 - Analogue Speaker PGA */ - 0x0001, /* R4 - VMID Control */ - 0x0068, /* R5 - Bias Control 1 */ - 0x0000, /* R6 */ - 0x0000, /* R7 - Analogue Mixer */ - 0x0000, /* R8 - Anti Pop Control */ - 0x01DB, /* R9 - Analogue Speaker 1 */ - 0x0018, /* R10 - Analogue Speaker 2 */ - 0x0180, /* R11 - Power Management */ - 0x0000, /* R12 - Clock Control 1 */ - 0x0038, /* R13 - Clock Control 2 */ - 0x4000, /* R14 - Clock Control 3 */ - 0x0000, /* R15 */ - 0x0000, /* R16 - FLL Control 1 */ - 0x0200, /* R17 - FLL Control 2 */ - 0x0000, /* R18 - FLL Control 3 */ - 0x0204, /* R19 - FLL Control 4 */ - 0x0000, /* R20 - FLL Control 5 */ - 0x0000, /* R21 */ - 0x0000, /* R22 - Audio Interface 1 */ - 0x0002, /* R23 - Audio Interface 2 */ - 0x0008, /* R24 - Audio Interface 3 */ - 0x0022, /* R25 - Audio Interface 4 */ - 0x0000, /* R26 - Interrupt Status */ - 0x0006, /* R27 - Interrupt Status Mask */ - 0x0000, /* R28 - Interrupt Polarity */ - 0x0000, /* R29 - Interrupt Control */ - 0x00C0, /* R30 - DAC Digital 1 */ - 0x0008, /* R31 - DAC Digital 2 */ - 0x09AF, /* R32 - DRC 1 */ - 0x4201, /* R33 - DRC 2 */ - 0x0000, /* R34 - DRC 3 */ - 0x0000, /* R35 - DRC 4 */ - 0x0000, /* R36 */ - 0x0000, /* R37 */ - 0x0000, /* R38 - Write Sequencer 1 */ - 0x0000, /* R39 - Write Sequencer 2 */ - 0x0002, /* R40 - MW Slave 1 */ - 0x0000, /* R41 */ - 0x0000, /* R42 - EQ 1 */ - 0x0000, /* R43 - EQ 2 */ - 0x0FCA, /* R44 - EQ 3 */ - 0x0400, /* R45 - EQ 4 */ - 0x00B8, /* R46 - EQ 5 */ - 0x1EB5, /* R47 - EQ 6 */ - 0xF145, /* R48 - EQ 7 */ - 0x0B75, /* R49 - EQ 8 */ - 0x01C5, /* R50 - EQ 9 */ - 0x169E, /* R51 - EQ 10 */ - 0xF829, /* R52 - EQ 11 */ - 0x07AD, /* R53 - EQ 12 */ - 0x1103, /* R54 - EQ 13 */ - 0x1C58, /* R55 - EQ 14 */ - 0xF373, /* R56 - EQ 15 */ - 0x0A54, /* R57 - EQ 16 */ - 0x0558, /* R58 - EQ 17 */ - 0x0564, /* R59 - EQ 18 */ - 0x0559, /* R60 - EQ 19 */ - 0x4000, /* R61 - EQ 20 */ +static struct reg_default wm9081_reg[] = { + { 0, 0x9081 }, /* R0 - Software Reset */ + { 1, 0x0000 }, /* R1 */ + { 2, 0x00B9 }, /* R2 - Analogue Lineout */ + { 3, 0x00B9 }, /* R3 - Analogue Speaker PGA */ + { 4, 0x0001 }, /* R4 - VMID Control */ + { 5, 0x0068 }, /* R5 - Bias Control 1 */ + { 6, 0x0000 }, /* R6 */ + { 7, 0x0000 }, /* R7 - Analogue Mixer */ + { 8, 0x0000 }, /* R8 - Anti Pop Control */ + { 9, 0x01DB }, /* R9 - Analogue Speaker 1 */ + { 10, 0x0018 }, /* R10 - Analogue Speaker 2 */ + { 11, 0x0180 }, /* R11 - Power Management */ + { 12, 0x0000 }, /* R12 - Clock Control 1 */ + { 13, 0x0038 }, /* R13 - Clock Control 2 */ + { 14, 0x4000 }, /* R14 - Clock Control 3 */ + { 16, 0x0000 }, /* R16 - FLL Control 1 */ + { 17, 0x0200 }, /* R17 - FLL Control 2 */ + { 18, 0x0000 }, /* R18 - FLL Control 3 */ + { 19, 0x0204 }, /* R19 - FLL Control 4 */ + { 20, 0x0000 }, /* R20 - FLL Control 5 */ + { 22, 0x0000 }, /* R22 - Audio Interface 1 */ + { 23, 0x0002 }, /* R23 - Audio Interface 2 */ + { 24, 0x0008 }, /* R24 - Audio Interface 3 */ + { 25, 0x0022 }, /* R25 - Audio Interface 4 */ + { 27, 0x0006 }, /* R27 - Interrupt Status Mask */ + { 28, 0x0000 }, /* R28 - Interrupt Polarity */ + { 29, 0x0000 }, /* R29 - Interrupt Control */ + { 30, 0x00C0 }, /* R30 - DAC Digital 1 */ + { 31, 0x0008 }, /* R31 - DAC Digital 2 */ + { 32, 0x09AF }, /* R32 - DRC 1 */ + { 33, 0x4201 }, /* R33 - DRC 2 */ + { 34, 0x0000 }, /* R34 - DRC 3 */ + { 35, 0x0000 }, /* R35 - DRC 4 */ + { 38, 0x0000 }, /* R38 - Write Sequencer 1 */ + { 39, 0x0000 }, /* R39 - Write Sequencer 2 */ + { 40, 0x0002 }, /* R40 - MW Slave 1 */ + { 42, 0x0000 }, /* R42 - EQ 1 */ + { 43, 0x0000 }, /* R43 - EQ 2 */ + { 44, 0x0FCA }, /* R44 - EQ 3 */ + { 45, 0x0400 }, /* R45 - EQ 4 */ + { 46, 0x00B8 }, /* R46 - EQ 5 */ + { 47, 0x1EB5 }, /* R47 - EQ 6 */ + { 48, 0xF145 }, /* R48 - EQ 7 */ + { 49, 0x0B75 }, /* R49 - EQ 8 */ + { 50, 0x01C5 }, /* R50 - EQ 9 */ + { 51, 0x169E }, /* R51 - EQ 10 */ + { 52, 0xF829 }, /* R52 - EQ 11 */ + { 53, 0x07AD }, /* R53 - EQ 12 */ + { 54, 0x1103 }, /* R54 - EQ 13 */ + { 55, 0x1C58 }, /* R55 - EQ 14 */ + { 56, 0xF373 }, /* R56 - EQ 15 */ + { 57, 0x0A54 }, /* R57 - EQ 16 */ + { 58, 0x0558 }, /* R58 - EQ 17 */ + { 59, 0x0564 }, /* R59 - EQ 18 */ + { 60, 0x0559 }, /* R60 - EQ 19 */ + { 61, 0x4000 }, /* R61 - EQ 20 */ }; static struct { @@ -156,7 +151,7 @@ static struct { }; struct wm9081_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; int sysclk_source; int mclk_rate; int sysclk_rate; @@ -169,20 +164,84 @@ struct wm9081_priv { struct wm9081_pdata pdata; }; -static int wm9081_volatile_register(struct snd_soc_codec *codec, unsigned int reg) +static bool wm9081_volatile_register(struct device *dev, unsigned int reg) { switch (reg) { case WM9081_SOFTWARE_RESET: case WM9081_INTERRUPT_STATUS: - return 1; + return true; default: - return 0; + return false; } } -static int wm9081_reset(struct snd_soc_codec *codec) +static bool wm9081_readable_register(struct device *dev, unsigned int reg) { - return snd_soc_write(codec, WM9081_SOFTWARE_RESET, 0); + switch (reg) { + case WM9081_SOFTWARE_RESET: + case WM9081_ANALOGUE_LINEOUT: + case WM9081_ANALOGUE_SPEAKER_PGA: + case WM9081_VMID_CONTROL: + case WM9081_BIAS_CONTROL_1: + case WM9081_ANALOGUE_MIXER: + case WM9081_ANTI_POP_CONTROL: + case WM9081_ANALOGUE_SPEAKER_1: + case WM9081_ANALOGUE_SPEAKER_2: + case WM9081_POWER_MANAGEMENT: + case WM9081_CLOCK_CONTROL_1: + case WM9081_CLOCK_CONTROL_2: + case WM9081_CLOCK_CONTROL_3: + case WM9081_FLL_CONTROL_1: + case WM9081_FLL_CONTROL_2: + case WM9081_FLL_CONTROL_3: + case WM9081_FLL_CONTROL_4: + case WM9081_FLL_CONTROL_5: + case WM9081_AUDIO_INTERFACE_1: + case WM9081_AUDIO_INTERFACE_2: + case WM9081_AUDIO_INTERFACE_3: + case WM9081_AUDIO_INTERFACE_4: + case WM9081_INTERRUPT_STATUS: + case WM9081_INTERRUPT_STATUS_MASK: + case WM9081_INTERRUPT_POLARITY: + case WM9081_INTERRUPT_CONTROL: + case WM9081_DAC_DIGITAL_1: + case WM9081_DAC_DIGITAL_2: + case WM9081_DRC_1: + case WM9081_DRC_2: + case WM9081_DRC_3: + case WM9081_DRC_4: + case WM9081_WRITE_SEQUENCER_1: + case WM9081_WRITE_SEQUENCER_2: + case WM9081_MW_SLAVE_1: + case WM9081_EQ_1: + case WM9081_EQ_2: + case WM9081_EQ_3: + case WM9081_EQ_4: + case WM9081_EQ_5: + case WM9081_EQ_6: + case WM9081_EQ_7: + case WM9081_EQ_8: + case WM9081_EQ_9: + case WM9081_EQ_10: + case WM9081_EQ_11: + case WM9081_EQ_12: + case WM9081_EQ_13: + case WM9081_EQ_14: + case WM9081_EQ_15: + case WM9081_EQ_16: + case WM9081_EQ_17: + case WM9081_EQ_18: + case WM9081_EQ_19: + case WM9081_EQ_20: + return true; + default: + return false; + } +} + +static int wm9081_reset(struct regmap *map) +{ + return regmap_write(map, WM9081_SOFTWARE_RESET, 0x9081); } static const DECLARE_TLV_DB_SCALE(drc_in_tlv, -4500, 75, 0); @@ -1202,25 +1261,14 @@ static int wm9081_probe(struct snd_soc_codec *codec) int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, wm9081->control_type); + codec->control_data = wm9081->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } - reg = snd_soc_read(codec, WM9081_SOFTWARE_RESET); - if (reg != 0x9081) { - dev_err(codec->dev, "Device is not a WM9081: ID=0x%x\n", reg); - ret = -EINVAL; - return ret; - } - - ret = wm9081_reset(codec); - if (ret < 0) { - dev_err(codec->dev, "Failed to issue reset\n"); - return ret; - } - reg = 0; if (wm9081->pdata.irq_high) reg |= WM9081_IRQ_POL; @@ -1263,15 +1311,9 @@ static int wm9081_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm9081_resume(struct snd_soc_codec *codec) { - u16 *reg_cache = codec->reg_cache; - int i; - - for (i = 0; i < codec->driver->reg_cache_size; i++) { - if (i == WM9081_SOFTWARE_RESET) - continue; + struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - snd_soc_write(codec, i, reg_cache[i]); - } + regcache_sync(wm9081->regmap); wm9081_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -1291,11 +1333,6 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .set_sysclk = wm9081_set_sysclk, .set_bias_level = wm9081_set_bias_level, - .reg_cache_size = ARRAY_SIZE(wm9081_reg_defaults), - .reg_word_size = sizeof(u16), - .reg_cache_default = wm9081_reg_defaults, - .volatile_register = wm9081_volatile_register, - .controls = wm9081_snd_controls, .num_controls = ARRAY_SIZE(wm9081_snd_controls), .dapm_widgets = wm9081_dapm_widgets, @@ -1304,11 +1341,24 @@ static struct snd_soc_codec_driver soc_codec_dev_wm9081 = { .num_dapm_routes = ARRAY_SIZE(wm9081_audio_paths), }; +static const struct regmap_config wm9081_regmap = { + .reg_bits = 8, + .val_bits = 16, + + .max_register = WM9081_MAX_REGISTER, + .reg_defaults = wm9081_reg, + .num_reg_defaults = ARRAY_SIZE(wm9081_reg), + .volatile_reg = wm9081_volatile_register, + .readable_reg = wm9081_readable_register, + .cache_type = REGCACHE_RBTREE, +}; + #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct wm9081_priv *wm9081; + unsigned int reg; int ret; wm9081 = kzalloc(sizeof(struct wm9081_priv), GFP_KERNEL); @@ -1316,7 +1366,30 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, return -ENOMEM; i2c_set_clientdata(i2c, wm9081); - wm9081->control_type = SND_SOC_I2C; + + wm9081->regmap = regmap_init_i2c(i2c, &wm9081_regmap); + if (IS_ERR(wm9081->regmap)) { + ret = PTR_ERR(wm9081->regmap); + dev_err(&i2c->dev, "regmap_init() failed: %d\n", ret); + goto err; + } + + ret = regmap_read(wm9081->regmap, WM9081_SOFTWARE_RESET, &reg); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to read chip ID: %d\n", ret); + goto err_regmap; + } + if (reg != 0x9081) { + dev_err(&i2c->dev, "Device is not a WM9081: ID=0x%x\n", reg); + ret = -EINVAL; + goto err_regmap; + } + + ret = wm9081_reset(wm9081->regmap); + if (ret < 0) { + dev_err(&i2c->dev, "Failed to issue reset\n"); + goto err_regmap; + } if (dev_get_platdata(&i2c->dev)) memcpy(&wm9081->pdata, dev_get_platdata(&i2c->dev), @@ -1325,13 +1398,24 @@ static __devinit int wm9081_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_wm9081, &wm9081_dai, 1); if (ret < 0) - kfree(wm9081); + goto err_regmap; + + return 0; + +err_regmap: + regmap_exit(wm9081->regmap); +err: + kfree(wm9081); + return ret; } static __devexit int wm9081_i2c_remove(struct i2c_client *client) { + struct wm9081_priv *wm9081 = i2c_get_clientdata(client); + snd_soc_unregister_codec(&client->dev); + regmap_exit(wm9081->regmap); kfree(i2c_get_clientdata(client)); return 0; } -- 1.7.7.1

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[alsa-devel] [PATCH] ASoC: Manage thermal shutdown for WM8962
by Mark Brown 04 Nov '11

04 Nov '11

Disable the thermal shutdown circuits for headphone and speaker when the relevant outputs are not enabled in order to save current in idle modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8962.c | 8 ++++++++ 1 files changed, 8 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 5202eff..dc85b85 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2692,6 +2692,8 @@ SND_SOC_DAPM_SUPPLY("TOCLK", WM8962_ADDITIONAL_CONTROL_1, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY_S("DSP2", 1, WM8962_DSP2_POWER_MANAGEMENT, WM8962_DSP2_ENA_SHIFT, 0, dsp2_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), +SND_SOC_DAPM_SUPPLY("TEMP_HP", WM8962_ADDITIONAL_CONTROL_4, 2, 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("TEMP_SPK", WM8962_ADDITIONAL_CONTROL_4, 1, 0, NULL, 0), SND_SOC_DAPM_MIXER("INPGAL", WM8962_LEFT_INPUT_PGA_CONTROL, 4, 0, inpgal, ARRAY_SIZE(inpgal)), @@ -2847,6 +2849,9 @@ static const struct snd_soc_dapm_route wm8962_intercon[] = { { "HPOUTL", NULL, "HPOUT" }, { "HPOUTR", NULL, "HPOUT" }, + + { "HPOUTL", NULL, "TEMP_HP" }, + { "HPOUTR", NULL, "TEMP_HP" }, }; static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { @@ -2863,6 +2868,7 @@ static const struct snd_soc_dapm_route wm8962_spk_mono_intercon[] = { { "Speaker Output", NULL, "Speaker PGA" }, { "Speaker Output", NULL, "SYSCLK" }, { "Speaker Output", NULL, "TOCLK" }, + { "Speaker Output", NULL, "TEMP_SPK" }, { "SPKOUT", NULL, "Speaker Output" }, }; @@ -2891,10 +2897,12 @@ static const struct snd_soc_dapm_route wm8962_spk_stereo_intercon[] = { { "SPKOUTL Output", NULL, "SPKOUTL PGA" }, { "SPKOUTL Output", NULL, "SYSCLK" }, { "SPKOUTL Output", NULL, "TOCLK" }, + { "SPKOUTL Output", NULL, "TEMP_SPK" }, { "SPKOUTR Output", NULL, "SPKOUTR PGA" }, { "SPKOUTR Output", NULL, "SYSCLK" }, { "SPKOUTR Output", NULL, "TOCLK" }, + { "SPKOUTR Output", NULL, "TEMP_SPK" }, { "SPKOUTL", NULL, "SPKOUTL Output" }, { "SPKOUTR", NULL, "SPKOUTR Output" }, -- 1.7.7.1

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[alsa-devel] Notes from the Audio BoFs at Prague
by Arun Raghavan 04 Nov '11

04 Nov '11

Hello, We had some pretty productive discussions at the Audio BoFs last week in Prague (since a bunch of us were there for various conferences). I've put up notes from these discussions for those who are interested: http://arunraghavan.net/2011/11/notes-from-the-prague-audio-bofs/ Cheers, Arun p.s.: I've cross-posted to 3 lists, so please be considerate while replying and prune the ones that are not relevant if there is a specific discussion.

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[alsa-devel] need some workarround or explanation.
by Anders Gnistrup 04 Nov '11

04 Nov '11

Hello I have a USB headset, where when plugged in, a volume setting is set on playback and capture channels using the ctl interface for the device. After the volume is set I open the capture interface. but I have i have a 2-3 sec. delay that i just don't understand. I might need to use another solution, but these are difficult to find. The Headset is a plantronics. The code is listed below. #include <alsa/asoundlib.h> #define qDebug printf void set_default_volume(const char *a_hw_dev, int a_percent) { int err; snd_mixer_t *handle; snd_mixer_selem_id_t *sid; snd_mixer_elem_t *elem; snd_mixer_selem_id_alloca(&sid); if ((err = snd_mixer_open(&handle, 0)) < 0) { qDebug("Mixer open error: %s\n", snd_strerror(err)); return; } if ((err = snd_mixer_attach(handle, a_hw_dev)) < 0) { qDebug("Mixer attach %s error: %s\n", a_hw_dev, snd_strerror(err)); snd_mixer_close(handle); return; } if ((err = snd_mixer_selem_register(handle, NULL, NULL)) < 0) { qDebug("Mixer register error: %s\n", snd_strerror(err)); snd_mixer_close(handle); return; } err = snd_mixer_load(handle); if (err < 0) { qDebug("Mixer %s load error: %s\n", a_hw_dev, snd_strerror(err)); snd_mixer_close(handle); return; } for (elem = snd_mixer_first_elem(handle); elem; elem = snd_mixer_elem_next(elem)) { snd_mixer_selem_get_id(elem, sid); qDebug("Simple mixer control '%s',%i\n", snd_mixer_selem_id_get_name(sid), snd_mixer_selem_id_get_index(sid)); if (snd_mixer_selem_has_playback_switch(elem)) { snd_mixer_selem_set_playback_switch_all(elem, 1); //unmute } if (snd_mixer_selem_has_playback_volume(elem)) { long min, max; snd_mixer_selem_get_playback_volume_range(elem, &min, &max); qDebug(" pvolume %ld/%ld\n", min , max); snd_mixer_selem_set_playback_volume_all(elem, (a_percent * (max-min))/100); } if (snd_mixer_selem_has_capture_switch(elem)) { snd_mixer_selem_set_capture_switch_all(elem, 1); // unmute all } if (snd_mixer_selem_has_capture_volume(elem)) { long min, max; snd_mixer_selem_get_capture_volume_range(elem, &min, &max); qDebug(" pvolume %ld/%ld\n", min , max); snd_mixer_selem_set_capture_volume_all(elem, (a_percent * (max-min))/100); //set capture volume for all } } snd_mixer_close(handle); return; } int main(int argc, char *argv[]) { snd_pcm_t *m_pcm_handle; int err,count; // Open connection to the PCM devic if ((err = snd_pcm_open(&m_pcm_handle, "plughw:1,0", SND_PCM_STREAM_CAPTURE, 0)) < 0) { for (count=0; count < 5 && err == -EBUSY; count++) { sleep(1); qDebug("AlsaStream:%d open error: %s (%d)\n ", __LINE__, snd_strerror(err), err); err = snd_pcm_open(&m_pcm_handle, "plughw:1,0", SND_PCM_STREAM_CAPTURE, 0); } if (err < 0) { qDebug("AlsaStream:%d open error: %s (%d)\n ", __LINE__, snd_strerror(err), err); return; } if (err>= 0) { printf("Success\n"); } } set_default_volume("hw:1", 20); snd_pcm_close(m_pcm_handle); return 0; }

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[alsa-devel] Question regarding "Fix backport of WM8994 thermal warning" patch
by Thomas Abraham 04 Nov '11

04 Nov '11

Hi Mark, I have a question regarding "Fix backport of WM8994 thermal warning" patch. I am trying sound on Exynos4 smdk board with the latest 3.1 kernel. In the function "wm8994_codec_probe", the kernel crashes due to null pointer when requesting "Thermal Warning" interrupt (line# 3183 of sound/soc/codecs/wm8994.c file). wm8994->control_data is actually NULL at this point. I have not debugged this much but if there is something obvious I am missing, could you please let me know. Thanks, Thomas.

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[alsa-devel] [PATCH 1/2] ASoC: Don't use wm8994->control_data in wm8994_readable_register()
by Mark Brown 04 Nov '11

04 Nov '11

The field is no longer initialised so this will crash if running on wm8958. Reported-by: Thomas Abraham <thomas.abraham(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 6b73efd..3d2c3d4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -56,7 +56,7 @@ static int wm8994_retune_mobile_base[] = { static int wm8994_readable(struct snd_soc_codec *codec, unsigned int reg) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); - struct wm8994 *control = wm8994->control_data; + struct wm8994 *control = codec->control_data; switch (reg) { case WM8994_GPIO_1: -- 1.7.7.1

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[alsa-devel] aplay - memory leak
by Appalayagari Sreedhar 04 Nov '11

04 Nov '11

Hi, I am working with DM368 based custom board. we have external codec interfaced to McBSP. when we run aplay command i am observing memory leak. the leak is around 12k everytime i do aplay. if i get any underruns the leak is more than 12k the kernel version i am working is "Linux-2.6.32.17-davinci1". The codec driver is in alsa model. Could you please help to resolve this problem. Thanks in advance, Sreedhar.

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