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

[alsa-devel] [PATCH 6/9] ASoC: da7210: bugfix for head phone volume control
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch takes care of reserved bits of headphone volume register by using correct volume range. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 60f403b..eb12c82 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -196,12 +196,15 @@ * mute : 0x10 * reserved : 0x00 - 0x0F * - * ** FIXME ** - * * Reserved area are considered as "mute". - * -> min = -79.5 dB */ -static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const unsigned int hp_out_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54 dB to +15 dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0), +}; + static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); -- 1.7.1

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[alsa-devel] [PATCH 3/9] ASoC: da7210: Add support for ADC & DAC equalizers
by Ashish Chavan 13 Oct '11

13 Oct '11

This patch adds support for ADC and DAC five band equalizers available on DA7210 codec. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 36 ++++++++++++++++++++++++++++++++++++ 1 files changed, 36 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 6039ada..b48a8af 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -38,10 +38,16 @@ #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F #define DA7210_ADC 0x10 +#define DA7210_ADC_EQ1_2 0X11 +#define DA7210_ADC_EQ3_4 0x12 +#define DA7210_ADC_EQ5 0x13 #define DA7210_DAC_HPF 0x14 #define DA7210_DAC_L 0x15 #define DA7210_DAC_R 0x16 #define DA7210_DAC_SEL 0x17 +#define DA7210_DAC_EQ1_2 0x19 +#define DA7210_DAC_EQ3_4 0x1A +#define DA7210_DAC_EQ5 0x1B #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D #define DA7210_OUT1_L 0x1E @@ -190,6 +196,8 @@ static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); +static const DECLARE_TLV_DB_SCALE(eq_gain_tlv, -1050, 150, 0); +static const DECLARE_TLV_DB_SCALE(adc_eq_master_gain_tlv, -1800, 600, 1); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -248,6 +256,34 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, inpga_gain_tlv), + + /* DAC Equalizer controls */ + SOC_SINGLE("DAC EQ Switch", DA7210_DAC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("DAC EQ1 Volume", DA7210_DAC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ2 Volume", DA7210_DAC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ3 Volume", DA7210_DAC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ4 Volume", DA7210_DAC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("DAC EQ5 Volume", DA7210_DAC_EQ5, 0, 0xf, 1, + eq_gain_tlv), + + /* ADC Equalizer controls */ + SOC_SINGLE("ADC EQ Switch", DA7210_ADC_EQ5, 7, 1, 0), + SOC_SINGLE_TLV("ADC EQ Master Volume", DA7210_ADC_EQ5, 4, 0x3, + 1, adc_eq_master_gain_tlv), + SOC_SINGLE_TLV("ADC EQ1 Volume", DA7210_ADC_EQ1_2, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ2 Volume", DA7210_ADC_EQ1_2, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ3 Volume", DA7210_ADC_EQ3_4, 0, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ4 Volume", DA7210_ADC_EQ3_4, 4, 0xf, 1, + eq_gain_tlv), + SOC_SINGLE_TLV("ADC EQ5 Volume", DA7210_ADC_EQ5, 0, 0xf, 1, + eq_gain_tlv), }; /* Codec private data */ -- 1.7.1

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[alsa-devel] [PATCH 2/9] ASoC: da7210: Add support for line input and mic
by Ashish Chavan 13 Oct '11

13 Oct '11

DA7210 has three line inputs (AUX1 Left, AUX1 Right and AUX2) and a stereo MIC. This patch adds gain controls for MIC, AUX1, AUX2 as well as INPGA. It also adds a control to set MIC BIAS voltage. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 79 +++++++++++++++++++++++++++++++++++++++----- 1 files changed, 70 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index eef861d..6039ada 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -30,6 +30,10 @@ #define DA7210_STARTUP1 0x03 #define DA7210_MIC_L 0x07 #define DA7210_MIC_R 0x08 +#define DA7210_AUX1_L 0x09 +#define DA7210_AUX1_R 0x0A +#define DA7210_AUX2 0x0B +#define DA7210_IN_GAIN 0x0C #define DA7210_INMIX_L 0x0D #define DA7210_INMIX_R 0x0E #define DA7210_ADC_HPF 0x0F @@ -147,10 +151,23 @@ #define DA7210_OUT2_OUTMIX_L (1 << 6) #define DA7210_OUT2_EN (1 << 7) +/* AUX1_L bit fields */ +#define DA7210_AUX1_L_EN (1 << 7) + +/* AUX1_R bit fields */ +#define DA7210_AUX1_R_EN (1 << 7) + +/* AUX2 bit fields */ +#define DA7210_AUX2_EN (1 << 3) + /* Default gain/vol values */ #define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ #define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ #define DA7210_DFLT_OUT2_VOL 0x06 /* 0dB */ +#define DA7210_DFLT_MIC_VOL 0x02 /* 6dB */ +#define DA7210_DFLT_AUX1_VOL 0x35 /* 6dB */ +#define DA7210_DFLT_AUX2_VOL 0x02 /* 6dB */ +#define DA7210_DFLT_INPGA_VOL 0x07 /* 6dB */ #define DA7210_VERSION "0.0.1" @@ -170,6 +187,9 @@ */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(aux2_vol_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(inpga_gain_tlv, -450, 150, 0); static const unsigned int lineout_vol_tlv[] = { TLV_DB_RANGE_HEAD(2), @@ -185,6 +205,20 @@ static const unsigned int mono_vol_tlv[] = { 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) }; +static const unsigned int aux1_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -48dB to 21dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-4800, 150, 0) +}; + +static const char *da7210_mic_bias_voltage_txt[] = { + "1.5", "1.6", "2.2", "2.3" +}; + +static const struct soc_enum da7210_mic_bias = + SOC_ENUM_SINGLE(DA7210_MIC_L, 4, 4, da7210_mic_bias_voltage_txt); + static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", @@ -198,6 +232,22 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { 0, 0x3f, 0, lineout_vol_tlv), SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, mono_vol_tlv), + + /* MIC related controls */ + SOC_DOUBLE_R_TLV("Mic Capture Volume", + DA7210_MIC_L, DA7210_MIC_R, + 0, 0x5, 0, mic_vol_tlv), + SOC_ENUM("Mic Bias Voltage", da7210_mic_bias), + + /* AUX related controls */ + SOC_DOUBLE_R_TLV("Aux1 Capture Volume", + DA7210_AUX1_L, DA7210_AUX1_R, + 0, 0x3f, 0, aux1_vol_tlv), + SOC_SINGLE_TLV("Aux2 Capture Volume", DA7210_AUX2, 0, 0x3, 0, + aux2_vol_tlv), + + SOC_DOUBLE_TLV("In PGA Capture Volume", DA7210_IN_GAIN, 0, 4, 0xF, 0, + inpga_gain_tlv), }; /* Codec private data */ @@ -251,13 +301,9 @@ static int da7210_startup(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, DA7210_OUTMIX_R, 0x1F, 0x10); } else { - /* Volume 7 */ - snd_soc_update_bits(codec, DA7210_MIC_L, 0x7, 0x7); - snd_soc_update_bits(codec, DA7210_MIC_R, 0x7, 0x7); - - /* Enable Mic */ - snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0x1); - snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0x1); + /* Enable Mic,AUX1 and AUX2 */ + snd_soc_update_bits(codec, DA7210_INMIX_L, 0x1F, 0xD); + snd_soc_update_bits(codec, DA7210_INMIX_R, 0x1F, 0xD); } return 0; @@ -492,8 +538,10 @@ static int da7210_probe(struct snd_soc_codec *codec) */ /* Enable Left & Right MIC PGA and Mic Bias */ - snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); - snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN | + DA7210_DFLT_MIC_VOL); + snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN | + DA7210_DFLT_MIC_VOL); /* Enable Left and Right input PGA */ snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); @@ -533,6 +581,19 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | DA7210_OUT2_OUTMIX_R); + /* Enable Aux1 and set default vol */ + snd_soc_write(codec, DA7210_AUX1_L, DA7210_DFLT_AUX1_VOL | + DA7210_AUX1_L_EN); + snd_soc_write(codec, DA7210_AUX1_R, DA7210_DFLT_AUX1_VOL | + DA7210_AUX1_R_EN); + /* Enable Aux2 and set default vol */ + snd_soc_write(codec, DA7210_AUX2, DA7210_DFLT_AUX2_VOL | + DA7210_AUX2_EN); + + /* Set default In PGA gain */ + snd_soc_write(codec, DA7210_IN_GAIN, DA7210_DFLT_INPGA_VOL | + (DA7210_DFLT_INPGA_VOL << 4)); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] [PATCH 1/9] ASoC: da7210: Add support for line out and DAC
by Ashish Chavan 13 Oct '11

13 Oct '11

DA7210 has three line outputs. OUT1 Left, OUT1 Right and OUT2 (mono). This patch adds support for gain controls for these three line outs. It also adds support for overall DAC gain control. Tested on Samsung SMDK6410 board with DA7210 evaluation board. Signed-off-by: Ashish Chavan <ashish.chavan(a)kpitcummins.com> Signed-off-by: David Dajun Chen <dchen(a)diasemi.com> --- sound/soc/codecs/da7210.c | 55 +++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 55 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index a9d9d39..eef861d 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -40,6 +40,9 @@ #define DA7210_DAC_SEL 0x17 #define DA7210_OUTMIX_L 0x1C #define DA7210_OUTMIX_R 0x1D +#define DA7210_OUT1_L 0x1E +#define DA7210_OUT1_R 0x1F +#define DA7210_OUT2 0x20 #define DA7210_HP_L_VOL 0x21 #define DA7210_HP_R_VOL 0x22 #define DA7210_HP_CFG 0x23 @@ -133,6 +136,22 @@ #define DA7210_PLL_FS_96000 (0xF << 0) #define DA7210_PLL_EN (0x1 << 7) +/* OUT1_L bit fields */ +#define DA7210_OUT1_L_EN (1 << 7) + +/* OUT1_R bit fields */ +#define DA7210_OUT1_R_EN (1 << 7) + +/* OUT2 bit fields */ +#define DA7210_OUT2_OUTMIX_R (1 << 5) +#define DA7210_OUT2_OUTMIX_L (1 << 6) +#define DA7210_OUT2_EN (1 << 7) + +/* Default gain/vol values */ +#define DA7210_DFLT_DAC_GAIN 0x10 /* 0dB */ +#define DA7210_DFLT_OUT1_VOL 0x35 /* 0dB */ +#define DA7210_DFLT_OUT2_VOL 0x06 /* 0dB */ + #define DA7210_VERSION "0.0.1" /* @@ -150,12 +169,35 @@ * -> min = -79.5 dB */ static const DECLARE_TLV_DB_SCALE(hp_out_tlv, -7950, 150, 1); +static const DECLARE_TLV_DB_SCALE(dac_gain_tlv, -7725, 75, 0); + +static const unsigned int lineout_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x10, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + /* -54dB to 15dB */ + 0x11, 0x3f, TLV_DB_SCALE_ITEM(-5400, 150, 0) +}; + +static const unsigned int mono_vol_tlv[] = { + TLV_DB_RANGE_HEAD(2), + 0x0, 0x2, TLV_DB_SCALE_ITEM(-1800, 0, 1), + /* -18dB to 6dB */ + 0x3, 0x7, TLV_DB_SCALE_ITEM(-1800, 600, 0) +}; static const struct snd_kcontrol_new da7210_snd_controls[] = { SOC_DOUBLE_R_TLV("HeadPhone Playback Volume", DA7210_HP_L_VOL, DA7210_HP_R_VOL, 0, 0x3F, 0, hp_out_tlv), + SOC_DOUBLE_R_TLV("DAC Playback Volume", + DA7210_DAC_L, DA7210_DAC_R, + 0, 0x77, 1, dac_gain_tlv), + SOC_DOUBLE_R_TLV("Lineout Playback Volume", + DA7210_OUT1_L, DA7210_OUT1_R, + 0, 0x3f, 0, lineout_vol_tlv), + SOC_SINGLE_TLV("Mono Playback Volume", DA7210_OUT2, 0, 0x7, 0, + mono_vol_tlv), }; /* Codec private data */ @@ -478,6 +520,19 @@ static int da7210_probe(struct snd_soc_codec *codec) DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); + /* Set default DAC gain */ + snd_soc_write(codec, DA7210_DAC_L, DA7210_DFLT_DAC_GAIN); + snd_soc_write(codec, DA7210_DAC_R, DA7210_DFLT_DAC_GAIN); + + /* Enable Line out amplifiers and set default vol */ + snd_soc_write(codec, DA7210_OUT1_L, DA7210_DFLT_OUT1_VOL | + DA7210_OUT1_L_EN); + snd_soc_write(codec, DA7210_OUT1_R, DA7210_DFLT_OUT1_VOL | + DA7210_OUT1_R_EN); + snd_soc_write(codec, DA7210_OUT2, DA7210_DFLT_OUT2_VOL | + DA7210_OUT2_EN | DA7210_OUT2_OUTMIX_L | + DA7210_OUT2_OUTMIX_R); + /* Diable PLL and bypass it */ snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); -- 1.7.1

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[alsa-devel] [smatch stuff] reading beyond the end of the array in build_feature_ctl()
by Dan Carpenter 13 Oct '11

13 Oct '11

Hi Daniel, Smatch complains about a potential read past the end of the array in build_feature_ctl(). sound/usb/mixer.c +1035 build_feature_ctl(34) error: buffer overflow 'audio_feature_info' 13 <= 14 This is in the init code so probably it's correct, and I've just missed how it works. Can you take a look? The problem is that audio_feature_info[] has 13 elements, but Smatch thinks control - 1 can go up to 14 so we're two elements past the end. 1035 cval->val_type = audio_feature_info[control-1].type; Smatch get the value for "control" from "i" in parse_audio_feature_unit(). sound/usb/mixer.c 1261 } else { /* UAC_VERSION_2 */ 1262 for (i = 0; i < 30/2; i++) { 1263 unsigned int ch_bits = 0; 1264 unsigned int ch_read_only = 0; 1265 1266 for (j = 0; j < channels; j++) { 1267 unsigned int mask = snd_usb_combine_bytes(bmaControls + csize * (j+1), csize); 1268 if (uac2_control_is_readable(mask, i)) { 1269 ch_bits |= (1 << j); 1270 if (!uac2_control_is_writeable(mask, i)) 1271 ch_read_only |= (1 << j); 1272 } 1273 } 1274 1275 /* NOTE: build_feature_ctl() will mark the control read-only if all channels 1276 * are marked read-only in the descriptors. Otherwise, the control will be 1277 * reported as writeable, but the driver will not actually issue a write 1278 * command for read-only channels */ 1279 if (ch_bits & 1) /* the first channel must be set (for ease of programming) */ 1280 build_feature_ctl(state, _ftr, ch_bits, i, &iterm, unitid, ch_read_only); ^ "i" can be 13 here. 1281 if (uac2_control_is_readable(master_bits, i)) 1282 build_feature_ctl(state, _ftr, 0, i, &iterm, unitid, ^ "i" maybe can go up to 14 here, I'm not sure. 1283 !uac2_control_is_writeable(master_bits, i)); 1284 } 1285 } regards, dan carpenter

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[alsa-devel] [PATCH] ALSA: HDA: Fixup Realtek headphone pin initialization
by David Henningsson 13 Oct '11

13 Oct '11

This typo caused headphone pins not to be initialized correctly. BugLink: https://bugs.launchpad.net/bugs/871582 Reported-by: Effenberg Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bf53663..6a4128d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3263,7 +3263,7 @@ static void alc_auto_init_extra_out(struct hda_codec *codec) int i; hda_nid_t pin, dac; - for (i = 0; i < spec->autocfg.speaker_outs; i++) { + for (i = 0; i < spec->autocfg.hp_outs; i++) { pin = spec->autocfg.hp_pins[i]; if (!pin) break; -- 1.7.5.4

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Re: [alsa-devel] [PATCH] ASoC: da7210: convert to soc-cache
by Ashish Chavan 13 Oct '11

13 Oct '11

Hi Axel, Today I was about to submit a patch series for da7210 codec driver and noticed your patch for the same driver. My patch series is mainly feature update kind of nature. It has total nine patches that add support for various things, starting from basic IOs to DAPM. I am thinking of going forward with my posting which is based on current version of da7210 mainline driver. In am ready to take the responsibility of updating(to base it on my patch series) and testing your patch on the hardware that we have. Other option is to wait till your changes gets approved and then base my patch series on updated driver with your changes. But as this may take some time and I definitely want to get more eyes on the patches early, this would be least preferred. Thanks, -- Ashish (GNU FAN) > Date: Wed, 12 Oct 2011 07:16:25 +0800 > From: Axel Lin <axel.lin(a)gmail.com> > Subject: [alsa-devel] [PATCH] ASoC: da7210: convert to soc-cache > To: linux-kernel(a)vger.kernel.org > Cc: David Chen <Dajun.chen(a)diasemi.com>, alsa-devel(a)alsa-project.org, > Mark Brown <broonie(a)opensource.wolfsonmicro.com>, Liam Girdwood > <lrg(a)ti.com>, Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> > Message-ID: <1318374985.6115.10.camel@phoenix> > Content-Type: text/plain; charset="UTF-8" > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > Hi Kuninori, > This patch converts da7210 to use soc-cache APIs. > I don't have this hardware, I'd appreciate if you can test this patch. > > Thanks, > Axel > sound/soc/codecs/da7210.c | 123 ++++++++++++++++---------------------------- > 1 files changed, 45 insertions(+), 78 deletions(-) > > diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c > index 33b4823..1f30ef9 100644 > --- a/sound/soc/codecs/da7210.c > +++ b/sound/soc/codecs/da7210.c > @@ -162,7 +162,6 @@ static const struct snd_kcontrol_new da7210_snd_controls[] = { > /* Codec private data */ > struct da7210_priv { > enum snd_soc_control_type control_type; > - void *control_data; > }; > > /* > @@ -189,50 +188,16 @@ static const u8 da7210_reg[] = { > 0x00, /* R88 */ > }; > > -/* > - * Read da7210 register cache > - */ > -static inline u32 da7210_read_reg_cache(struct snd_soc_codec *codec, u32 reg) > -{ > - u8 *cache = codec->reg_cache; > - BUG_ON(reg >= ARRAY_SIZE(da7210_reg)); > - return cache[reg]; > -} > - > -/* > - * Write to the da7210 register space > - */ > -static int da7210_write(struct snd_soc_codec *codec, u32 reg, u32 value) > -{ > - u8 *cache = codec->reg_cache; > - u8 data[2]; > - > - BUG_ON(codec->driver->volatile_register); > - > - data[0] = reg & 0xff; > - data[1] = value & 0xff; > - > - if (reg >= codec->driver->reg_cache_size) > - return -EIO; > - > - if (2 != codec->hw_write(codec->control_data, data, 2)) > - return -EIO; > - > - cache[reg] = value; > - return 0; > -} > - > -/* > - * Read from the da7210 register space. > - */ > -static inline u32 da7210_read(struct snd_soc_codec *codec, u32 reg) > +static int da7210_volatile_register(struct snd_soc_codec *codec, > + unsigned int reg) > { > - if (DA7210_STATUS == reg) > - return i2c_smbus_read_byte_data(codec->control_data, reg); > - > - return da7210_read_reg_cache(codec, reg); > + switch (reg) { > + case DA7210_STATUS: > + return 1; > + default: > + return 0; > + } > } > - > static int da7210_startup(struct snd_pcm_substream *substream, > struct snd_soc_dai *dai) > { > @@ -271,13 +236,13 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, > u32 fs, bypass; > > /* set DAI source to Left and Right ADC */ > - da7210_write(codec, DA7210_DAI_SRC_SEL, > + snd_soc_write(codec, DA7210_DAI_SRC_SEL, > DA7210_DAI_OUT_R_SRC | DA7210_DAI_OUT_L_SRC); > > /* Enable DAI */ > - da7210_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); > + snd_soc_write(codec, DA7210_DAI_CFG3, DA7210_DAI_OE | DA7210_DAI_EN); > > - dai_cfg1 = 0xFC & da7210_read(codec, DA7210_DAI_CFG1); > + dai_cfg1 = 0xFC & snd_soc_read(codec, DA7210_DAI_CFG1); > > switch (params_format(params)) { > case SNDRV_PCM_FORMAT_S16_LE: > @@ -290,7 +255,7 @@ static int da7210_hw_params(struct snd_pcm_substream *substream, > return -EINVAL; > } > > - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); > + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); > > hpf_reg = (SNDRV_PCM_STREAM_PLAYBACK == substream->stream) ? > DA7210_DAC_HPF : DA7210_ADC_HPF; > @@ -383,8 +348,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) > u32 dai_cfg1; > u32 dai_cfg3; > > - dai_cfg1 = 0x7f & da7210_read(codec, DA7210_DAI_CFG1); > - dai_cfg3 = 0xfc & da7210_read(codec, DA7210_DAI_CFG3); > + dai_cfg1 = 0x7f & snd_soc_read(codec, DA7210_DAI_CFG1); > + dai_cfg3 = 0xfc & snd_soc_read(codec, DA7210_DAI_CFG3); > > switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { > case SND_SOC_DAIFMT_CBM_CFM: > @@ -412,8 +377,8 @@ static int da7210_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) > */ > dai_cfg1 |= DA7210_DAI_FLEN_64BIT; > > - da7210_write(codec, DA7210_DAI_CFG1, dai_cfg1); > - da7210_write(codec, DA7210_DAI_CFG3, dai_cfg3); > + snd_soc_write(codec, DA7210_DAI_CFG1, dai_cfg1); > + snd_soc_write(codec, DA7210_DAI_CFG3, dai_cfg3); > > return 0; > } > @@ -452,11 +417,15 @@ static struct snd_soc_dai_driver da7210_dai = { > static int da7210_probe(struct snd_soc_codec *codec) > { > struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); > + int ret; > > dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); > > - codec->control_data = da7210->control_data; > - codec->hw_write = (hw_write_t)i2c_master_send; > + ret = snd_soc_codec_set_cache_io(codec, 8, 8, da7210->control_type); > + if (ret < 0) { > + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); > + return ret; > + } > > /* FIXME > * > @@ -473,8 +442,8 @@ static int da7210_probe(struct snd_soc_codec *codec) > /* > * make sure that DA7210 use bypass mode before start up > */ > - da7210_write(codec, DA7210_STARTUP1, 0); > - da7210_write(codec, DA7210_PLL_DIV3, > + snd_soc_write(codec, DA7210_STARTUP1, 0); > + snd_soc_write(codec, DA7210_PLL_DIV3, > DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); > > /* > @@ -482,36 +451,36 @@ static int da7210_probe(struct snd_soc_codec *codec) > */ > > /* Enable Left & Right MIC PGA and Mic Bias */ > - da7210_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); > - da7210_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); > + snd_soc_write(codec, DA7210_MIC_L, DA7210_MIC_L_EN | DA7210_MICBIAS_EN); > + snd_soc_write(codec, DA7210_MIC_R, DA7210_MIC_R_EN); > > /* Enable Left and Right input PGA */ > - da7210_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); > - da7210_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); > + snd_soc_write(codec, DA7210_INMIX_L, DA7210_IN_L_EN); > + snd_soc_write(codec, DA7210_INMIX_R, DA7210_IN_R_EN); > > /* Enable Left and Right ADC */ > - da7210_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); > + snd_soc_write(codec, DA7210_ADC, DA7210_ADC_L_EN | DA7210_ADC_R_EN); > > /* > * DAC settings > */ > > /* Enable Left and Right DAC */ > - da7210_write(codec, DA7210_DAC_SEL, > + snd_soc_write(codec, DA7210_DAC_SEL, > DA7210_DAC_L_SRC_DAI_L | DA7210_DAC_L_EN | > DA7210_DAC_R_SRC_DAI_R | DA7210_DAC_R_EN); > > /* Enable Left and Right out PGA */ > - da7210_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); > - da7210_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); > + snd_soc_write(codec, DA7210_OUTMIX_L, DA7210_OUT_L_EN); > + snd_soc_write(codec, DA7210_OUTMIX_R, DA7210_OUT_R_EN); > > /* Enable Left and Right HeadPhone PGA */ > - da7210_write(codec, DA7210_HP_CFG, > + snd_soc_write(codec, DA7210_HP_CFG, > DA7210_HP_2CAP_MODE | DA7210_HP_SENSE_EN | > DA7210_HP_L_EN | DA7210_HP_MODE | DA7210_HP_R_EN); > > /* Diable PLL and bypass it */ > - da7210_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); > + snd_soc_write(codec, DA7210_PLL, DA7210_PLL_FS_48000); > > /* > * If 48kHz sound came, it use bypass mode, > @@ -522,22 +491,22 @@ static int da7210_probe(struct snd_soc_codec *codec) > * DA7210_PLL_DIV3 :: DA7210_PLL_BYP bit. > * see da7210_hw_params > */ > - da7210_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ > - da7210_write(codec, DA7210_PLL_DIV2, 0x99); > - da7210_write(codec, DA7210_PLL_DIV3, 0x0A | > + snd_soc_write(codec, DA7210_PLL_DIV1, 0xE5); /* MCLK = 12.288MHz */ > + snd_soc_write(codec, DA7210_PLL_DIV2, 0x99); > + snd_soc_write(codec, DA7210_PLL_DIV3, 0x0A | > DA7210_MCLK_RANGE_10_20_MHZ | DA7210_PLL_BYP); > snd_soc_update_bits(codec, DA7210_PLL, DA7210_PLL_EN, DA7210_PLL_EN); > > /* As suggested by Dialog */ > - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ > - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); > - da7210_write(codec, DA7210_A_PLL1, 0x01); > - da7210_write(codec, DA7210_A_CP_MODE, 0x7C); > - da7210_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ > - da7210_write(codec, DA7210_A_TEST_UNLOCK, 0x00); > + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x8B); /* unlock */ > + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0xB4); > + snd_soc_write(codec, DA7210_A_PLL1, 0x01); > + snd_soc_write(codec, DA7210_A_CP_MODE, 0x7C); > + snd_soc_write(codec, DA7210_A_HID_UNLOCK, 0x00); /* re-lock */ > + snd_soc_write(codec, DA7210_A_TEST_UNLOCK, 0x00); > > /* Activate all enabled subsystem */ > - da7210_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); > + snd_soc_write(codec, DA7210_STARTUP1, DA7210_SC_MST_EN); > > snd_soc_add_controls(codec, da7210_snd_controls, > ARRAY_SIZE(da7210_snd_controls)); > @@ -549,11 +518,10 @@ static int da7210_probe(struct snd_soc_codec *codec) > > static struct snd_soc_codec_driver soc_codec_dev_da7210 = { > .probe = da7210_probe, > - .read = da7210_read, > - .write = da7210_write, > .reg_cache_size = ARRAY_SIZE(da7210_reg), > .reg_word_size = sizeof(u8), > .reg_cache_default = da7210_reg, > + .volatile_register = da7210_volatile_register, > }; > > #if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) > @@ -568,7 +536,6 @@ static int __devinit da7210_i2c_probe(struct i2c_client *i2c, > return -ENOMEM; > > i2c_set_clientdata(i2c, da7210); > - da7210->control_data = i2c; > da7210->control_type = SND_SOC_I2C; > > ret = snd_soc_register_codec(&i2c->dev, > -- > 1.7.4.1 >

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[alsa-devel] [PATCH] [RESEND] Aplay: Fix spelling error
by David Henningsson 12 Oct '11

12 Oct '11

BugLink: https://bugs.launchpad.net/bugs/816388 Reported-by: Chad Dunlap <cldunlap1(a)gmail.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Resending patch to more maintainers, as nobody seem to have picked it up the previous time. aplay/aplay.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 07e6bc0..5ab63a9 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -209,7 +209,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --disable-format disable automatic format conversions\n" " --disable-softvol disable software volume control (softvol)\n" " --test-position test ring buffer position\n" -" --test-coef=# test coeficient for ring buffer position (default 8)\n" +" --test-coef=# test coefficient for ring buffer position (default 8)\n" " expression for validation is: coef * (buffer_size / 2)\n" " --test-nowait do not wait for ring buffer - eats whole CPU\n" " --max-file-time=# start another output file when the old file has recorded\n" -- 1.7.5.4

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[alsa-devel] SOC: using a codec (ADAU1761) just as an analog mixer
by Matteo Fortini 12 Oct '11

12 Oct '11

I'm using an ADAU1761 codec in a project, but we'd like, for now, to use it just as a mixer, i.e. without passing through the ADC/DAC and DSP stages. Which is the best way of accomplishing that? In the past I tried to add virtual DAPM routes in order to get the relevant power systems powered up, and bypassing the standby modes, since the device wouldn't let me use the mixer controls through I2C if I don't clock it. Another question: how do I tell ALSA to power up a stage (in this case, the ALC/differential input stage)? TIA, Matteo

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[alsa-devel] [PATCH] ASoC: Make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X
by Axel Lin 12 Oct '11

12 Oct '11

In saarb_pm860x_init() and evb3_pm860x_init(), we call pm860x_hs_jack_detect() and pm860x_mic_jack_detect() which in turn calls pm860x_set_bits(). Thus make SND_SOC_SAARB and SND_SOC_TAVOREVB3 select MFD_88PM860X. This patch fixes below build error if CONFIG_MFD_88PM860X is not configured. LD .tmp_vmlinux1 sound/built-in.o: In function `pm860x_write_reg_cache': last.c:(.text+0x29e9c): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_set_bias_level': last.c:(.text+0x29ecc): undefined reference to `pm860x_set_bits' last.c:(.text+0x29f00): undefined reference to `pm860x_reg_write' last.c:(.text+0x29f18): undefined reference to `pm860x_reg_write' sound/built-in.o: In function `pm860x_read_reg_cache': last.c:(.text+0x29f40): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_probe': last.c:(.text+0x2a034): undefined reference to `pm860x_bulk_read' sound/built-in.o: In function `pm860x_codec_handler': last.c:(.text+0x2a344): undefined reference to `pm860x_reg_read' last.c:(.text+0x2a354): undefined reference to `pm860x_reg_read' sound/built-in.o: In function `pm860x_mic_jack_detect': last.c:(.text+0x2a450): undefined reference to `pm860x_set_bits' sound/built-in.o: In function `pm860x_hs_jack_detect': last.c:(.text+0x2a4d0): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a4f8): undefined reference to `pm860x_set_bits' last.c:(.text+0x2a510): undefined reference to `pm860x_set_bits' make: *** [.tmp_vmlinux1] Error 1 Signed-off-by: Axel Lin <axel.lin(a)gmail.com> --- sound/soc/pxa/Kconfig | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 33ebc46b..ffd2242 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -121,6 +121,7 @@ config SND_PXA2XX_SOC_PALM27X config SND_SOC_SAARB tristate "SoC Audio support for Marvell Saarb" depends on SND_PXA2XX_SOC && MACH_SAARB + select MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help @@ -130,6 +131,7 @@ config SND_SOC_SAARB config SND_SOC_TAVOREVB3 tristate "SoC Audio support for Marvell Tavor EVB3" depends on SND_PXA2XX_SOC && MACH_TAVOREVB3 + select MFD_88PM860X select SND_PXA_SOC_SSP select SND_SOC_88PM860X help -- 1.7.4.1

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