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[alsa-devel] [PATCH] ASoC: core - make IO calls return a valid errno on failure.
by Liam Girdwood 26 Jul '11

26 Jul '11

IO call failures should return an errno instead of -1 Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 4 ++-- sound/soc/soc-dapm.c | 4 ++-- sound/soc/soc-io.c | 6 +++--- 3 files changed, 7 insertions(+), 7 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 159f144..fcf69f4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1661,7 +1661,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, if (!platform->driver->read) { dev_err(platform->dev, "platform has no read back\n"); - return -1; + return -EINVAL; } ret = platform->driver->read(platform, reg); @@ -1677,7 +1677,7 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, { if (!platform->driver->write) { dev_err(platform->dev, "platform has no write back\n"); - return -1; + return -EINVAL; } dev_dbg(platform->dev, "write %x = %x\n", reg, val); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3956670..45141fa 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -132,7 +132,7 @@ static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) return snd_soc_platform_read(w->platform, reg); dev_err(w->dapm->dev, "no valid widget read method\n"); - return -1; + return -EINVAL; } static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) @@ -143,7 +143,7 @@ static int soc_widget_write(struct snd_soc_dapm_widget *w, int reg, int val) return snd_soc_platform_write(w->platform, reg, val); dev_err(w->dapm->dev, "no valid widget write method\n"); - return -1; + return -EINVAL; } static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index cca490c..166186c 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -41,7 +41,7 @@ static int do_hw_write(struct snd_soc_codec *codec, unsigned int reg, !codec->cache_bypass) { ret = snd_soc_cache_write(codec, reg, value); if (ret < 0) - return -1; + return ret; } if (codec->cache_only) { @@ -67,7 +67,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) snd_soc_codec_volatile_register(codec, reg) || codec->cache_bypass) { if (codec->cache_only) - return -1; + return -EINVAL; BUG_ON(!codec->hw_read); return codec->hw_read(codec, reg); @@ -75,7 +75,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) ret = snd_soc_cache_read(codec, reg, &val); if (ret < 0) - return -1; + return ret; return val; } -- 1.7.4.1

2 3

[alsa-devel] [PATCH] Cirrus Logic CS421x support
by Tim Howe 26 Jul '11

26 Jul '11

This update includes the changes necessary for supporting the CS421x family of codecs. Previously this file only supported the CS420x family of codecs. This file also contains init verbs to correct several issues in the CS421x hardware. Behavior between the CS421x and CS420x codec families is similar, so several functions have been reused with "if" statements to determine which codec family (CS421x or CS420x) is present. Also, this file will be updated sometime in the near future in order to add support for a system using CS421x that requires mono mix on the speaker output only. Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/pci/hda/patch_cirrus.c | 786 +++++++++++++++++++++++++++++++++++++++--- 1 files changed, 745 insertions(+), 41 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index 7f93739..27dbd5c 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -25,6 +25,7 @@ #include <sound/core.h> #include "hda_codec.h" #include "hda_local.h" +#include <sound/tlv.h> /* */ @@ -61,9 +62,15 @@ struct cs_spec { unsigned int hp_detect:1; unsigned int mic_detect:1; + /* CS421x */ + unsigned int spdif_detect:1; + unsigned int sense_b:1; + hda_nid_t vendor_nid; + struct hda_input_mux input_mux; + unsigned int last_input; }; -/* available models */ +/* available models with CS420x */ enum { CS420X_MBP53, CS420X_MBP55, @@ -72,6 +79,12 @@ enum { CS420X_MODELS }; +/* CS421x boards */ +enum { + CS421X_CDB4210, + CS421X_MODELS +}; + /* Vendor-specific processing widget */ #define CS420X_VENDOR_NID 0x11 #define CS_DIG_OUT1_PIN_NID 0x10 @@ -111,21 +124,42 @@ enum { /* 0x0009 - 0x0014 -> 12 test regs */ /* 0x0015 - visibility reg */ +/* + * Cirrus Logic CS4210 + * + * 1 DAC => HP(sense) / Speakers, + * 1 ADC <= LineIn(sense) / MicIn / DMicIn, + * 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ +#define CS4210_DAC_NID 0x02 +#define CS4210_ADC_NID 0x03 +#define CS421X_VENDOR_NID 0x0B +#define CS421X_DMIC_PIN_NID 0x09 /* Port E */ +#define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ + +#define CS421X_IDX_DEV_CFG 0x01 +#define CS421X_IDX_ADC_CFG 0x02 +#define CS421X_IDX_DAC_CFG 0x03 +#define CS421X_IDX_SPK_CTL 0x04 + +#define SPDIF_EVENT 0x04 static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + return snd_hda_codec_read(codec, spec->vendor_nid, 0, AC_VERB_GET_PROC_COEF, 0); } static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, unsigned int coef) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_PROC_COEF, coef); } @@ -343,18 +377,44 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, { int i; hda_nid_t nid; + struct cs_spec *spec = codec->spec; nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { unsigned int type; - int idx; - type = get_wcaps_type(get_wcaps(codec, nid)); - if (type != AC_WID_AUD_IN) - continue; - idx = snd_hda_get_conn_index(codec, nid, pin, 0); - if (idx >= 0) { - *idxp = idx; - return nid; + int j, nums; + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + hda_nid_t pins[HDA_MAX_NUM_INPUTS]; + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + if (type != AC_WID_AUD_IN) + continue; + nums = snd_hda_get_connections(codec, nid, pins, + ARRAY_SIZE(pins)); + if (nums <= 0) + continue; + for (j = 0; j < nums; j++) { + if (pins[j] == pin) { + *idxp = j; + return nid; + } + } + } else { + hda_nid_t pins[2]; + type = get_wcaps_type(get_wcaps(codec, nid)); + if (type != AC_WID_AUD_IN) + continue; + nums = snd_hda_get_connections(codec, nid, pins, + ARRAY_SIZE(pins)); + if (nums <= 0) + continue; + for (j = 0; j < nums; j++) { + if (pins[j] == pin) { + *idxp = j; + return nid; + } + } } } return 0; @@ -835,6 +895,8 @@ static int build_digital_input(struct hda_codec *codec) /* * auto-mute and auto-mic switching + * CS421x auto-output redirecting + * HP/SPK/SPDIF */ static void cs_automute(struct hda_codec *codec) @@ -842,9 +904,25 @@ static void cs_automute(struct hda_codec *codec) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int hp_present; + unsigned int spdif_present; hda_nid_t nid; int i; + spdif_present = 0; + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + if (is_jack_detectable(codec, nid)) { + /* + TODO: SPDIF output redirect when SENSE_B is enabled. + Shared (SENSE_A) jack (e.g HP/mini-TOSLINK) + assumed. + */ + if (snd_hda_jack_detect(codec, nid) + /* && spec->sense_b */) + spdif_present = 1; + } + } + hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; @@ -854,11 +932,19 @@ static void cs_automute(struct hda_codec *codec) if (hp_present) break; } + + /* mute speakers if spdif or hp jack is plugged in */ for (i = 0; i < cfg->speaker_outs; i++) { nid = cfg->speaker_pins[i]; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, hp_present ? 0 : PIN_OUT); + /* detect on spdif is specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? 0 : PIN_OUT); + } } if (spec->board_config == CS420X_MBP53 || spec->board_config == CS420X_MBP55 || @@ -867,21 +953,62 @@ static void cs_automute(struct hda_codec *codec) snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, gpio); } + + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + /* mute HPs if spdif jack (SENSE_B) is present */ + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (spdif_present && spec->sense_b) ? 0 : PIN_HP); + } + + /* SPDIF TX on/off */ + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? PIN_OUT : 0); + + } + /* Update board GPIOs if neccessary ... */ + } } +/* + * Auto-input redirect for CS421x + * Switch max 3 inputs of a single ADC (nid 3) +*/ + static void cs_automic(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t nid; unsigned int present; - + nid = cfg->inputs[spec->automic_idx].pin; present = snd_hda_jack_detect(codec, nid); - if (present) - change_cur_input(codec, spec->automic_idx, 0); - else - change_cur_input(codec, !spec->automic_idx, 0); + + /* specific to CS421x, single ADC */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (present) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } else { + spec->cur_input = spec->last_input; + } + + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } else { + if (present) + change_cur_input(codec, spec->automic_idx, 0); + else + change_cur_input(codec, !spec->automic_idx, 0); + } } /* @@ -911,23 +1038,28 @@ static void init_output(struct hda_codec *codec) for (i = 0; i < cfg->line_outs; i++) snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* HP */ for (i = 0; i < cfg->hp_outs; i++) { hda_nid_t nid = cfg->hp_pins[i]; snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (is_jack_detectable(codec, nid)) { + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); spec->hp_detect = 1; } } + + /* Speaker */ for (i = 0; i < cfg->speaker_outs; i++) snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); - if (spec->hp_detect) + + /* SPDIF is enabled on presence detect for CS421x */ + if (spec->hp_detect || spec->spdif_detect) cs_automute(codec); } @@ -961,19 +1093,31 @@ static void init_input(struct hda_codec *codec) AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | MIC_EVENT); } - change_cur_input(codec, spec->cur_input, 1); - if (spec->mic_detect) - cs_automic(codec); - - coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ - if (is_active_pin(codec, CS_DMIC2_PIN_NID)) - coef |= 0x0500; /* DMIC2 enable 2 channels, disable GPIO1 */ - if (is_active_pin(codec, CS_DMIC1_PIN_NID)) - coef |= 0x1800; /* DMIC1 enable 2 channels, disable GPIO0 - * No effect if SPDIF_OUT2 is selected in - * IDX_SPDIF_CTL. - */ - cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + /* specific to CS421x */ + if (spec->vendor_nid == CS421X_VENDOR_NID) { + if (spec->mic_detect) + cs_automic(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } + } else { + change_cur_input(codec, spec->cur_input, 1); + if (spec->mic_detect) + cs_automic(codec); + + coef = 0x000a; /* ADC1/2 - Digital and Analog Soft Ramp */ + if (is_active_pin(codec, CS_DMIC2_PIN_NID)) + coef |= 0x0500; /* DMIC2 2 chan on, GPIO1 off */ + if (is_active_pin(codec, CS_DMIC1_PIN_NID)) + coef |= 0x1800; /* DMIC1 2 chan on, GPIO0 off + * No effect if SPDIF_OUT2 is + * selected in IDX_SPDIF_CTL. + */ + cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); + } } static const struct hda_verb cs_coef_init_verbs[] = { @@ -1173,7 +1317,7 @@ struct cs_pincfg { u32 val; }; -static const struct cs_pincfg mbp53_pincfgs[] = { +static struct cs_pincfg mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1187,7 +1331,7 @@ static const struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg mbp55_pincfgs[] = { +static struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1201,7 +1345,7 @@ static const struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg imac27_pincfgs[] = { +static struct cs_pincfg imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1215,22 +1359,22 @@ static const struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { +static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, }; -static void fix_pincfg(struct hda_codec *codec, int model) +static void fix_pincfg(struct hda_codec *codec, int model, + struct cs_pincfg **pin_configs) { - const struct cs_pincfg *cfg = cs_pincfgs[model]; + const struct cs_pincfg *cfg = pin_configs[model]; if (!cfg) return; for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1241,11 +1385,13 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->vendor_nid = CS420X_VENDOR_NID; + spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config); + fix_pincfg(codec, spec->board_config, cs_pincfgs); switch (spec->board_config) { case CS420X_IMAC27: @@ -1272,6 +1418,562 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + * Cirrus Logic CS4210 + * + * 1 DAC => HP(sense) / Speakers, + * 1 ADC <= LineIn(sense) / MicIn / DMicIn, + * 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ + +/* CS4210 board names */ +static const char *cs421x_models[CS421X_MODELS] = { + [CS421X_CDB4210] = "cdb4210", +}; + +static struct snd_pci_quirk cs421x_cfg_tbl[] = { + /* Test Intel board + CDB2410 */ + SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), + {} /* terminator */ +}; + +/* CS4210 board pinconfigs */ +/* Default CS4210 (CDB4210)*/ +static struct cs_pincfg cdb4210_pincfgs[] = { + { 0x05, 0x0321401f }, + { 0x06, 0x90170010 }, + { 0x07, 0x03813031 }, + { 0x08, 0xb7a70037 }, + { 0x09, 0xb7a6003e }, + { 0x0a, 0x034510f0 }, + {} /* terminator */ +}; + +static struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { + [CS421X_CDB4210] = cdb4210_pincfgs, +}; + +static struct hda_verb cs421x_coef_init_verbs[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 1}, + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG}, + /* + Disable Coefficient Index Auto-Increment(DAI)=1, + PDREF=0 + */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0001 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG}, + /* ADC SZCMode = Digital Soft Ramp */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0002 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG}, + {0x0B, AC_VERB_SET_PROC_COEF, + (0x0002 /* DAC SZCMode = Digital Soft Ramp */ + | 0x0004 /* Mute DAC on FIFO error */ + | 0x0008 /* Enable DAC High Pass Filter */ + )}, + {} /* terminator */ +}; + +/* Errata: CS4210 rev A1 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ + * + * Description: + * 1. Performance degredation is present in the ADC. + * 2. Speaker output is not completely muted upon HP detect. + * 3. Noise is present when clipping occurs on the amplified + * speaker outputs. + * + * Workaround: + * The following verb sequence written to the registers during + * initialization will correct the issues listed above. + */ + +static struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0006}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x000A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0011}, + {0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001B}, + {0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */ + + {} /* terminator */ +}; + +/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */ +static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0); + +static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 3; + return 0; +} + +static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003; + return 0; +} + +static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + unsigned int vol = ucontrol->value.integer.value[0]; + unsigned int coef = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL); + unsigned int original_coef = coef; + + coef &= ~0x0003; + coef |= (vol & 0x0003); + if (original_coef == coef) + return 0; + else { + cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef); + return 1; + } +} + +static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = { + + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Speaker Boost Playback Volume", + .info = cs421x_boost_vol_info, + .get = cs421x_boost_vol_get, + .put = cs421x_boost_vol_put, + .tlv = { .p = cs421x_speaker_boost_db_scale }, +}; + +static void cs421x_pinmux_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + unsigned int def_conf, coef; + + /* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */ + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + + if (spec->gpio_mask) + coef |= 0x0008; /* B1,B2 are GPIOs */ + else + coef &= ~0x0008; + + if (spec->sense_b) + coef |= 0x0010; /* B2 is SENSE_B, not inverted */ + else + coef &= ~0x0010; + + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + if ((spec->gpio_mask || spec->sense_b) && + is_active_pin(codec, CS421X_DMIC_PIN_NID)) { + + /* + GPIO or SENSE_B forced - disconnect the DMIC pin. + */ + def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID); + def_conf &= ~AC_DEFCFG_PORT_CONN; + def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT); + snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf); + } +} + +static void init_cs421x_digital(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + + for (i = 0; i < cfg->dig_outs; i++) { + hda_nid_t nid = cfg->dig_out_pins[i]; + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | SPDIF_EVENT); + spec->spdif_detect = 1; + } + } +} + +static int cs421x_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + + cs421x_pinmux_init(codec); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_output(codec); + init_input(codec); + init_cs421x_digital(codec); + + return 0; +} + +/* + * CS4210 Input MUX (1 ADC) + */ +static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_input; + return 0; +} + +static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_nid[0], &spec->cur_input); + +} + +static struct snd_kcontrol_new cs421x_capture_source = { + + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs421x_mux_enum_info, + .get = cs421x_mux_enum_get, + .put = cs421x_mux_enum_put, +}; + +static int cs421x_add_input_volume_control(struct hda_codec *codec, int item) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + const struct hda_input_mux *imux = &spec->input_mux; + hda_nid_t pin = cfg->inputs[item].pin; + struct snd_kcontrol *kctl; + u32 caps; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + + return add_volume(codec, imux->items[item].label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + +/* add a (input-boost) volume control to the given input pin */ +static int build_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->input_mux; + int i, err, type_idx; + const char *label; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + int n; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); + if (err < 0) + return err; + } + } + + /* Add Input MUX Items + Capture Volume/Switch */ + for (i = 0; i < spec->num_inputs; i++) { + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx); + + err = cs421x_add_input_volume_control(codec, i); + if (err < 0) + return err; + } + + /* + Add 'Capture Source' Switch if + * 2 inputs and no mic detec + * 3 inputs + */ + if ((spec->num_inputs == 2 && !spec->mic_detect) || + (spec->num_inputs == 3)) { + + err = snd_hda_ctl_add(codec, spec->adc_nid[0], + snd_ctl_new1(&cs421x_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* Single DAC (Mute/Gain) */ +static int build_cs421x_output(struct hda_codec *codec) +{ + hda_nid_t dac = CS4210_DAC_NID; + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct snd_kcontrol *kctl; + int err; + char *name = "HP/Speakers"; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + err = add_mute(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + if (cfg->speaker_outs) { + err = snd_hda_ctl_add(codec, 0, + snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); + if (err < 0) + return err; + } + return err; +} + +static int cs421x_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_cs421x_output(codec); + if (err < 0) + return err; + err = build_cs421x_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + return cs421x_init(codec); +} + +static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x3f) { + case HP_EVENT: + case SPDIF_EVENT: + cs_automute(codec); + break; + + case MIC_EVENT: + cs_automic(codec); + break; + } +} + +static int parse_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + spec->cur_input = spec->last_input = i; + spec->num_inputs++; + + /* check whether the automatic mic switch is available */ + if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) { + spec->mic_detect = 1; + spec->automic_idx = i; + } + } + return 0; +} + +static int cs421x_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = parse_output(codec); + if (err < 0) + return err; + err = parse_cs421x_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +/* + Manage PDREF, when transitioning to D3hot + (DAC,ADC) -> D3, PDREF=1, AFG->D3 +*/ +static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +{ + unsigned int coef; + + snd_hda_shutup_pins(codec); + + snd_hda_codec_write(codec, CS4210_DAC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, CS4210_ADC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + return 0; +} +#endif + +static struct hda_codec_ops cs4210_patch_ops = { + .build_controls = cs421x_build_controls, + .build_pcms = cs_build_pcms, + .init = cs421x_init, + .free = cs_free, + .unsol_event = cs421x_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = cs421x_suspend, +#endif +}; + +static int patch_cs421x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS421X_VENDOR_NID; + + spec->board_config = + snd_hda_check_board_config(codec, CS421X_MODELS, + cs421x_models, cs421x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config, cs421x_pincfgs); + /* + Setup GPIO/SENSE for each board (if used) + */ + switch (spec->board_config) { + case CS421X_CDB4210: + snd_printd("CS4210 board: %s\n", + cs421x_models[spec->board_config]); +/* spec->gpio_mask = 3; + spec->gpio_dir = 3; + spec->gpio_data = 3; +*/ + spec->sense_b = 1; + + break; + } + + /* + Update the GPIO/DMIC/SENSE_B pinmux before the configuration + is auto-parsed. If GPIO or SENSE_B is forced, DMIC input + is disabled. + */ + cs421x_pinmux_init(codec); + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs4210_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + /* * patch entries @@ -1279,11 +1981,13 @@ static int patch_cs420x(struct hda_codec *codec) static const struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -- 1.7.0.4

2 2

[alsa-devel] snd_mask_min
by Tim Blechmann 26 Jul '11

26 Jul '11

hi all, i've got a problem using params_format(): it uses an inline function snd_mask_min, which is declared in sound/pcm_params.h, but this file is not included by sound/pcm.h, where it is defined. including it manually, causes a huge number of errors. it would be very helpful, if pcm.h would be self-contained. cheers, tim

2 3

[alsa-devel] [PATCH] alsa: lx6464es - initialization fixes
by Tim Blechmann 26 Jul '11

26 Jul '11

this using this fix, the driver is usable with non-rt kernels. the card initialization is moved from the prepare to the hw_params callback. also the error handling is improved. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 72 +++++++++++++++++++++-------------------- sound/pci/lx6464es/lx6464es.h | 1 + sound/pci/lx6464es/lx_core.c | 43 +++++++++++++++---------- sound/pci/lx6464es/lx_core.h | 3 +- 4 files changed, 66 insertions(+), 53 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 1bd7a54..722212e 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -103,16 +103,14 @@ static int lx_set_granularity(struct lx6464es *chip, u32 gran); static int lx_hardware_open(struct lx6464es *chip, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream, + int channels, snd_pcm_uframes_t period_size) { int err = 0; - struct snd_pcm_runtime *runtime = substream->runtime; - int channels = runtime->channels; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_pcm_uframes_t period_size = runtime->period_size; - - snd_printd(LXP "allocating pipe for %d channels\n", channels); + snd_printk(LXP "allocating pipe for %d channels, %lu period size\n", + channels, period_size); err = lx_pipe_allocate(chip, 0, is_capture, channels); if (err < 0) { snd_printk(KERN_ERR LXP "allocating pipe failed\n"); @@ -130,14 +128,14 @@ static int lx_hardware_open(struct lx6464es *chip, } static int lx_hardware_start(struct lx6464es *chip, - struct snd_pcm_substream *substream) + struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) { int err = 0; - struct snd_pcm_runtime *runtime = substream->runtime; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); snd_printd(LXP "setting stream format\n"); - err = lx_stream_set_format(chip, runtime, 0, is_capture); + err = lx_stream_set_format(chip, hw_params, 0, is_capture); if (err < 0) { snd_printk(KERN_ERR LXP "setting stream format failed\n"); return err; @@ -295,13 +293,33 @@ static snd_pcm_uframes_t lx_pcm_stream_pointer(struct snd_pcm_substream return pos; } + static int lx_pcm_prepare(struct snd_pcm_substream *substream) { struct lx6464es *chip = snd_pcm_substream_chip(substream); int err = 0; - const int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_printdd("->lx_pcm_prepare\n"); + snd_printd("->lx_pcm_prepare %d\n", is_capture); + + mutex_lock(&chip->setup_mutex); + + if (chip->board_sample_rate != substream->runtime->rate) { + if (!err) + chip->board_sample_rate = substream->runtime->rate; + } + + mutex_unlock(&chip->setup_mutex); + snd_printd("<-lx_pcm_prepare\n"); + return err; +} + +static int lx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params, int is_capture) +{ + struct lx6464es *chip = snd_pcm_substream_chip(substream); + int err = 0; + + snd_printd("->lx_pcm_hw_params\n"); mutex_lock(&chip->setup_mutex); @@ -322,14 +340,16 @@ static int lx_pcm_prepare(struct snd_pcm_substream *substream) } snd_printd(LXP "opening hardware\n"); - err = lx_hardware_open(chip, substream); + err = lx_hardware_open(chip, substream, + params_channels(hw_params), + params_period_size(hw_params)); if (err < 0) { snd_printk(KERN_ERR LXP "failed to open hardware. " "Error code %d\n", err); goto exit; } - err = lx_hardware_start(chip, substream); + err = lx_hardware_start(chip, substream, hw_params); if (err < 0) { snd_printk(KERN_ERR LXP "failed to start hardware. " "Error code %d\n", err); @@ -338,26 +358,6 @@ static int lx_pcm_prepare(struct snd_pcm_substream *substream) chip->hardware_running[is_capture] = 1; - if (chip->board_sample_rate != substream->runtime->rate) { - if (!err) - chip->board_sample_rate = substream->runtime->rate; - } - -exit: - mutex_unlock(&chip->setup_mutex); - return err; -} - -static int lx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params, int is_capture) -{ - struct lx6464es *chip = snd_pcm_substream_chip(substream); - int err = 0; - - snd_printdd("->lx_pcm_hw_params\n"); - - mutex_lock(&chip->setup_mutex); - /* set dma buffer */ err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); @@ -367,6 +367,7 @@ static int lx_pcm_hw_params(struct snd_pcm_substream *substream, else chip->playback_stream.stream = substream; +exit: mutex_unlock(&chip->setup_mutex); return err; } @@ -389,7 +390,7 @@ static int lx_pcm_hw_free(struct snd_pcm_substream *substream) int err = 0; int is_capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); - snd_printdd("->lx_pcm_hw_free\n"); + snd_printd("->lx_pcm_hw_free\n"); mutex_lock(&chip->setup_mutex); if (chip->hardware_running[is_capture]) { @@ -477,7 +478,7 @@ static void lx_trigger_stop(struct lx6464es *chip, struct lx_stream *lx_stream) int err; snd_printd(LXP "stopping: stopping stream\n"); - err = lx_stream_stop(chip, 0, is_capture); + err = lx_stream_pause(chip, 0, is_capture); if (err < 0) snd_printk(KERN_ERR LXP "couldn't stop stream\n"); else @@ -1005,6 +1006,7 @@ static int __devinit snd_lx6464es_create(struct snd_card *card, chip->card = card; chip->pci = pci; chip->irq = -1; + chip->hardware_running[0] = chip->hardware_running[1] = 0; /* initialize synchronization structs */ spin_lock_init(&chip->lock); diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index aea621e..2a97a21 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -30,6 +30,7 @@ #include <sound/core.h> #include <sound/pcm.h> +#include <sound/pcm_params.h> #include "lx_core.h" diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 617f98b..3a9e1dd 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -353,8 +353,11 @@ polling_successful: lx_dsp_reg_readbuf(chip, eReg_CRM2, rmh->stat, rmh->stat_len); } - } else + lx_message_dump(rmh); + } else { snd_printk(LXP "rmh error: %08x\n", reg); + dump_stack(); + } /* clear Reg_CSM_MR */ lx_dsp_reg_write(chip, eReg_CSM, 0); @@ -362,15 +365,11 @@ polling_successful: switch (reg) { case ED_DSP_TIMED_OUT: snd_printk(KERN_WARNING LXP "lx_message_send: dsp timeout\n"); - return -ETIMEDOUT; case ED_DSP_CRASHED: snd_printk(KERN_WARNING LXP "lx_message_send: dsp crashed\n"); - return -EAGAIN; } - lx_message_dump(rmh); - return reg; } @@ -532,10 +531,22 @@ int lx_pipe_allocate(struct lx6464es *chip, u32 pipe, int is_capture, err = lx_message_send_atomic(chip, &chip->rmh); spin_unlock_irqrestore(&chip->msg_lock, flags); - if (err != 0) - snd_printk(KERN_ERR "lx6464es: could not allocate pipe\n"); + if (err == 0) + return 0; - return err; + switch (err) { + case EB_ALLOCATE_AUDIO_IMPOSSIBLE: + snd_printk(KERN_ERR "lx6464es: could not allocate pipe " + "(EB_ALLOCATE_AUDIO_IMPOSSIBLE)\n"); + break; + + case EB_INVALID_AUDIO: + snd_printk(KERN_ERR "lx6464es: could not allocate pipe " + "(EB_INVALID_AUDIO)\n"); + break; + } + + return -1; } int lx_pipe_release(struct lx6464es *chip, u32 pipe, int is_capture) @@ -781,7 +792,8 @@ int lx_stream_set_state(struct lx6464es *chip, u32 pipe, return err; } -int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, +int lx_stream_set_format(struct lx6464es *chip, + struct snd_pcm_hw_params *params, u32 pipe, int is_capture) { int err; @@ -789,22 +801,19 @@ int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, u32 pipe_cmd = PIPE_INFO_TO_CMD(is_capture, pipe); - u32 channels = runtime->channels; - - if (runtime->channels != channels) - snd_printk(KERN_ERR LXP "channel count mismatch: %d vs %d", - runtime->channels, channels); + const u32 channels = params_channels(params); + const snd_pcm_format_t format = params_format(params); spin_lock_irqsave(&chip->msg_lock, flags); lx_message_init(&chip->rmh, CMD_0C_DEF_STREAM); chip->rmh.cmd[0] |= pipe_cmd; - if (runtime->sample_bits == 16) + if (snd_pcm_format_width(format) == 16) /* 16 bit format */ chip->rmh.cmd[0] |= (STREAM_FMT_16b << STREAM_FMT_OFFSET); - if (snd_pcm_format_little_endian(runtime->format)) + if (snd_pcm_format_little_endian(format)) /* little endian/intel format */ chip->rmh.cmd[0] |= (STREAM_FMT_intel << STREAM_FMT_OFFSET); @@ -974,7 +983,7 @@ int lx_level_unmute(struct lx6464es *chip, int is_capture, int unmute) chip->rmh.cmd[1] = (u32)(mute_mask >> (u64)32); /* hi part */ chip->rmh.cmd[2] = (u32)(mute_mask & (u64)0xFFFFFFFF); /* lo part */ - snd_printk("mute %x %x %x\n", chip->rmh.cmd[0], chip->rmh.cmd[1], + snd_printd("mute %x %x %x\n", chip->rmh.cmd[0], chip->rmh.cmd[1], chip->rmh.cmd[2]); err = lx_message_send_atomic(chip, &chip->rmh); diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 6bd9cbb..378721a 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -134,7 +134,8 @@ int lx_pipe_wait_for_start(struct lx6464es *chip, u32 pipe, int is_capture); int lx_pipe_wait_for_idle(struct lx6464es *chip, u32 pipe, int is_capture); /* low-level stream handling */ -int lx_stream_set_format(struct lx6464es *chip, struct snd_pcm_runtime *runtime, +int lx_stream_set_format(struct lx6464es *chip, + struct snd_pcm_hw_params *runtime, u32 pipe, int is_capture); int lx_stream_state(struct lx6464es *chip, u32 pipe, int is_capture, int *rstate); -- 1.7.4.1

2 2

[alsa-devel] HDAudio period/buffer sizes
by pl bossart 26 Jul '11

26 Jul '11

When I look at the HDAudio spec, I see that the BDL entries need to be aligned on a 128-byte boundary, but the buffer length can be any value. When I look at the HDAudio driver (azx_pcm_open in hda_codec.c), I see that we enforce a 128-byte granularity on the period/buffer sizes, but this was added in a patch that fixed alignment issues (5f1545bc) Any technical reason why the driver is more demanding than the spec? Or is this a confusion between alignment and buffer size? I can understand that rounding to multiple of 128 bytes would be more efficient in terms of data access, but it also prevents applications from getting an interrupt when they want (eg it's impossible to use 20ms periods with a 44.1kHz sampling rate, you'll get 19.59 or 20.31ms). Thanks, -Pierre

2 1

[alsa-devel] [PATCH] ASoC: dapm - change stream event dbg to vdgb
by Liam Girdwood 26 Jul '11

26 Jul '11

Stream event debug can be noisy on larger audio devices so improve the debug SNR by changing it to the verbose level. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 120df5b..3956670 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2710,7 +2710,7 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, { if (!w->sname || w->dapm != dapm) continue; - dev_dbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", + dev_vdbg(w->dapm->dev, "widget %s\n %s stream %s event %d\n", w->name, w->sname, stream, event); if (strstr(w->sname, stream)) { switch(event) { -- 1.7.4.1

2 1

[alsa-devel] [PATCH v2] ASoC: dapm - Add DAPM stream completion event.
by Liam Girdwood 26 Jul '11

26 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. This adds a callback function to be called at the completion of a DAPM stream event. This can be used by DSP components to perform calculations based on DAPM graphs after completion of stream events. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 4 ++++ 4 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..5833f01 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -518,6 +518,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..64a9dd5 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -634,6 +634,9 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + /* codec stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -669,6 +672,9 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + /* platform stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm, int event); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..159f144 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,6 +3140,7 @@ int snd_soc_register_platform(struct device *dev, platform->driver = platform_drv; platform->dapm.dev = dev; platform->dapm.platform = platform; + platform->dapm.stream_event = platform_drv->stream_event; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); @@ -3252,6 +3253,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.dev = dev; codec->dapm.codec = codec; codec->dapm.seq_notifier = codec_drv->seq_notifier; + codec->dapm.stream_event = codec_drv->stream_event; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fbfcda0..80f97b2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2574,6 +2574,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); + + /* do we need to notify any clients that DAPM stream is complete */ + if (dapm->stream_event) + dapm->stream_event(dapm, event); } /** -- 1.7.4.1

2 1

[alsa-devel] M-Audio FTU issues
by Daniel Mack 25 Jul '11

25 Jul '11

So, I've looked into this today and as expected, the device normally operates in implicit feedback mode. After installing a proprietary driver on Mac OS X (that regularily crashed my kernel), the analyzer shows that for every inbound packet, an out packet is sent back with the same amount of data in it. And there are no feedback endpoints. On Tue, Jun 14, 2011 at 11:21 AM, Aurélien Leblond <blablack(a)gmail.com> wrote: > There are still the issue of samples dropping. > > Tto reproduce it: > - Open Audacity and set it up to be using Jack. > - Click Generate -> Tone -> Ok. > > "Normally" you should hear very regular pops... > What surprises me though, is that I can't hear the clicks and pops you guys describe. Instead, I hear a quite annoying distortion for both 44100 and 48000 Hz which I can't explain either. My spontanous guess is that there is a resampler in the chain which compensates the sample drift and notably reduces the audio quality. I patched the driver so that it supports implicit feedback, but that didn't change anything in that regard. Hence my question is: How do you guys test? I renderened a sine tone in different samplerates and bit depths to .wav files and play them with aplay (no pulseaudio or anything in the chain). Could you try the same and report what happens? Daniel

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[alsa-devel] [PATCH V4 0/3] ASoC for Alchemy Au1000/1500/1100
by Manuel Lauss 25 Jul '11

25 Jul '11

Hello, Here's V4 of the AC97/I2S ASoC patchset for early Alchemy chips and their respective evaluation boards. The patches are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V3: - dropped the hunk which removed the I2S constants from the au1000.h header to avoid merge conflicts with other patches. - use the context structure declared in psc.h. Follow-up patches for psc* code depend on this. Changes since V2: - implemented changes after feedback from Lars-Peter Clausen: * split patch 1 in two, one for the ASoC drivers, and a separate for DB1000 machine code. * get rid of automatic dma device registration * tidied the I2S/AC97 sources - mark sound/mips/au1x00.c as DEPRECATED instead of removing it outright. Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (3): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver arch/mips/alchemy/devboards/db1x00/platform.c | 48 ++++ sound/mips/Kconfig | 5 +- sound/soc/au1x/Kconfig | 28 ++ sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 365 ++++++++++++++++++++++++ sound/soc/au1x/db1000.c | 75 +++++ sound/soc/au1x/dma.c | 374 +++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 342 ++++++++++++++++++++++ sound/soc/au1x/psc.h | 19 +- 9 files changed, 1256 insertions(+), 10 deletions(-) create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 7

Re: [alsa-devel] ac97: quiet make W=1 noise by initializing all fields of snd_ac97_codec_id_vendors and snd_ac97_codec_ids.
by Takashi Iwai 25 Jul '11

25 Jul '11

At Sun, 24 Jul 2011 11:40:56 +0200, Lionel Debroux wrote: > > make W=1 bitterly complains about missing initializations in > snd_ac97_codec_id_vendors and snd_ac97_codec_ids. > Initialize the last field, to make the output of make W=1 more readable. > > Patch against current Linus tree. If we change the whole table entries, it'd be better to use C99 style init, e.g. #define _CODEC(_id, _mask, _name) \ .id = _id, .mask = _mask, .name = _name static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { { _CODEC(0x41445300, 0xffffff00, "Analog Devices"),}, ... static const struct ac97_codec_id snd_ac97_codec_ids[] = { { _CODEC(0x41445303, 0xffffffff, "AD1819"), .patch = patch_ad1819 }, ... { _CODEC(0x53494c20, 0xffffffe0, "Si3036,8"), .patch = mpatch_si3036, \ .mpatch = mpatch_si3036, .flags = AC97_MODEM_PATCH }, ... thanks, Takashi

1 0

[alsa-devel] [PATCH] ASoC: Add regmap as a control type
by Mark Brown 25 Jul '11

25 Jul '11

Allow drivers to set up their own regmap API structures. This is mainly useful with MFDs where the core driver will have set up regmap at the minute, though it may make sense to push the existing regmap setup out of the core into the drivers. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 1 + sound/soc/soc-io.c | 4 ++++ 2 files changed, 5 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 4d04b4b..d022694 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -261,6 +261,7 @@ extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { SND_SOC_I2C = 1, SND_SOC_SPI, + SND_SOC_REGMAP, }; enum snd_soc_compress_type { diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index b56e1c4..e471ed6 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -132,6 +132,10 @@ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, &config); break; + case SND_SOC_REGMAP: + /* Device has made its own regmap arrangements */ + break; + default: return -EINVAL; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH 0/2] ASoC: au1x: update PSC AC97/I2S code
by Manuel Lauss 25 Jul '11

25 Jul '11

The following two patches depend on patchset "ASoC for Alchemy Au1000/1500/1100" to apply and work. Patch #1 moves the DBDMA (PCM) device registration from the AC97/I2S drivers into the board code. Patch #2 changes use of "soc-audio" platform device int db1200 machine code to the new way. Run-tested on DB1200, DB1300 and DB1550 boards. Manuel Lauss (2): ASoC: au1x: remove automatic DMA device registration from PSC drivers ASoC: au1x: update db1200 machine to the new way of things arch/mips/alchemy/devboards/db1200/platform.c | 16 +++++ sound/soc/au1x/db1200.c | 64 +++++++++++++------ sound/soc/au1x/dbdma2.c | 83 ++++--------------------- sound/soc/au1x/psc-ac97.c | 34 +++++++--- sound/soc/au1x/psc-i2s.c | 32 +++++++--- sound/soc/au1x/psc.h | 5 -- 6 files changed, 117 insertions(+), 117 deletions(-) -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 3

[alsa-devel] ASoC updates for 3.1
by Mark Brown 25 Jul '11

25 Jul '11

The following changes since commit 440085598672c0e3fde8a48495f61fea418b06d1: Merge branch 'for-3.0' into for-3.1 (2011-07-17 18:25:58 +0900) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Sorry, I thought I'd sent this out and then the release caught me on the hop a bit. Apart from Liam's changes and the new driver these are all bug fixes of one form or another - the sgtl5000 driver apparently just didn't work most of the time. Liam Girdwood (2): ASoC: dapm - add DAPM macro for external enum widgets ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context. Mark Brown (3): ASoC: Mark cache as dirty when suspending ASoC: Acknowledge WM8962 interrupts before acting on them ASoC: Revert "ASoC: SAMSUNG: Add I2S0 internal dma driver" Rajashekhara, Sudhakar (2): ASoC: davinci: fix codec start and stop functions ASoC: davinci: add missing break statement Sangbeom Kim (2): ASoC: SAMSUNG: Modify I2S driver to support idma ASoC: SAMSUNG: Add I2S0 internal dma driver Wolfram Sang (3): ASoC: core: make comments fit the code ASoC: sgtl5000: refactor registering internal ldo ASoC: sgtl5000: guide user when regulator support is needed include/sound/soc-dapm.h | 6 +++ sound/soc/codecs/sgtl5000.c | 70 +++++++++++++++++-------------------- sound/soc/codecs/wm8962.c | 6 ++-- sound/soc/davinci/davinci-vcif.c | 9 +++-- sound/soc/samsung/i2s.c | 7 ++++ sound/soc/soc-core.c | 5 ++- sound/soc/soc-dapm.c | 30 ++++++++++++++++ 7 files changed, 86 insertions(+), 47 deletions(-)

2 1

[alsa-devel] lib pcm DMA problems
by Horner, Richard 24 Jul '11

24 Jul '11

To Whom It May Concern: I see some weird behavior from the PCI audio driver for the Hauppauge analog tv card (model 44842, rev D1B2) where the card is tuning properly but audio is no longer available over PCI bus. This condition can happen within an hour or as late as after multiple months of 24x7 operation. Re-loading the TV card and ALSA driver stacks do not clear this condition. Only a warm boot clears it up. Debug prints in the PCI device sound driver, bt87x.c. "snd_bt87x_interrupt()" handler indicate it is returning only IRQ_NONE when in this state. We also get a DMA error message from pcm_lib.c which states "capture write error (DMA or IRQ trouble?)", which I presume means DMA is turned off for this device. How can I recover from this without a system reboot? Thank you so much! Here is my system info: # cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 7 model name : VIA Samuel 2 stepping : 3 cpu MHz : 533.309 cache size : 64 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu de tsc msr cx8 mtrr pge mmx 3dnow up bogomips : 1066.61 clflush size : 32 power management: ----------------------------------------------------------- Interrupts are still active on IRQ 5 during this state: # cat /proc/interrupts CPU0 0: 177526733 XT-PIC-XT timer 1: 2 XT-PIC-XT i8042 2: 0 XT-PIC-XT cascade 4: 1 XT-PIC-XT 5: 18834539 XT-PIC-XT bttv0, Bt87x audio 7: 0 XT-PIC-XT parport0 8: 0 XT-PIC-XT rtc0 9: 1587422 XT-PIC-XT eth0 10: 12183885 XT-PIC-XT VIA686A 11: 77875422 XT-PIC-XT uhci_hcd:usb1, uhci_hcd:usb2 12: 7 XT-PIC-XT i8042 14: 17133058 XT-PIC-XT ide0 15: 0 XT-PIC-XT ide1 NMI: 0 Non-maskable interrupts LOC: 0 Local timer interrupts SPU: 0 Spurious interrupts RES: 0 Rescheduling interrupts CAL: 0 Function call interrupts TLB: 0 TLB shootdowns TRM: 0 Thermal event interrupts ERR: 0 MIS: 0 Rich Horner Engineering and Technology The Nielsen Company 813-366-5060 www.nielsen.com

1 0

[alsa-devel] How to loopback two PCM interface?
by onlyfever 24 Jul '11

24 Jul '11

Hi, I configure McBSP3 and McBSP4 as PCM interface. I want to connect two PCM interface internally. The data captured from PCM1 can be directly sent to PCM2 playback interface. How can I achieve that? Please give me some advise. thank you!

1 0

Re: [alsa-devel] sound: constify snd_ac97_bus_ops
by Takashi Iwai 24 Jul '11

24 Jul '11

At Sun, 24 Jul 2011 11:40:48 +0200, Lionel Debroux wrote: > > The fields of this struct type don't seem to be used as rvalues, besides > the initialization of struct instances. Let's make const the struct > instances. > > Maybe the Documentation/DocBook/writing-an-alsa-driver.tmpl, and > checkpatch.pl should be updated to reflect this change ? Yes, please either update or give an additional patch for document fixes. I'll apply these series not for 3.1 since this changes API. Will be applied the branches for 3.2. thanks, Takashi

1 0

[alsa-devel] [PATCH] ALSA: Make snd_pcm_debug_name usable outside pcm_lib
by linux＠audioscience.com 24 Jul '11

24 Jul '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Formatting a PCM name is useful for module debug too. Add snd_prefix when making function public. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- include/sound/pcm.h | 12 ++++++++++++ sound/core/pcm_lib.c | 10 +++++----- sound/pci/asihpi/asihpi.c | 21 --------------------- 3 files changed, 17 insertions(+), 26 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e1bad11..c83c754 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -507,6 +507,18 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream); void snd_pcm_vma_notify_data(void *client, void *data); int snd_pcm_mmap_data(struct snd_pcm_substream *substream, struct file *file, struct vm_area_struct *area); + +#if defined CONFIG_SND_DEBUG +void snd_pcm_debug_name(struct snd_pcm_substream *substream, + char *name, size_t len); +#else +static inline void +snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) +{ + *buf = 0; +} +#endif + /* * PCM library */ diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index f134130..f82329d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -128,7 +128,7 @@ void snd_pcm_playback_silence(struct snd_pcm_substream *substream, snd_pcm_ufram } } -static void pcm_debug_name(struct snd_pcm_substream *substream, +void snd_pcm_debug_name(struct snd_pcm_substream *substream, char *name, size_t len) { snprintf(name, len, "pcmC%dD%d%c:%d", @@ -168,7 +168,7 @@ static void xrun(struct snd_pcm_substream *substream) snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); if (xrun_debug(substream, XRUN_DEBUG_BASIC)) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); snd_printd(KERN_DEBUG "XRUN: %s\n", name); dump_stack_on_xrun(substream); } @@ -243,7 +243,7 @@ static void xrun_log_show(struct snd_pcm_substream *substream) return; if (xrun_debug(substream, XRUN_DEBUG_LOGONCE) && log->hit) return; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); for (cnt = 0, idx = log->idx; cnt < XRUN_LOG_CNT; cnt++) { entry = &log->entries[idx]; if (entry->period_size == 0) @@ -319,7 +319,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (pos >= runtime->buffer_size) { if (printk_ratelimit()) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); xrun_log_show(substream); snd_printd(KERN_ERR "BUG: %s, pos = %ld, " "buffer size = %ld, period size = %ld\n", @@ -364,7 +364,7 @@ static int snd_pcm_update_hw_ptr0(struct snd_pcm_substream *substream, if (xrun_debug(substream, in_interrupt ? XRUN_DEBUG_PERIODUPDATE : XRUN_DEBUG_HWPTRUPDATE)) { char name[16]; - pcm_debug_name(substream, name, sizeof(name)); + snd_pcm_debug_name(substream, name, sizeof(name)); snd_printd("%s_update: %s: pos=%u/%u/%u, " "hwptr=%ld/%ld/%ld/%ld\n", in_interrupt ? "period" : "hwptr", diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index b941d25..eae62eb 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -41,31 +41,10 @@ #include <sound/tlv.h> #include <sound/hwdep.h> - MODULE_LICENSE("GPL"); MODULE_AUTHOR("AudioScience inc. <support(a)audioscience.com>"); MODULE_DESCRIPTION("AudioScience ALSA ASI5000 ASI6000 ASI87xx ASI89xx"); -#if defined CONFIG_SND_DEBUG -/* copied from pcm_lib.c, hope later patch will make that version public -and this copy can be removed */ -static inline void -snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) -{ - snprintf(buf, size, "pcmC%dD%d%c:%d", - substream->pcm->card->number, - substream->pcm->device, - substream->stream ? 'c' : 'p', - substream->number); -} -#else -static inline void -snd_pcm_debug_name(struct snd_pcm_substream *substream, char *buf, size_t size) -{ - *buf = 0; -} -#endif - #if defined CONFIG_SND_DEBUG_VERBOSE /** * snd_printddd - very verbose debug printk -- 1.7.0.4

3 4

Re: [alsa-devel] [arch-general] Panic - no sound devices after upgrade
by Adrian Knoth 23 Jul '11

23 Jul '11

On 07/23/11 16:21, Fons Adriaensen wrote: Note: I suggest to narrow the CC list a bit and better move the discussion to alsa-devel. At least, let's drop LKML and arch. >>> Seems to have solved the problem :-) :-) >> Good to know. I'll wait for your final confirmation and then commit it. > > One problem: the gain settings have changed, now having a range of 0..64. > How do these values map to the internal ones (0..0xffff) ? I *hope* there > is value corresponding to EXACTLY 0x8000 (gain = 1) !!! For the simple ALSA mixer, the driver does gain = ucontrol->value.integer.value[0]*UNITY_GAIN/64; UNITY_GAIN is 32768 (0x8000), so if you select 64 on the ALSA side, it gives you exactly what you need. Or in other words: faders to 100% in alsamixer, that's it. With hdspmixer or any other software that's using the full mixer, you can still use the fine-grained 65k steps. > BTW, why was this change made? No idea. I reckon faberman was tired of stepping through 65k mixer levels in alsamixer. ;) (disclaimer: I never used the old driver, so I don't know how it was before) Cheers

1 0

[alsa-devel] Sending headset insertion events to user space on bootup
by Brad Rubin 23 Jul '11

23 Jul '11

Hi all, I’m having an issue passing events when a headset is inserted upon bootup. I’m sending a headset insertion event to user space through snd_soc_jack_report, but user space is not yet ready to receive events and therefore misses the event. Since the kernel driver doesn’t resend this event, then there’s no way for user space to see it. User space is still booting up when the event is sent. Does anyone know of a solution so user space can receive headset insertion events on bootup? Is is possible to either allow the kernel to postpone sending events until user space is ready or to store events in some sort of cache so that user space can access them? Would it be possible to use ALSA to store the headset status in a device node? Thanks, Brad -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 1

[alsa-devel] ASoC Peek individual register debugfs
by Patrick Lai 23 Jul '11

23 Jul '11

Hi, Does latest soc-core.c provide debugfs interface for peeking individual register? If not, I plan to add this interface. Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 3

[alsa-devel] ABE routing table without TWL6040
by hong zhang 23 Jul '11

23 Jul '11

List, My custom OMAP4 board does not have TWL6040 device. What I do is 1. comment out TWL6040 in arch/arm/mach-omap2/Kconfig 2. comment out twl6040 inside sdp4430_dai[] at sound/soc/omap/sdp4430.c I find ABE routing table is no longer useful because of changes in steps 1 and 2. run "arecord -f S16_LE -r 48000 -c 2 -D plughw:0,0 try.wav" returns error: Recording WAVE 'try.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo try.wav: No such file or directory dmesg shows dmic_trigger is called. 1. Could anyone please give me hint where I should look into? 2. Try Legacy DMIC to turn off ABE by comment out ABE in sound/soc/omap/kconfig but "No Sound card Found" when boot. Thanks. ---henry

1 0

[alsa-devel] [PATCH] sound, hda_codec: Fix compilation warnings
by Borislav Petkov 22 Jul '11

22 Jul '11

From: Borislav Petkov <borislav.petkov(a)amd.com> Convert macro to an inline function to shut up gcc: sound/pci/hda/patch_sigmatel.c: In function `stac92xx_init´: sound/pci/hda/patch_sigmatel.c:4385:3: warning: statement with no effect sound/pci/hda/patch_sigmatel.c: In function `stac92xx_resume´: sound/pci/hda/patch_sigmatel.c:4925:3: warning: statement with no effect sound/pci/hda/hda_codec.c: In function `snd_hda_mixer_amp_switch_put´: sound/pci/hda/patch_realtek.c:4176:2: warning: statement with no effect sound/pci/hda/hda_codec.c:2320:2: warning: statement with no effect sound/pci/hda/patch_realtek.c: In function `alc_resume´: sound/pci/hda/patch_realtek.c:4558:2: warning: statement with no effect sound/pci/hda/patch_realtek.c: In function `alc269_resume´: sound/pci/hda/patch_realtek.c:14878:2: warning: statement with no effect Signed-off-by: Borislav Petkov <borislav.petkov(a)amd.com> --- sound/pci/hda/hda_codec.h | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 59c9730..79ede92 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -1005,8 +1005,12 @@ int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) return codec->patch_ops.check_power_status(codec, nid); return 0; } -#else -#define hda_call_check_power_status(codec, nid) 0 +#else +static inline +int hda_call_check_power_status(struct hda_codec *codec, hda_nid_t nid) +{ + return 0; +} #endif /* -- 1.7.4.rc2

2 1

[alsa-devel] additional problem
by David Henderson 22 Jul '11

22 Jul '11

Hi gang! I've successfully been able to compile the alsa-utils package with the "--with-alsa-inc-prefix=/opt/staging/alsa/var/share/include --with-alsa-prefix=/opt/staging/alsa/lib", but the problem I'm having now is that the compiled binaries are looking for those directories during run-time and not just compile-time. Does anyone have any thoughts on using those directories for package creation, but that the software doesn't use the '/opt/staging/alsa' prefix during run-time? Thanks, Dave

4 25

[alsa-devel] [PATCH V3 0/3] ASoC for Alchemy Au1000/1500/1100
by Manuel Lauss 22 Jul '11

22 Jul '11

Hello, These 3 patches implement ASoC drivers for the AC97 and I2S controllers found on early Alchemy chips. They are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V2: - implemented changes after feedback from Lars-Peter Clausen: * split patch 1 in two, one for the ASoC drivers, and a separate for DB1000 machine code. * get rid of automatic dma device registration * tidied the I2S/AC97 sources - mark sound/mips/au1x00.c as DEPRECATED instead of removing it outright. Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (3): ASoC: Alchemy AC97C/I2SC audio support ASoC: Add a DB1x00 AC97 machine driver ALSA: deprecate MIPS AU1X00 AC97 driver arch/mips/alchemy/devboards/db1x00/platform.c | 48 +++ arch/mips/include/asm/mach-au1x00/au1000.h | 31 -- sound/mips/Kconfig | 5 +- sound/soc/au1x/Kconfig | 28 ++ sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 375 ++++++++++++++++++++++++ sound/soc/au1x/db1000.c | 75 +++++ sound/soc/au1x/dma.c | 381 +++++++++++++++++++++++++ sound/soc/au1x/i2sc.c | 348 ++++++++++++++++++++++ sound/soc/au1x/psc.h | 22 +- 10 files changed, 1282 insertions(+), 41 deletions(-) create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 3

[alsa-devel] [PATCH] regulator: Make core more chatty about some errors
by Mark Brown 22 Jul '11

22 Jul '11

Prevent some head scratching by making the core log about some rare but possible errors with invalid voltage ranges and modes being set. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/regulator/core.c | 11 +++++++++-- 1 files changed, 9 insertions(+), 2 deletions(-) diff --git a/drivers/regulator/core.c b/drivers/regulator/core.c index f59821f..3700d09 100644 --- a/drivers/regulator/core.c +++ b/drivers/regulator/core.c @@ -151,8 +151,11 @@ static int regulator_check_voltage(struct regulator_dev *rdev, if (*min_uV < rdev->constraints->min_uV) *min_uV = rdev->constraints->min_uV; - if (*min_uV > *max_uV) + if (*min_uV > *max_uV) { + rdev_err(rdev, "unsupportable voltage range: %d-%duV\n", + min_uV, max_uV); return -EINVAL; + } return 0; } @@ -205,8 +208,11 @@ static int regulator_check_current_limit(struct regulator_dev *rdev, if (*min_uA < rdev->constraints->min_uA) *min_uA = rdev->constraints->min_uA; - if (*min_uA > *max_uA) + if (*min_uA > *max_uA) { + rdev_err(rdev, "unsupportable current range: %d-%duA\n", + min_uA, max_uA); return -EINVAL; + } return 0; } @@ -221,6 +227,7 @@ static int regulator_mode_constrain(struct regulator_dev *rdev, int *mode) case REGULATOR_MODE_STANDBY: break; default: + rdev_err(rdev, "invalid mode %x specified\n", *mode); return -EINVAL; } -- 1.7.5.4

2 1

[alsa-devel] [PATCH v2] ASoC: dapm - Add API call to query the widgets of valid DAPM paths.
by Liam Girdwood 22 Jul '11

22 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. Add a DAPM API call to determine whether a DAPM audio path is valid between source and sink widgets and return the path widgets. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the Dynamic PCM core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. This modifies the existing is_connected_input_ep() and is_connected_output_ep() by adding a new parameter to store any valid widgets in a list. We also add some extra debug to provide further insight into the DAPM path and widget discovery. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++ sound/soc/soc-dapm.c | 184 +++++++++++++++++++++++++++++++++++++++++++--- 2 files changed, 178 insertions(+), 12 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..198997d 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,11 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index fbfcda0..4205a97 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -605,11 +605,51 @@ static int snd_soc_dapm_suspend_check(struct snd_soc_dapm_widget *widget) } } +/* add widget to list if it's not already in the list */ +static int dapm_list_add_widget(struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + if (*list == NULL) + return -EINVAL; + + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + /* allocate some new space */ + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(w->dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + wlist = *list; + + /* insert the widget */ + dev_dbg(w->dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + /* * Recursively check for a completed path to an active or physically connected * output widget. Returns number of complete paths. */ -static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_output_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -638,15 +678,38 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sinks, list_source) { + if (path->weak) continue; if (path->walked) continue; - if (path->sink && path->connect) { + + if (path->sink) { + + dev_dbg(widget->dapm->dev," %c : %s -> %s -> %s\n", + path->sink && path->connect ? '*' : ' ', + widget->name, path->name, path->sink->name); + + if (!path->connect) + continue; + path->walked = 1; - con += is_connected_output_ep(path->sink); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->sink); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_output_ep(path->sink, list); } } @@ -657,7 +720,8 @@ static int is_connected_output_ep(struct snd_soc_dapm_widget *widget) * Recursively check for a completed path to an active or physically connected * input widget. Returns number of complete paths. */ -static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) +static int is_connected_input_ep(struct snd_soc_dapm_widget *widget, + struct snd_soc_dapm_widget_list **list) { struct snd_soc_dapm_path *path; int con = 0; @@ -691,21 +755,117 @@ static int is_connected_input_ep(struct snd_soc_dapm_widget *widget) } list_for_each_entry(path, &widget->sources, list_sink) { + if (path->weak) continue; if (path->walked) continue; - if (path->source && path->connect) { + if (path->source) { + + dev_dbg(widget->dapm->dev," %c : %s <- %s <- %s\n", + path->source && path->connect ? '*' : ' ', + widget->name, path->name, path->source->name); + + if (!path->connect) + continue; + path->walked = 1; - con += is_connected_input_ep(path->source); + + /* do we need to add this widget to the list ? */ + if (list) { + int err; + + err = dapm_list_add_widget(list, path->source); + if (err < 0) { + dev_err(widget->dapm->dev, "could not add widget %s\n", + widget->name); + return con; + } + } + + con += is_connected_input_ep(path->source, list); } } return con; } +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + paths = is_connected_output_ep(root, list); + dev_dbg(dapm->dev, "Playback: found %d paths from %s\n", paths, root->name); + + return paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + int paths; + + dev_dbg(dapm->dev, "Capture: checking paths from %s\n",root->name); + paths = is_connected_input_ep(root, list); + dev_dbg(dapm->dev, "Capture: found %d paths from %s\n", paths, root->name); + + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial widget specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream) +{ + struct snd_soc_dapm_widget *w; + enum snd_soc_dapm_type type; + int paths; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + type = snd_soc_dapm_aif_in; + else + type = snd_soc_dapm_aif_out; + + /* get stream root widget AIF from stream string and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (w->id != type) + continue; + + if (strstr(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget for %s not found\n", stream_name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_walk(dapm); + return paths; +} + /* * Handler for generic register modifier widget. */ @@ -732,9 +892,9 @@ static int dapm_generic_check_power(struct snd_soc_dapm_widget *w) { int in, out; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0 && in != 0; } @@ -745,7 +905,7 @@ static int dapm_adc_check_power(struct snd_soc_dapm_widget *w) int in; if (w->active) { - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); return in != 0; } else { @@ -759,7 +919,7 @@ static int dapm_dac_check_power(struct snd_soc_dapm_widget *w) int out; if (w->active) { - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); return out != 0; } else { @@ -1315,9 +1475,9 @@ static ssize_t dapm_widget_power_read_file(struct file *file, if (!buf) return -ENOMEM; - in = is_connected_input_ep(w); + in = is_connected_input_ep(w, NULL); dapm_clear_walk(w->dapm); - out = is_connected_output_ep(w); + out = is_connected_output_ep(w, NULL); dapm_clear_walk(w->dapm); ret = snprintf(buf, PAGE_SIZE, "%s: %s in %d out %d", -- 1.7.4.1

2 3

[alsa-devel] [PATCH V2 0/2] ALSA: ASoC for Au1000/1500/1100
by Manuel Lauss 22 Jul '11

22 Jul '11

Hello, These 2 patches implement ASoC drivers for the AC97 and I2S controllers found on early Alchemy chips. They are largely based on the old mips/au1x00.c driver which they replace. AC97 Tested on a Db1500 development board; I2S untested since none of the testboards I have actually have an I2S codec (just testpoints). Changes since V1: - added untested I2S controller driver for completeness, removed the audio defines from the au1000 header. Manuel Lauss (2): ALSA: Alchemy AC97C/I2SC audio support ALSA: delete MIPS au1x00 driver arch/mips/alchemy/devboards/db1x00/platform.c | 37 ++ arch/mips/include/asm/mach-au1x00/au1000.h | 61 --- sound/mips/Kconfig | 8 - sound/mips/Makefile | 2 - sound/mips/au1x00.c | 695 ------------------------- sound/soc/au1x/Kconfig | 28 + sound/soc/au1x/Makefile | 10 + sound/soc/au1x/ac97c.c | 398 ++++++++++++++ sound/soc/au1x/db1000.c | 75 +++ sound/soc/au1x/dma.c | 470 +++++++++++++++++ sound/soc/au1x/i2sc.c | 353 +++++++++++++ sound/soc/au1x/psc.h | 31 +- 12 files changed, 1393 insertions(+), 775 deletions(-) delete mode 100644 sound/mips/au1x00.c create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c create mode 100644 sound/soc/au1x/i2sc.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

3 9

Re: [alsa-devel] asihpi - update from HPI 4.06 to 4.08 - Firmware
by Takashi Iwai 22 Jul '11

22 Jul '11

At Fri, 22 Jul 2011 16:05:43 +1200, Eliot Blennerhassett wrote: > > On 22/07/11 15:52, linux(a)audioscience.com wrote: > > From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> > > > > Main visible change is the addition of mute switch to volume controls. > > The rest are relatively minor changes and cleanups to HPI code. > > I also forgot to mention that the associated firmware/asihpi should be > updated with the contents of > > http://www.audioscience.com/internet/download/drivers/released/v4/08/02/asi… OK, now updated alsa-firmware tree, too. thanks, Takashi

1 0

[alsa-devel] [PATCH 00/15] ALSA: asihpi - update from HPI 4.06 to 4.08
by linux＠audioscience.com 22 Jul '11

22 Jul '11

From: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> Main visible change is the addition of mute switch to volume controls. The rest are relatively minor changes and cleanups to HPI code. patch#6 reverts commit 4a122c10f "check adapter index in hpi_ioctl" The associated 4.08 firmware release adds support for new cards ASI5544/5541/5540/5520 Eliot Blennerhassett (15): ALSA: asihpi - Give more meaningful name to hpi request message type ALSA: asihpi - Increase request and response buffer sizes ALSA: asihpi - Remove controlex structs and associated special data transfer code ALSA: asihpi - DSP code loader API now independent of OS ALSA: asihpi - Revise snd_pcm_debug_name, get rid of DEBUG_NAME macro ALSA: asihpi - Remove spurious adapter index check ALSA: asihpi - Remove unused structures, macros and functions ALSA: asihpi - Fix minor typos and spelling ALSA: asihpi - Make local function static ALSA: asihpi - Add new node and message defines ALSA: asihpi - Explicitly include mutex.h ALSA: asihpi - Use size_t for sizeof result ALSA: asihpi - Control name updates ALSA: asihpi - Add volume mute controls ALSA: asihpi - HPI version 4.08 sound/pci/asihpi/asihpi.c | 87 ++++++++++++++++++---- sound/pci/asihpi/hpi.h | 24 ++++-- sound/pci/asihpi/hpi6000.c | 11 +-- sound/pci/asihpi/hpi6205.c | 52 ++++--------- sound/pci/asihpi/hpi6205.h | 25 +++++-- sound/pci/asihpi/hpi_internal.h | 155 ++++++++++---------------------------- sound/pci/asihpi/hpicmn.c | 17 ++-- sound/pci/asihpi/hpidspcd.c | 137 ++++++++++++++-------------------- sound/pci/asihpi/hpidspcd.h | 72 ++++++++++-------- sound/pci/asihpi/hpifunc.c | 86 ++++++++++----------- sound/pci/asihpi/hpimsginit.c | 4 +- sound/pci/asihpi/hpimsgx.c | 6 +- sound/pci/asihpi/hpioctl.c | 10 +-- sound/pci/asihpi/hpios.c | 8 -- sound/pci/asihpi/hpios.h | 1 + 15 files changed, 319 insertions(+), 376 deletions(-)

3 34

[alsa-devel] Alsa Driver Compilation Problem Report.
by Georges Jahchan 21 Jul '11

21 Jul '11

Attached is the ALSA driver compile script log on Fedora Core 13 x86_64 on an HP Pavilion DV7-1223CA with AMD Turion X2 64-bit CPU, ATI HDMI Audio and Intel PCI Audio. uname -a Linux hostname 2.6.34.9-69.fc13.x86_64 #1 SMP Tue May 3 09:23:03 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux

1 0

[alsa-devel] [REPOST][PATCH v4 3/4] ASoC: SAMSUNG: Modify I2S driver to support idma
by Sangbeom Kim 21 Jul '11

21 Jul '11

Previously, I2S driver only can support system dma. In this patch, i2s driver can support internal dma too. IDMA h/w configuration is initialized on idma.c Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- As Mark brown request, I repost i2s v4 patch sound/soc/samsung/i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1568eea..c086b78 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -21,6 +21,7 @@ #include <plat/audio.h> #include "dma.h" +#include "idma.h" #include "i2s.h" #include "i2s-regs.h" @@ -60,6 +61,7 @@ struct i2s_dai { /* DMA parameters */ struct s3c_dma_params dma_playback; struct s3c_dma_params dma_capture; + struct s3c_dma_params idma_playback; u32 quirks; u32 suspend_i2smod; u32 suspend_i2scon; @@ -877,6 +879,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (i2s->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, i2s->addr + I2SCON); + if (i2s->quirks & QUIRK_SEC_DAI) + idma_reg_addr_init((void *)i2s->addr, + i2s->sec_dai->idma_playback.dma_addr); + probe_exit: /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; @@ -1077,6 +1083,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) sec_dai->dma_playback.dma_size = 4; sec_dai->base = regs_base; sec_dai->quirks = quirks; + sec_dai->idma_playback.dma_addr = i2s_cfg->idma_addr; sec_dai->pri_dai = pri_dai; pri_dai->sec_dai = sec_dai; } -- 1.7.1

3 5

[alsa-devel] [PATCH] mfd: Acknowledge WM8994 IRQs before reporting
by Mark Brown 21 Jul '11

21 Jul '11

This ensures we never have a window where we've handled an interrupt but not told the hardware about it. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- drivers/mfd/wm8994-irq.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/drivers/mfd/wm8994-irq.c b/drivers/mfd/wm8994-irq.c index 71c6e8f..d682f7b 100644 --- a/drivers/mfd/wm8994-irq.c +++ b/drivers/mfd/wm8994-irq.c @@ -231,12 +231,6 @@ static irqreturn_t wm8994_irq_thread(int irq, void *data) status[i] &= ~wm8994->irq_masks_cur[i]; } - /* Report */ - for (i = 0; i < ARRAY_SIZE(wm8994_irqs); i++) { - if (status[wm8994_irqs[i].reg - 1] & wm8994_irqs[i].mask) - handle_nested_irq(wm8994->irq_base + i); - } - /* Ack any unmasked IRQs */ for (i = 0; i < ARRAY_SIZE(status); i++) { if (status[i]) @@ -244,6 +238,12 @@ static irqreturn_t wm8994_irq_thread(int irq, void *data) status[i]); } + /* Report */ + for (i = 0; i < ARRAY_SIZE(wm8994_irqs); i++) { + if (status[wm8994_irqs[i].reg - 1] & wm8994_irqs[i].mask) + handle_nested_irq(wm8994->irq_base + i); + } + return IRQ_HANDLED; } -- 1.7.5.4

1 0

[alsa-devel] [PATCH 0/2] ALSA: ASoC for Au1000/1500/1100
by Manuel Lauss 21 Jul '11

21 Jul '11

Hello, These 2 patches implement ASoC drivers for the AC97 controller found on early Alchemy chips. They are largely based on the old mips/au1x00.c driver which they replace. Tested on a Db1500 development board. Manuel Lauss (2): ALSA: Alchemy AC97C audio support ALSA: delete MIPS au1x00 driver arch/mips/alchemy/devboards/db1x00/platform.c | 37 ++ sound/mips/Kconfig | 8 - sound/mips/Makefile | 2 - sound/mips/au1x00.c | 695 ------------------------- sound/soc/au1x/Kconfig | 25 + sound/soc/au1x/Makefile | 9 + sound/soc/au1x/ac97c.c | 398 ++++++++++++++ sound/soc/au1x/db1000.c | 75 +++ sound/soc/au1x/dma.c | 470 +++++++++++++++++ sound/soc/au1x/psc.h | 31 +- 10 files changed, 1036 insertions(+), 714 deletions(-) delete mode 100644 sound/mips/au1x00.c create mode 100644 sound/soc/au1x/ac97c.c create mode 100644 sound/soc/au1x/db1000.c create mode 100644 sound/soc/au1x/dma.c -- 1.7.6 -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 3

[alsa-devel] [PATCH V2 0/4] Add Samsung internal dma driver support
by Sangbeom Kim 21 Jul '11

21 Jul '11

I2S in samsung audio subsytem has a small internal dma(idma). IDMA can be used in low power audio mode and I2S0 secondary fifo data tranfer. In this patchsets, idma driver is added. Original code is made by jassi brar on SMDKV210 After idma driver merge, I will submit low power audio support with idma driver and samsung dsp driver. Changes since v1: 1. Change idma_new by new pcm_new api 2. Add handling different idma address 3. As jassi comments, Modify the idma driver This patchset contains followings o To Jassi Brar, Mark Brown and Liam Girdwood, [PATCH V2 1/4] ASoC: SAMSUNG: Modify I2S driver to support idma [PATCH V2 2/4] ASoC: SAMSUNG: Add I2S0 internal dma driver o To Kukjin Kim [PATCH V2 3/4] ARM: SAMSUNG: Add platform device for idma [PATCH V2 4/4] ARM: SAMSUNG: Add IRQ_I2S0 definition Thanks SB Kim

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Re: [alsa-devel] [PATCH REPOST] sound: codec: tlv320aic3x: remove mono controls
by Mark Brown 21 Jul '11

21 Jul '11

On Fri, Jul 15, 2011 at 05:40:54PM +0200, Jan Weitzel wrote: > if codec driver is used for AIC3X_MODEL_3007 the mono iout controls overwrite > registers for class-d amplifier. > classd amplifier controls are only used for AIC3X_MODEL_3007. > Removing all mono snd_kcontrol_new, snd_soc_dapm_widget, snd_soc_dapm_route > and aic3x_init stuff from common code and call only for not AIC3X_MODEL_3007 > codecs. This is clearly not what your subject line says... > if (aic3x->model == AIC3X_MODEL_3007) { > aic3x_init_3007(codec); > snd_soc_write(codec, CLASSD_CTRL, 0); > + > + return 0; > } > + /* Not on AIC3X_MODEL_3007 init */ > + This is really not great legibility and it's going to get terribly confusing if someone comes up with some other device that also needs custom init. A similar issue applied to some of the other stuff but this is the most obvious example. I'd suggest restructuring this to have all the mono-specific code in an if statement that postively selects for the feature. > if (aic3x->model == AIC3X_MODEL_3007) > snd_soc_add_controls(codec, &aic3x_classd_amp_gain_ctrl, 1); > - > + else > + snd_soc_add_controls(codec, aic3x_mono_controls, > + ARRAY_SIZE(aic3x_mono_controls)); This if ... else should really be a switch.

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[alsa-devel] [PATCH] ASoC: dapm - Add methods to retrieve snd_card and soc_card from dapm context.
by Liam Girdwood 20 Jul '11

20 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Provide convenience methods to retrieve the soc_card or snd_card from a DAPM context. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-dapm.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 4205a97..aaef0ef 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -124,6 +124,36 @@ static inline struct snd_soc_dapm_widget *dapm_cnew_widget( return kmemdup(_widget, sizeof(*_widget), GFP_KERNEL); } +/* get snd_card from DAPM context */ +static inline struct snd_card *dapm_get_snd_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card->snd_card; + else if (dapm->platform) + return dapm->platform->card->snd_card; + else + BUG(); + + /* unreachable */ + return NULL; +} + +/* get soc_card from DAPM context */ +static inline struct snd_soc_card *dapm_get_soc_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card; + else if (dapm->platform) + return dapm->platform->card; + else + BUG(); + + /* unreachable */ + return NULL; +} + static int soc_widget_read(struct snd_soc_dapm_widget *w, int reg) { if (w->codec) -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: dapm - add DAPM macro for external enum widgets
by Liam Girdwood 20 Jul '11

20 Jul '11

Add a convenience macro for external enumerated widgets. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 6 ++++++ 1 files changed, 6 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 596387b..655acc5 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -266,6 +266,12 @@ .get = snd_soc_dapm_get_enum_virt, \ .put = snd_soc_dapm_put_enum_virt, \ .private_value = (unsigned long)&xenum } +#define SOC_DAPM_ENUM_EXT(xname, xenum, xget, xput) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ + .info = snd_soc_info_enum_double, \ + .get = xget, \ + .put = xput, \ + .private_value = (unsigned long)&xenum } #define SOC_DAPM_VALUE_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ .info = snd_soc_info_enum_double, \ -- 1.7.4.1

2 1

[alsa-devel] [PATCH v2 2/2] ASoC: davinci: add missing break statement
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

In davinci_vcif_trigger() function, a break() statement was missing causing the davinci_vcif_stop() function to be called as a fallback after calling davinci_vcif_start(). Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index c957e9e..1f11525 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -159,6 +159,7 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_vcif_start(substream); + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -- 1.7.1

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: davinci: fix codec start and stop functions
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

According to DM365 voice codec data sheet at [1], before starting recording or playback, ADC/DAC modules should follow a reset and enable cycle. Writing a 1 to the ADC/DAC bit in the register resets the module and clearing the bit to 0 will enable the module. But the driver seems to be doing the reverse of it. [1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 8 ++++---- 1 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9259f1f..c957e9e 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -62,9 +62,9 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -80,9 +80,9 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) /* Reset transmitter/receiver and sample rate/frame sync generators */ w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } -- 1.7.1

3 2

[alsa-devel] [PATCH v4 00/10] ARM: mxs: add audio support
by Dong Aisheng 20 Jul '11

20 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. For the ASoC-patches (1-4): Tested-by: Wolfram Sang <w.sang(a)pengutronix.de> For the mach-mxs-Patches (5-10): Reviewed-by: Wolfram Sang <w.sang(a)pengutronix.de> Changes since v3: This version only has one patch changed with another two for rebase the changed one. * [PATCH v4 06/10] ARM: mxs: add saif device Drop DIGCTRL-handling based on Wolfram's comment * [PATCH v4 07/10] ARM: mxs: add sgtl5000 i2c device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device * [PATCH v4 08/10] ARM: mxs: add mxs-sgtl5000 device Resend for rebase [PATCH v4 06/10] ARM: mxs: add saif device changes since v2: * mxs-pcm - fix wrong periods_max. - For robustness replace the channels_min check with a check for the relevant substream being present * mxs-saif - Add IRQ_NONE handle - improve mxs_saif_set_clk to make it more easy to read - correct return value - request_irq sequence changed to after ioremap * mxs-sgtl5000 - clean up a few unused code * Kconfig&Makefile - SND_SOC_MXS_SGTL5000 should depend on I2C in Kconfig * mxs - Select MXS_I2C in kconfig for mxs-sgtl5000 changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 2 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 35 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 677 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 165 ++++++ 18 files changed, 1530 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

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[alsa-devel] [PATCH 1/2] ASoC: Acknowledge WM8996 interrupts before acting on them
by Mark Brown 20 Jul '11

20 Jul '11

This closes the small race between a status being read in response to an interrupt and clearing the interrupt, meaning that if the status changes between those periods we might not get a reassertion of the interrupt. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8996.c | 9 ++++----- 1 files changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 00f9ace..ab8e9d1 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2404,6 +2404,8 @@ static irqreturn_t wm8996_irq(int irq, void *data) } irq_val &= ~snd_soc_read(codec, WM8996_INTERRUPT_STATUS_2_MASK); + snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); + if (irq_val & (WM8996_DCS_DONE_01_EINT | WM8996_DCS_DONE_23_EINT)) { dev_dbg(codec->dev, "DC servo IRQ\n"); complete(&wm8996->dcs_done); @@ -2420,13 +2422,10 @@ static irqreturn_t wm8996_irq(int irq, void *data) if (irq_val & WM8996_MICD_EINT) wm8996_micd(codec); - if (irq_val) { - snd_soc_write(codec, WM8996_INTERRUPT_STATUS_2, irq_val); - + if (irq_val) return IRQ_HANDLED; - } else { + else return IRQ_NONE; - } } static irqreturn_t wm8996_edge_irq(int irq, void *data) -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: davinci: bug fixes for DM365 voice codec driver
by Rajashekhara, Sudhakar 20 Jul '11

20 Jul '11

This patch fixes the following bugs found in DaVinci voice codec driver. a. According to DM365 voice codec data sheet at [1], before starting recording or playback, ADC/DAC modules should follow a reset and enable cycle. Writing a 1 to the ADC/DAC bit in the register resets the module and clearing the bit to 0 will enable the module. But the driver seems to be doing the reverse of it. b. In davinci_vcif_trigger() function, a break() statement was missing causing the davinci_vcif_stop() function to be called as a fallback after calling davinci_vcif_start(). [1] http://focus.ti.com/lit/ug/sprufi9b/sprufi9b.pdf Signed-off-by: Rajashekhara, Sudhakar <sudhakar.raj(a)ti.com> --- sound/soc/davinci/davinci-vcif.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-vcif.c b/sound/soc/davinci/davinci-vcif.c index 9259f1f..1f11525 100644 --- a/sound/soc/davinci/davinci-vcif.c +++ b/sound/soc/davinci/davinci-vcif.c @@ -62,9 +62,9 @@ static void davinci_vcif_start(struct snd_pcm_substream *substream) w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -80,9 +80,9 @@ static void davinci_vcif_stop(struct snd_pcm_substream *substream) /* Reset transmitter/receiver and sample rate/frame sync generators */ w = readl(davinci_vc->base + DAVINCI_VC_CTRL); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1); else - MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0); + MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1); writel(w, davinci_vc->base + DAVINCI_VC_CTRL); } @@ -159,6 +159,7 @@ static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: davinci_vcif_start(substream); + break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: -- 1.7.1

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[alsa-devel] [PATCH] ASoC: dapm - Add DAPM stream completion event.
by Liam Girdwood 20 Jul '11

20 Jul '11

In preparation for Dynamic PCM (AKA DSP) support. This adds a callback function to be called at the completion of a DAPM stream event. This can be used by DSP components to perform calculations based on DAPM graphs after completion of stream events. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 2 ++ include/sound/soc.h | 6 ++++++ sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 4 ++++ 4 files changed, 14 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 198997d..596387b 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -524,6 +524,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int (*stream_event)(struct snd_soc_dapm_context *dapm); + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/include/sound/soc.h b/include/sound/soc.h index aa19f5a..4c42f02 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -634,6 +634,9 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + /* codec stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; @@ -669,6 +672,9 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + /* platform stream completion event */ + int (*stream_event)(struct snd_soc_dapm_context *dapm); + /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..159f144 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3140,6 +3140,7 @@ int snd_soc_register_platform(struct device *dev, platform->driver = platform_drv; platform->dapm.dev = dev; platform->dapm.platform = platform; + platform->dapm.stream_event = platform_drv->stream_event; mutex_lock(&client_mutex); list_add(&platform->list, &platform_list); @@ -3252,6 +3253,7 @@ int snd_soc_register_codec(struct device *dev, codec->dapm.dev = dev; codec->dapm.codec = codec; codec->dapm.seq_notifier = codec_drv->seq_notifier; + codec->dapm.stream_event = codec_drv->stream_event; codec->dev = dev; codec->driver = codec_drv; codec->num_dai = num_dai; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aaef0ef..931dd7a 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2764,6 +2764,10 @@ static void soc_dapm_stream_event(struct snd_soc_dapm_context *dapm, } dapm_power_widgets(dapm, event); + + /* do we need to notify any clients that DAPM stream is complete */ + if (dapm->stream_event) + dapm->stream_event(dapm); } /** -- 1.7.4.1

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Re: [alsa-devel] [patch][saa7134] do not change mute state for capturing audio
by Stas Sergeev 20 Jul '11

20 Jul '11

15.07.2011 05:38, Mauro Carvalho Chehab wrote: > If you want, feel free to propose a patch fixing that logic at saa7134, instead > of just removing it. Hi, I've just verified that pulseaudio indeed does the sound capturing on startup: --- saa7134[0]/alsa: saa7134[0] at 0xfe8fb800 irq 22 registered as card 2 saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=1 => fmt=0xcd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: rec_start: afmt=2 ch=2 => fmt=0xdd swap=- saa7134[0]/alsa: irq: field oops [even] --- So your proposal is not going to fix anything at all. Can we get back to discussing/applying mine then? And if the other drivers has that autounmute logic, then I suggest removing it there as well. You have not named any use-case for it, so I think there is none. I also think that the whole auto-unmute logic in your drivers is entirely flawed: for instance, I don't think recording from the sound card will automatically unmute its line-in or something else, so you are probably not following the generic alsa style here. I am adding alsa-devel to CC to find out what they think about that whole auto-unmute question.

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[alsa-devel] [PATCH v3 00/10] ARM: mxs: add audio support
by Dong Aisheng 20 Jul '11

20 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. changes since v2: * mxs-pcm - fix wrong periods_max. - For robustness replace the channels_min check with a check for the relevant substream being present * mxs-saif - Add IRQ_NONE handle - improve mxs_saif_set_clk to make it more easy to read - correct return value - request_irq sequence changed to after ioremap * mxs-sgtl5000 - clean up a few unused code * Kconfig&Makefile - SND_SOC_MXS_SGTL5000 should depend on I2C in Kconfig * mxs - Select MXS_I2C in kconfig for mxs-sgtl5000 changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 2 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 40 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 359 ++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 677 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 165 ++++++ 18 files changed, 1535 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

5 22

[alsa-devel] [PATCH 0/3] make the sgtl5000-codec work
by Wolfram Sang 19 Jul '11

19 Jul '11

These are three patches I developed while trying to make the sgtl5000 work on an mx28evk with the mxs-support provided by Dong Aisheng. The chip used to lock up before, because a wrong value was written to the power control. The first patch removes code duplication The second one is for user-comfort The third one makes the device work Aisheng: This should make your hack for the sgtl5000 obsolete. Could you please test it and add your Tested-by-tag if it also works for you? (Or Reviewed-by if you want to review the code). Zeng Zhaoming: Did this code really work for you? Regards, Wolfram Wolfram Sang (3): ASoC: sgtl5000: refactor registering internal ldo ASoC: sgtl5000: guide user when regulator support is needed ASoC: sgtl5000: fix cache handling sound/soc/codecs/sgtl5000.c | 198 +++++++++++++++---------------------------- 1 files changed, 67 insertions(+), 131 deletions(-) -- 1.7.5.4

6 15

[alsa-devel] [PATCH] ASoC: core: make comments fit the code
by Wolfram Sang 19 Jul '11

19 Jul '11

In one comment, cpu_dai was mentioned although codec_dai was used in the code. Also, fix the name for the card dai list which has no seperation into card_dai and codec_dai. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d75043e..ba95444 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1631,7 +1631,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } } cpu_dai->probed = 1; - /* mark cpu_dai as probed and add to card cpu_dai list */ + /* mark cpu_dai as probed and add to card dai list */ list_add(&cpu_dai->card_list, &card->dai_dev_list); } @@ -1672,7 +1672,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } } - /* mark cpu_dai as probed and add to card cpu_dai list */ + /* mark codec_dai as probed and add to card dai list */ codec_dai->probed = 1; list_add(&codec_dai->card_list, &card->dai_dev_list); } -- 1.7.5.4

3 2

[alsa-devel] [PATCH v5 4/4] ASoC: SAMSUNG: Add I2S0 internal dma driver
by Sangbeom Kim 19 Jul '11

19 Jul '11

I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- V2: change idma_new by new pcm_new api add handling different idma address as jassi comments, Modify the idma driver V3: merge idma_reg_int and idma_addr_init V4: move idma setting in i2s.c to idma_hw_params in idma.c move idma_reg_addr_init calling modify idma address handling error modify module information delete useless dev_dbg V5: delete redundant error print and check routine modify irq handler as jassi comment fix the indent error sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 +++ 3 files changed, 481 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 00ba557..41becdd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 0000000..ebde074 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_PERIOD, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 0000000..4827321 --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + +#define MAX_IDMA_PERIOD (128 * 1024) +#define MAX_IDMA_BUFFER (160 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.1

6 11

Re: [alsa-devel] HP-path handling on VIA codecs
by alex dot baldacchino dot alsasub at gmail dot com 18 Jul '11

18 Jul '11

2011/7/8 Takashi Iwai <tiwai(a)suse.de>: > Hi, > > while testing with the gathered alsa-info.sh outputs, now I'm > considering how to improve the handling of the headphone (and speaker) > output paths. > > On VIA codecs, it looks like that the loopback mixer always contains > the input from the front DAC, and the front DAC is the only DAC that > can be mixed with loopback. (It might be not always, but on many > codecs, at least). > > I mean, some codecs of other vendors have a connection like: > Input-pin(s) -> Mix1 -> Mix2 -> Output1 > DAC1 -> Mix2 -> Output1 > Input-pin(s) -> Mix3 -> Mix4 -> Output2 > DAC2 -> Mix4 -> Output2 > .... > For such devices, you can use DAC as the volume control of each output > channel (HP, line-out, speaker, etc). So I started implementing in > that way for VIA, too. It's found as two parsed output paths, > spec->hp_out, and spec->hp_dep_out, > > But later I found that VIA has only > Input-pin(s) -> Mix1 -> Output1 > DAC1 -> Mix1 -> Output1 > or > DAC1 -> Output1 > > Input-pin(s) -> Mix1 -> Output1 > DAC1 -> Mix1 -> Output2 > or > DAC2 -> Output2 > ... > > Suppose the output2 is HP, it means that you can't have an individual > volume control via DAC2 if you want to mix the loopback. > Now we face a dilemma: usually an individual volume control is really > nice. But, if we take the individual control, it looses the loopback, > as these are mutual-exclusive for HP (or speaker) outputs. > > Thus we need to decide a policy: either > > A. HP has no volume unless independent mode > > B. Don't mix loopback to HP or others (but front-out) > > C. Allow to turn on/off aa-loopback mode; when loopback is enabled, > switch the path to hp_dep without volume control (only for front). > When loopback is disabled, switch to the direct DAC path. > > A is the old driver behavior, I guess. B is somewhat similar to now. > C would be smart, but need more implementation and add yet another > mixer element. > > What do you think? > > > thanks, > > Takashi > Personally, if I'm not misunderstanding it, I'd vote for C. I had tried something similar for old implementation, and can tell it worked and was easy enough to implement, AFAICT (I can resend the code if found of any interest, since it seems to me my first mail about it went lost, at least that's not in this mailing list archives, though that's based on the old code, and could be useful only to exchange ideas). It was quite simple: I used a few new fields in struct via_spec and two input mux controls, one for hp and one for front (to have a fine grained control and disable/enable aa path independently for those lines - and test if it could work this way), called "Direct Output" and "HP Direct Output" (I chose names thinking about the sound coming straight from analog inputs and going to analog outputs through the AA mixer, but could be, for instance, "({Front, HP}) Output Mode" (or "Output Source") with items labeled "Mixed with input" (or "Stereo mixer") and "Playback only", or the like). Corresponding controls callbacks worked on the mixer connected to each output jack and getting input from a dac (through an audio selector) and from the aa mixer, choosing the right mixer basing on the control they were invoked for, and the corresponding values in via_spec new fields: .get() returned the (negated) value of the mute bit of aa mixer connection in the chosen mixer connection list, and .put() muted/un-muted such a connection if in a different state (a better behavior would be to alternatively mute/unmute the connection to the dac and the connection to the loopback mixer, taking care of hp state (working on aa mixer connection only for hp mixer while in independent mode), and, of course, there could be a single switch/mux control to enable/disable the aa path globally). My reason to test such a solution was some noise I constantly got (and still get with newer code, but for Independent HP in latest version) from my headset microphone to each such line out (front and hp), but I can think, for instance, of someone wishing to listen to something while recording something else from line-in and not aiming to mix both sounds in output. But I guess some problems could arise in actual implementation, unless I'm overvaluing/misunderstanding them. First of all, now there's a pair of PCM Playback controls (switch and volume) attached to aa mixer and I'm not sure if muting connections to it could affect those controls (specially the volume). In such a case, they could be moved to the front dac, but this problem, if existing, could be re-raised for hp path in independent mode (in which case front dac and attached controls would be out of the game). An alternative could be leaving those controls where they're now, but reimplementing callbacks so that they could use a different nid (and work on more than one nid at a time). This way: - the .put callback for turning aa-loopback on/off for a certain nid (or all affected nids at once, if only one such control existed) would be responsible: 1. to record somewhere (accessible by via_spec) the nid(s) to work on (quite easy with separate controls attached to each affected mixer: the nid to gather would be just the actual one being unmuted in the two-entries connection list of passed-in kcontrol nid or the non loopback connection for hp mixer in independent mode), 2. to set the right volume (not just un-muting) for the non-loopback path(s) when switching to it (basing on pcm volume setting) and muting/unmuting the connection(s) to aa mixer as needed; - the .put callback for adjusting pcm volume setting would work as now for aa mixer nid (both to let it work as expected when used as source for output and to gather correct values from it, but also for correct handling of capture from stereo mixer), while performing same operations for each alternative nid, gathered from a proper list and being non-null and different from aa mixer nid; - the .get callback for "PCM Playback Volume" could be left untouched (to always gather infos from the 'base' nid the control is attacthed to). Similar considerations could/should be done for the pcm switch, that (now, in my case) has the effect to mute/unmute front output (only). However, I might be missing the point since I'm not sure how those controls are expected to work properly. Previously, with old code, I had no PCM Playback Switch and a "standalone" PCM Playback Volume, detached from any nid, and such seemed to work for all outputs, as far as I remember (I should re-install old implementation to make a test); alsa-info.sh reported following output for alsactl: control.31 { iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 comment { access 'read write user' type INTEGER count 2 range '0 - 255' tlv '0000000100000008ffffec1400000014' dbmin -5100 dbmax 0 dbvalue.0 0 dbvalue.1 0 } } Now I have both controls attached to 0x21, they seem to have effect on front path only (throughout aa mixer - previously its connection to front dac, for my codec, was just un-muted with an initialization verb) and alsa-info.sh reports: control.9 { iface MIXER name 'PCM Playback Volume' value.0 22 value.1 22 comment { access 'read write' type INTEGER count 2 range '0 - 31' dbmin -3450 dbmax 1200 dbvalue.0 -150 dbvalue.1 -150 } } control.10 { iface MIXER name 'PCM Playback Switch' value.0 true value.1 true comment { access 'read write' type BOOLEAN count 2 } } Similar concerns could be raised for Master Playback controls, given some codecs (but not mine - vt2020) holds them via their aa mixer nid. Another, possibly minor, problem, as pointed out in Lydia's answer, could be the somewhat inconsistence introduced between newer codecs supporting solution C and older ones unable to mute their aa mixer without loosing their front dac (because their aa mixer is just between their front dac and their rear green jack, but also between front dac and hp jack when in redirected mode). Such could be simply disregarded, taken as a consequence of a different set of supported features; therefore, it should be possible to choose automatically whether to create the proper control(s) for turning on/off aa-loopback or not while parsing output paths, looking at the connection lists of nids in a certain path, starting from an output jack and moving towards a dac, and looking if an element (a mixer or the jack nid itself) is found with at least two connections, one of which should be the aa mixer nid and the other one either the front dac nid, or an audio selector connected to it. For hp path the non-loopback connection shouldn't be hp dac directly (not passing through a selector), or the implementation would conflict with redirected output if such a dac were shared, which is the case for several codecs, specially older ones (newer ones seem to use an audio selector, instead, for front and hp), though I'm not sure if that's the same for all such codecs - hp_indep_shared could tell that, so that a few older codecs could support solution C for hp only, but in such a case solution C would behave mostly as B in current implementation, but for redirected mode, given front stream would be 'cloned' for hp dac if not shared, so hp could get loopback mixed in or not at whim, even if such would be inconsistent with front being always mixed with loopback). Regards, Alex

2 5

Re: [alsa-devel] model needed for Clevo P150HM (Realtek ALC892)
by Takashi Iwai 18 Jul '11

18 Jul '11

At Thu, 30 Jun 2011 08:30:10 +0200, Julian Sikorski wrote: > > W dniu 30.06.2011 07:26, Takashi Iwai pisze: > > At Wed, 29 Jun 2011 21:29:38 +0200, > > Julian Sikorski wrote: > >> > >> I also tried merging sound-2.6/master with Linus's tree, but this also > >> did not help. > > > > It's really strange. Are you sure that this is the kernel you > > modified properly? Before further debugging, check two things: > > > > - Run "git log sound/pci/hda/patch_realtek.c" and the first few > > entries, whether they come from sound git tree. For example, > > check the commit below is contained: > > > > commit 7ec9c6ccc6007b14a916021d4ba7ffbcc7822ae3 > > Author: Takashi Iwai <tiwai(a)suse.de> > > Date: Mon Jun 27 15:53:38 2011 +0200 > > > > ALSA: hda - Fix volume-init for ALC259 with invalid widget caps > > commit 0b0c8490670a77b937f502491ae9d5236aa4054a > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Tue Jun 28 12:45:47 2011 +0200 > > ALSA: hda - Create snd_hda_get_conn_index() helper function > > Create snd_hda_get_conn_index() helper function for obtaining the > connection index of the widget. Replaced the similar codes used in > several codec-drivers with this common helper. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > commit 4f574b7b1a1cc8aac617e938459e8f03a641e678 > Author: Takashi Iwai <tiwai(a)suse.de> > Date: Mon Jun 27 16:17:07 2011 +0200 > > ALSA: hda - More volume-init fixes for ALC267 codec > > More similar fixes like previous commits: handle the exceptional case > like ALC267 where no volume amp is found in ADC widget but in the > capsrc widget instead. > > Also minor checks for avoiding possible erros: no connection-select > when the pin has a single selection, and add beep verbs only when the > 0x1d is used for beep. > > Signed-off-by: Takashi Iwai <tiwai(a)suse.de> > > To be even more sure, I re-synced your branch again and built a new > kernel. I am attaching patch_realtek.c file being built. Fresh > alsa-info.sh is here: > http://www.alsa-project.org/db/?f=5051b2cb788b1db4273e06e98a78783dd0ec679f OK, then it's correct. > > > > - Confirm the running kernel is what you really built and installed. > > You can modify the suffix via $CONFIG_LOCALVERSION in .config file. > > Name it uniquely, rebuild, re-install and reboot. The new suffix > > should appear in the output of "uname -r". > > belegdol4 is the suffix I added: > $ uname -r > 3.0-0.rc5.git0.1.belegdol4.fc15.x86_64 > > > - Once when these are correct, check /proc/asound/card0/codec#0: > > the widget NID 0x02 should contain "Speaker Playback Volume" instead > > of "Headphone Playback Volume" with the latest driver. > > Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out > Control: name="Headphone Playback Volume", index=0, device=0 > > Perhaps I need to specify a correct model to enable this code? No, you shouldn't pass any model option. Could you apply the patch below to add some debug prints, and give the relevant kernel messages you get? Just grep lines with 'XXX'. Takashi --- diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7b96dce..3134555 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -18770,8 +18770,10 @@ static hda_nid_t alc_auto_look_for_dac(struct hda_codec *codec, hda_nid_t pin) static hda_nid_t get_dac_if_single(struct hda_codec *codec, hda_nid_t pin) { hda_nid_t sel = alc_go_down_to_selector(codec, pin); - if (snd_hda_get_conn_list(codec, sel, NULL) == 1) + if (snd_hda_get_conn_list(codec, sel, NULL) == 1) { + printk(KERN_DEBUG "XXX pin %x coupled with a single DAC\n", pin); return alc_auto_look_for_dac(codec, pin); + } return 0; } @@ -18783,6 +18785,7 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) bool redone; int i; + printk(KERN_DEBUG "XXX alc662_auto_fill_dac_nids called\n"); again: spec->multiout.num_dacs = 0; spec->multiout.hp_nid = 0; @@ -18813,6 +18816,7 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) * checking the hard-wired DACs */ redone = true; + printk(KERN_DEBUG "XXX redoing...\n"); goto again; } } @@ -18826,6 +18830,17 @@ static int alc662_auto_fill_dac_nids(struct hda_codec *codec) sizeof(hda_nid_t) * (cfg->line_outs - i - 1)); } + printk(KERN_DEBUG "XXX filled dacs: %x/%x/%x/%x/%x, hp %x, ext %x/%x/%x\n", + spec->private_dac_nids[0], + spec->private_dac_nids[1], + spec->private_dac_nids[2], + spec->private_dac_nids[3], + spec->private_dac_nids[4], + spec->multiout.hp_nid, + spec->multiout.extra_out_nid[0], + spec->multiout.extra_out_nid[1], + spec->multiout.extra_out_nid[2]); + return 0; } @@ -18877,6 +18892,7 @@ static int alc662_auto_create_multi_out_ctls(struct hda_codec *codec, mix = alc_auto_dac_to_mix(codec, pin, nid); if (!mix) continue; + printk(KERN_DEBUG "XXX create mixer ch %d pin %x dac %x mix %x\n", i, pin, nid, mix); name = alc_get_line_out_pfx(spec, i, true, &index); if (!name) { /* Center/LFE */ @@ -19026,6 +19042,7 @@ static int alc_auto_fill_multi_ios(struct hda_codec *codec, dac = alc_auto_look_for_dac(codec, nid); if (!dac) continue; + printk(KERN_DEBUG "XXX added multi-io pin %x, dac %x\n", nid, dac); spec->multi_io[num_pins].pin = nid; spec->multi_io[num_pins].dac = dac; num_pins++; @@ -19128,6 +19145,7 @@ static int alc_auto_add_multi_channel_mode(struct hda_codec *codec, if (cfg->line_out_type == AUTO_PIN_SPEAKER_OUT && cfg->hp_outs == 1) { /* use HP as primary out */ + printk(KERN_DEBUG "XXX swap primary out to HP\n"); cfg->speaker_outs = cfg->line_outs; memcpy(cfg->speaker_pins, cfg->line_out_pins, sizeof(cfg->speaker_pins));

3 37

[alsa-devel] [PATCH] ASoC: Mark cache as dirty when suspending
by Mark Brown 18 Jul '11

18 Jul '11

Since quite a few drivers are not managing to flag the cache as needing to be resynced after suspend and it's a reasonable thing to do flag the cache as needing sync automatically when suspending. The expectation is that systems will mainly only keep the CODEC powered when doing audio through the CODEC so we won't actually suspend the device anyway; drivers which want to can override this behaviour when they resume. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e44267f..93109a4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -577,6 +577,7 @@ int snd_soc_suspend(struct device *dev) case SND_SOC_BIAS_OFF: codec->driver->suspend(codec, PMSG_SUSPEND); codec->suspended = 1; + codec->cache_sync = 1; break; default: dev_dbg(codec->dev, "CODEC is on over suspend\n"); -- 1.7.5.4

2 1

[alsa-devel] [RESEND][PATCH v5 4/4] ASoC: SAMSUNG: Add I2S0 internal dma driver
by Sangbeom Kim 18 Jul '11

18 Jul '11

I2S in Exynos4 and S5PC110(S5PV210) has a internal dma. It can be used low power audio mode and 2nd channel transfer. This patch can support idma. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- V2: change idma_new by new pcm_new api add handling different idma address as jassi comments, Modify the idma driver V3: merge idma_reg_int and idma_addr_init V4: move idma setting in i2s.c to idma_hw_params in idma.c move idma_reg_addr_init calling modify idma address handling error modify module information delete useless dev_dbg V5: delete redundant error print and check routine modify irq handler as jassi comment fix the indent error modify period_bytes_max value to set half size of max buffer sound/soc/samsung/Makefile | 2 + sound/soc/samsung/idma.c | 453 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/idma.h | 26 +++ 3 files changed, 481 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/idma.c create mode 100644 sound/soc/samsung/idma.h diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 00ba557..41becdd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -1,5 +1,6 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := dma.o +snd-soc-idma-objs := idma.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-ac97-objs := ac97.o @@ -16,6 +17,7 @@ obj-$(CONFIG_SND_S3C_I2SV2_SOC) += snd-soc-s3c-i2s-v2.o obj-$(CONFIG_SND_SAMSUNG_SPDIF) += snd-soc-samsung-spdif.o obj-$(CONFIG_SND_SAMSUNG_PCM) += snd-soc-pcm.o obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-i2s.o +obj-$(CONFIG_SND_SAMSUNG_I2S) += snd-soc-idma.o # S3C24XX Machine Support snd-soc-jive-wm8750-objs := jive_wm8750.o diff --git a/sound/soc/samsung/idma.c b/sound/soc/samsung/idma.c new file mode 100644 index 0000000..ebde074 --- /dev/null +++ b/sound/soc/samsung/idma.c @@ -0,0 +1,453 @@ +/* + * sound/soc/samsung/idma.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * I2S0's Internal DMA driver + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include <linux/interrupt.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <linux/slab.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "i2s.h" +#include "idma.h" +#include "dma.h" +#include "i2s-regs.h" + +#define ST_RUNNING (1<<0) +#define ST_OPENED (1<<1) + +static const struct snd_pcm_hardware idma_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_U16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_U24_LE | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S8, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = MAX_IDMA_BUFFER, + .period_bytes_min = 128, + .period_bytes_max = MAX_IDMA_BUFFER/2, + .periods_min = 1, + .periods_max = 2, +}; + +struct idma_ctrl { + spinlock_t lock; + int state; + dma_addr_t start; + dma_addr_t pos; + dma_addr_t end; + dma_addr_t period; + dma_addr_t periodsz; + void *token; + void (*cb)(void *dt, int bytes_xfer); +}; + +static struct idma_info { + spinlock_t lock; + void __iomem *regs; + dma_addr_t lp_tx_addr; +} idma; + +static void idma_getpos(dma_addr_t *src) +{ + *src = idma.lp_tx_addr + + (readl(idma.regs + I2STRNCNT) & 0xffffff) * 4; +} + +static int idma_enqueue(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 val; + + spin_lock(&prtd->lock); + prtd->token = (void *) substream; + spin_unlock(&prtd->lock); + + /* Internal DMA Level0 Interrupt Address */ + val = idma.lp_tx_addr + prtd->periodsz; + writel(val, idma.regs + I2SLVL0ADDR); + + /* Start address0 of I2S internal DMA operation. */ + val = idma.lp_tx_addr; + writel(val, idma.regs + I2SSTR0); + + /* + * Transfer block size for I2S internal DMA. + * Should decide transfer size before start dma operation + */ + val = readl(idma.regs + I2SSIZE); + val &= ~(I2SSIZE_TRNMSK << I2SSIZE_SHIFT); + val |= (((runtime->dma_bytes >> 2) & + I2SSIZE_TRNMSK) << I2SSIZE_SHIFT); + writel(val, idma.regs + I2SSIZE); + + val = readl(idma.regs + I2SAHB); + val |= AHB_INTENLVL0; + writel(val, idma.regs + I2SAHB); + + return 0; +} + +static void idma_setcallbk(struct snd_pcm_substream *substream, + void (*cb)(void *, int)) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + spin_lock(&prtd->lock); + prtd->cb = cb; + spin_unlock(&prtd->lock); +} + +static void idma_control(int op) +{ + u32 val = readl(idma.regs + I2SAHB); + + spin_lock(&idma.lock); + + switch (op) { + case LPAM_DMA_START: + val |= (AHB_INTENLVL0 | AHB_DMAEN); + break; + case LPAM_DMA_STOP: + val &= ~(AHB_INTENLVL0 | AHB_DMAEN); + break; + default: + spin_unlock(&idma.lock); + return; + } + + writel(val, idma.regs + I2SAHB); + spin_unlock(&idma.lock); +} + +static void idma_done(void *id, int bytes_xfer) +{ + struct snd_pcm_substream *substream = id; + struct idma_ctrl *prtd = substream->runtime->private_data; + + if (prtd && (prtd->state & ST_RUNNING)) + snd_pcm_period_elapsed(substream); +} + +static int idma_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = substream->runtime->private_data; + u32 mod = readl(idma.regs + I2SMOD); + u32 ahb = readl(idma.regs + I2SAHB); + + ahb |= (AHB_DMARLD | AHB_INTMASK); + mod |= MOD_TXS_IDMA; + writel(ahb, idma.regs + I2SAHB); + writel(mod, idma.regs + I2SMOD); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + runtime->dma_bytes = params_buffer_bytes(params); + + prtd->start = prtd->pos = runtime->dma_addr; + prtd->period = params_periods(params); + prtd->periodsz = params_period_bytes(params); + prtd->end = runtime->dma_addr + runtime->dma_bytes; + + idma_setcallbk(substream, idma_done); + + return 0; +} + +static int idma_hw_free(struct snd_pcm_substream *substream) +{ + snd_pcm_set_runtime_buffer(substream, NULL); + + return 0; +} + +static int idma_prepare(struct snd_pcm_substream *substream) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + + prtd->pos = prtd->start; + + /* flush the DMA channel */ + idma_control(LPAM_DMA_STOP); + idma_enqueue(substream); + + return 0; +} + +static int idma_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct idma_ctrl *prtd = substream->runtime->private_data; + int ret = 0; + + spin_lock(&prtd->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + prtd->state |= ST_RUNNING; + idma_control(LPAM_DMA_START); + break; + + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + prtd->state &= ~ST_RUNNING; + idma_control(LPAM_DMA_STOP); + break; + + default: + ret = -EINVAL; + break; + } + + spin_unlock(&prtd->lock); + + return ret; +} + +static snd_pcm_uframes_t + idma_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + dma_addr_t src; + unsigned long res; + + spin_lock(&prtd->lock); + + idma_getpos(&src); + res = src - prtd->start; + + spin_unlock(&prtd->lock); + + return bytes_to_frames(substream->runtime, res); +} + +static int idma_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long size, offset; + int ret; + + /* From snd_pcm_lib_mmap_iomem */ + vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot); + vma->vm_flags |= VM_IO; + size = vma->vm_end - vma->vm_start; + offset = vma->vm_pgoff << PAGE_SHIFT; + ret = io_remap_pfn_range(vma, vma->vm_start, + (runtime->dma_addr + offset) >> PAGE_SHIFT, + size, vma->vm_page_prot); + + return ret; +} + +static irqreturn_t iis_irq(int irqno, void *dev_id) +{ + struct idma_ctrl *prtd = (struct idma_ctrl *)dev_id; + u32 iiscon, iisahb, val, addr; + + iisahb = readl(idma.regs + I2SAHB); + iiscon = readl(idma.regs + I2SCON); + + val = (iisahb & AHB_LVL0INT) ? AHB_CLRLVL0INT : 0; + + if (val) { + iisahb |= val; + writel(iisahb, idma.regs + I2SAHB); + + addr = readl(idma.regs + I2SLVL0ADDR) - idma.lp_tx_addr; + addr += prtd->periodsz; + addr %= (prtd->end - prtd->start); + addr += idma.lp_tx_addr; + + writel(addr, idma.regs + I2SLVL0ADDR); + + if (prtd->cb) + prtd->cb(prtd->token, prtd->period); + } + + return IRQ_HANDLED; +} + +static int idma_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd; + int ret; + + snd_soc_set_runtime_hwparams(substream, &idma_hardware); + + prtd = kzalloc(sizeof(struct idma_ctrl), GFP_KERNEL); + if (prtd == NULL) + return -ENOMEM; + + ret = request_irq(IRQ_I2S0, iis_irq, 0, "i2s", prtd); + if (ret < 0) { + pr_err("fail to claim i2s irq , ret = %d\n", ret); + kfree(prtd); + return ret; + } + + spin_lock_init(&prtd->lock); + + runtime->private_data = prtd; + + return 0; +} + +static int idma_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct idma_ctrl *prtd = runtime->private_data; + + free_irq(IRQ_I2S0, prtd); + + if (!prtd) + pr_err("idma_close called with prtd == NULL\n"); + + kfree(prtd); + + return 0; +} + +static struct snd_pcm_ops idma_ops = { + .open = idma_open, + .close = idma_close, + .ioctl = snd_pcm_lib_ioctl, + .trigger = idma_trigger, + .pointer = idma_pointer, + .mmap = idma_mmap, + .hw_params = idma_hw_params, + .hw_free = idma_hw_free, + .prepare = idma_prepare, +}; + +static void idma_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + + substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + if (!substream) + return; + + buf = &substream->dma_buffer; + if (!buf->area) + return; + + iounmap(buf->area); + + buf->area = NULL; + buf->addr = 0; +} + +static int preallocate_idma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + + /* Assign PCM buffer pointers */ + buf->dev.type = SNDRV_DMA_TYPE_CONTINUOUS; + buf->addr = idma.lp_tx_addr; + buf->bytes = idma_hardware.buffer_bytes_max; + buf->area = (unsigned char *)ioremap(buf->addr, buf->bytes); + + return 0; +} + +static u64 idma_mask = DMA_BIT_MASK(32); + +static int idma_new(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &idma_mask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + + if (dai->driver->playback.channels_min) + ret = preallocate_idma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + + return ret; +} + +void idma_reg_addr_init(void *regs, dma_addr_t addr) +{ + spin_lock_init(&idma.lock); + idma.regs = regs; + idma.lp_tx_addr = addr; +} + +struct snd_soc_platform_driver asoc_idma_platform = { + .ops = &idma_ops, + .pcm_new = idma_new, + .pcm_free = idma_free, +}; + +static int __devinit asoc_idma_platform_probe(struct platform_device *pdev) +{ + return snd_soc_register_platform(&pdev->dev, &asoc_idma_platform); +} + +static int __devexit asoc_idma_platform_remove(struct platform_device *pdev) +{ + snd_soc_unregister_platform(&pdev->dev); + return 0; +} + +static struct platform_driver asoc_idma_driver = { + .driver = { + .name = "samsung-idma", + .owner = THIS_MODULE, + }, + + .probe = asoc_idma_platform_probe, + .remove = __devexit_p(asoc_idma_platform_remove), +}; + +static int __init asoc_idma_init(void) +{ + return platform_driver_register(&asoc_idma_driver); +} +module_init(asoc_idma_init); + +static void __exit asoc_idma_exit(void) +{ + platform_driver_unregister(&asoc_idma_driver); +} +module_exit(asoc_idma_exit); + +MODULE_AUTHOR("Jaswinder Singh, <jassisinghbrar(a)gmail.com>"); +MODULE_DESCRIPTION("Samsung ASoC IDMA Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/idma.h b/sound/soc/samsung/idma.h new file mode 100644 index 0000000..4827321 --- /dev/null +++ b/sound/soc/samsung/idma.h @@ -0,0 +1,26 @@ +/* + * sound/soc/samsung/idma.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef __SND_SOC_SAMSUNG_IDMA_H_ +#define __SND_SOC_SAMSUNG_IDMA_H_ + +extern void idma_reg_addr_init(void *regs, dma_addr_t addr); + +/* dma_state */ +#define LPAM_DMA_STOP 0 +#define LPAM_DMA_START 1 + + +#define MAX_IDMA_BUFFER (128 * 1024) + +#endif /* __SND_SOC_SAMSUNG_IDMA_H_ */ -- 1.7.1

2 3

[alsa-devel] ASoC updates for 3.1
by Mark Brown 18 Jul '11

18 Jul '11

The following changes since commit e320bc42be746b8f006d163b6e340e5fb79ce7b2: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-09 11:43:04 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Axel Lin (1): ASoC: wm8900: fix a memory leak if wm8900_set_fll fails Dimitris Papastamos (1): ASoC: WM8983: Initial driver Giridhar Maruthy (1): ASoC: SAMSUNG: 24-bit audio playback on Exynos4210 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Johannes Stezenbach (2): ASoC: STA32x: Add mixer controls for biquad coefficients ASoC: STA32x: Preserve reserved register bits Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (19): Merge branch 'for-3.0' into for-3.1 Merge branch 'for-3.0' into for-3.1 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Conditionalize the enable of WM8994 ADC TDM mode ASoC: Use late enable handling for direct voice, speaker and headphone ASoC: Implement DC servo completion IRQ handling for wm_hubs devices ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958 ASoC: Use WM8994 FLL lock interrupt ASoC: Don't warn on low WM8994/58 AIFnCLKs ASoC: Log WM8994 FIFO errors from the interrupt ASoC: Fix shift in WM8958 accessory detection default implementation Merge branch 'for-3.0' into for-3.1 ASoC: Handle spurious wm_hubs DC servo done interrupts ASoC: Handle failed WM8994 FLL lock waits ASoC: Report an error for unknown adav80x formats ASoC: Reduce power consumption for idle DAIs in WM8994 ASoC: Don't use -1 to boostrap subseq so it can be used by drivers ASoC: Correct WM8994 MICBIAS supply widget hookup Merge branch 'for-3.0' into for-3.1 Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration Takashi Iwai (4): Merge branch 'for-3.0' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc ASoC: Don't set invalid name string to snd_card->driver field Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 4 +- sound/soc/codecs/sta32x.c | 140 +++++ sound/soc/codecs/tlv320aic26.c | 14 +- sound/soc/codecs/tlv320aic3x.c | 9 +- sound/soc/codecs/wm8900.c | 1 + sound/soc/codecs/wm8983.c | 1203 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8983.h | 1029 +++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 156 +++++- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 33 +- sound/soc/codecs/wm_hubs.h | 8 + sound/soc/samsung/smdk_wm8994.c | 5 +- sound/soc/sh/fsi-ak4642.c | 16 +- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 +- sound/soc/soc-core.c | 5 +- sound/soc/soc-dapm.c | 2 +- 20 files changed, 2593 insertions(+), 49 deletions(-) create mode 100644 sound/soc/codecs/wm8983.c create mode 100644 sound/soc/codecs/wm8983.h

2 1

[alsa-devel] ASoC updates for 3.0
by Mark Brown 18 Jul '11

18 Jul '11

The following changes since commit 18361bbe3180eca62796188d62aefac1519f4c83: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-07-09 09:44:09 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (2): ASoC: Fix shift in WM8958 accessory detection default implementation ASoC: Correct WM8994 MICBIAS supply widget hookup sound/soc/codecs/wm8994.c | 9 +++++---- sound/soc/sh/fsi-ak4642.c | 16 ++++++++-------- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 ++-- 4 files changed, 16 insertions(+), 15 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: Report an error for unknown adav80x formats
by Mark Brown 16 Jul '11

16 Jul '11

Not only fixes error handling but also some uninitialized variable warnings. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/adav80x.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index e30fba6..300c04b 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -456,7 +456,7 @@ static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, val = ADAV80X_CAPTURE_WORD_LEN24; break; default: - break; + return -EINVAL; } snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], @@ -488,7 +488,7 @@ static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; break; default: - break; + return -EINVAL; } snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], -- 1.7.5.4

2 2

[alsa-devel] [PATCH] ASoC: Don't use -1 to boostrap subseq so it can be used by drivers
by Mark Brown 16 Jul '11

16 Jul '11

Makes life a little easier if you want to add subsequences to an existing driver as you can use -1 to put things at the start of sequences. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-dapm.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 54fa2e5..fbfcda0 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -989,7 +989,7 @@ static void dapm_seq_run(struct snd_soc_dapm_context *dapm, INIT_LIST_HEAD(&pending); cur_sort = -1; - cur_subseq = -1; + cur_subseq = INT_MIN; cur_reg = SND_SOC_NOPM; cur_dapm = NULL; } -- 1.7.5.4

1 0

[alsa-devel] [PATCH V2 00/10]
by Dong Aisheng 15 Jul '11

15 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work as a master due to hw limitation(rx can work in both master & slave mode). Code is written based on the branch imx-for-3.1 in sascha's tree http://git.pengutronix.de/git/imx/linux-2.6.git Tested on MX28EVK. Main changes since v1: * export saif mclk control interface to machine driver. Machine driver can decide whether it needs to use mclk supplied by saif according to its requirement. * error checking improvement * remove a few unneccesary code Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 1 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 3 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 ++ arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 39 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 19 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 367 +++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 656 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 130 +++++ sound/soc/mxs/mxs-sgtl5000.c | 175 ++++++ 18 files changed, 1529 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

7 51

Re: [alsa-devel] [PATCH] ASoC: WM8983: Initial driver
by Mark Brown 15 Jul '11

15 Jul '11

On Fri, Jul 15, 2011 at 01:51:30PM +0100, Dimitris Papastamos wrote: > The WM8983 is a low power, high quality stereo CODEC > designed for portable multimedia applications. Highly flexible > analogue mixing functions enable new application features, > combining hi-fi quality audio with voice communication. I've applied this but there's one issue you should fix up later: > + /* mute all outputs and set PGAs to minimum gain */ > + for (i = WM8983_LOUT1_HP_VOLUME_CTRL; > + i <= WM8983_OUT4_MONO_MIX_CTRL; ++i) > + snd_soc_update_bits(codec, i, 0x40, 0x40); I know you're doing this to minimise artifacts on initial bias raise but we should really be doing this without affecting the default register settings.

2 2

[alsa-devel] [PATCH] ASoC: Fix shift in WM8958 accessory detection default implementation
by Mark Brown 15 Jul '11

15 Jul '11

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 bc62b6e..bfb7e66 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2834,7 +2834,7 @@ static void wm8958_default_micdet(u16 status, void *data) report = SND_JACK_MICROPHONE; /* Everything else is buttons; just assign slots */ - if (status & 0x1c0) + if (status & 0x1c) report |= SND_JACK_BTN_0; done: -- 1.7.5.4

1 0

Re: [alsa-devel] [PATCH] sound: codec: tlv320aic3x: remove mono controls
by Mark Brown 15 Jul '11

15 Jul '11

On Fri, Jul 15, 2011 at 12:59:15PM +0200, Jan Weitzel wrote: > Any rework needed? It'd be easier to review the patch if you sent me a copy...

1 0

[alsa-devel] [PATCH] ASoC: Handle spurious wm_hubs DC servo done interrupts
by Mark Brown 15 Jul '11

15 Jul '11

Don't assume the first fire indicates that we're done. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 30 ++++++++++++++++-------------- 1 files changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index ced41e8..4de6794 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -66,8 +66,8 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); unsigned int reg; int count = 0; + int timeout; unsigned int val; - unsigned long timeout; val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; @@ -76,21 +76,23 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) dev_dbg(codec->dev, "Waiting for DC servo...\n"); - if (hubs->dcs_done_irq) { - timeout = wait_for_completion_timeout(&hubs->dcs_done, - msecs_to_jiffies(500)); - if (timeout == 0) - dev_warn(codec->dev, "No DC servo interrupt\n"); + if (hubs->dcs_done_irq) + timeout = 4; + else + timeout = 400; - reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - } else { - do { - count++; + do { + count++; + + if (hubs->dcs_done_irq) + wait_for_completion_timeout(&hubs->dcs_done, + msecs_to_jiffies(250)); + else msleep(1); - reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & op && count < 400); - } + + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); + dev_dbg(codec->dev, "DC servo: %x\n", reg); + } while (reg & op && count < timeout); if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", -- 1.7.5.4

1 0

[alsa-devel] Major reworks on patch_via.c (TESTERS WANTED)
by Takashi Iwai 15 Jul '11

15 Jul '11

Hi, as there have many problems reported for VIA codecs, I started looking at the driver code, and decided to rework on it. The current development is found in sound git tree topic/via-cleanup branch: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git You can pull into the latest Linus tree from topic/via-cleanup branch: # this is the vanilla tree % cd linux-2.6 % git pull \ git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git \ topic/via-cleanup A good thing is that a fair amount of code reduction is done there: % git diff HEAD..topic/via-cleanup --stat sound/pci/hda/patch_via.c sound/pci/hda/patch_via.c | 5892 +++++++++++---------------------------------- 1 files changed, 1453 insertions(+), 4439 deletions(-) But, the driver isn't tested with real machines, yet. That is, I need testers with real machines. Could VIA machine owners take a look at these commits and test it if possible? Especially, I'd like to know whether the headphone auto-mute works in this form or not. FWIW, the branch was merged to sound-unstable tree, so you can build from alsa-driver-unstable snapshot tarball if you'd like to build external modules without the kernel tree and git-merge. ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… Also, if anyone has alsa-info.sh outputs of VIA codecs, please let me know. It'd be very helpful for debugging, since I can check it via hda-emu with alsa-info.sh output. For now, I have the files of only VT1702, VT1708S, VT1718S, VT1828S and VT2020 codecs. The alsa-info.sh outputs of other VIA codecs would be greatly appreciated. thanks, Takashi

5 18

[alsa-devel] [PATCH] ALSA: intel8x0: Apply headphones+mute LED quirk for Dell Inspiron 9300
by Dan Chen 15 Jul '11

15 Jul '11

Hi, Attached is a patch against sound-2.6 master HEAD resolving the original reporter's volume keys issue. Please consider applying. Cheers, -Dan

2 1

[alsa-devel] [PATCH 11/20] ARM i.MX dma: Fix burstsize settings
by Sascha Hauer 14 Jul '11

14 Jul '11

From: Wolfram Sang <w.sang(a)pengutronix.de> Already upstreamed. dmaengine expects the maxburst parameter in words, not bytes. The imxdma driver and its users do this wrong. Fix this. As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work with imx-dma. This broke the driver with imx-sdma support which correctly takes the maxburst parameter in words. This patch puts the sdma based sound back to work. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Cc: Vinod Koul <vinod.koul(a)intel.com> Cc: Chris Ball <cjb(a)laptop.org> Cc: linux-mmc(a)vger.kernel.org Cc: Liam Girdwood <lrg(a)ti.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: alsa-devel(a)alsa-project.org --- drivers/dma/imx-dma.c | 3 ++- sound/soc/imx/imx-pcm-dma-mx2.c | 4 ++-- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/drivers/dma/imx-dma.c b/drivers/dma/imx-dma.c index e18eaab..d99f71c 100644 --- a/drivers/dma/imx-dma.c +++ b/drivers/dma/imx-dma.c @@ -135,7 +135,8 @@ static int imxdma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, if (ret) return ret; - imx_dma_config_burstlen(imxdmac->imxdma_channel, imxdmac->watermark_level); + imx_dma_config_burstlen(imxdmac->imxdma_channel, + imxdmac->watermark_level * imxdmac->word_size); return 0; default: diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 4173b3d..43fdc24f 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize * buswidth; + slave_config.dst_maxburst = dma_params->burstsize; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize * buswidth; + slave_config.src_maxburst = dma_params->burstsize; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); -- 1.7.5.4

1 1

[alsa-devel] [PATCH] ALSA: hda - Disable automute feature by default
by Ben Hutchings 14 Jul '11

14 Jul '11

The default behaviour should be the same as before this feature was introduced, to avoid user confusion. Reported-by: Toni Mueller <support(a)oeko.net> References: http://bugs.debian.org/631963 Signed-off-by: Ben Hutchings <ben(a)decadent.org.uk> --- The code is rather inconsistent in whether 'mode' means the behaviour or mechanism of automute. I didn't touch that but you might want to consider some renaming. Ben. sound/pci/hda/patch_realtek.c | 74 ++--------------------------------------- 1 files changed, 3 insertions(+), 71 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b48fb43..1247a04 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -306,7 +306,7 @@ struct alc_multi_io { }; enum { - ALC_AUTOMUTE_PIN, /* change the pin control */ + ALC_AUTOMUTE_PIN = 1, /* change the pin control */ ALC_AUTOMUTE_AMP, /* mute/unmute the pin AMP */ ALC_AUTOMUTE_MIXER, /* mute/unmute mixer widget AMP */ }; @@ -1599,10 +1599,9 @@ static void alc_init_auto_hp(struct hda_codec *codec) snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | ALC880_HP_EVENT); - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } - if (spec->automute && cfg->line_out_pins[0] && + if (spec->automute_mode && cfg->line_out_pins[0] && cfg->line_out_pins[0] != cfg->hp_pins[0] && cfg->line_out_pins[0] != cfg->speaker_pins[0]) { for (i = 0; i < cfg->line_outs; i++) { @@ -1619,7 +1618,7 @@ static void alc_init_auto_hp(struct hda_codec *codec) spec->automute_lines = spec->detect_line; } - if (spec->automute) { + if (spec->automute_mode) { /* create a control for automute mode */ alc_add_automute_mode_enum(codec); spec->unsol_event = alc_sku_unsol_event; @@ -2241,7 +2240,6 @@ static void alc889_automute_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x17; spec->autocfg.speaker_pins[3] = 0x19; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2259,7 +2257,6 @@ static void alc888_fujitsu_xa3530_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1b; /* hp */ spec->autocfg.speaker_pins[0] = 0x14; /* speaker */ spec->autocfg.speaker_pins[1] = 0x15; /* bass */ - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2543,7 +2540,6 @@ static void alc888_acer_aspire_4930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2555,7 +2551,6 @@ static void alc888_acer_aspire_6530g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2567,7 +2562,6 @@ static void alc888_acer_aspire_7730g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -2579,7 +2573,6 @@ static void alc889_acer_aspire_8930g_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3667,7 +3660,6 @@ static void alc880_uniwill_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3699,7 +3691,6 @@ static void alc880_uniwill_p53_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -3969,7 +3960,6 @@ static void alc880_lg_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -4055,7 +4045,6 @@ static void alc880_lg_lw_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -4128,7 +4117,6 @@ static void alc880_medion_rim_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -6277,7 +6265,6 @@ static void alc260_hp_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x0f; spec->autocfg.speaker_pins[0] = 0x10; spec->autocfg.speaker_pins[1] = 0x11; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -6308,7 +6295,6 @@ static void alc260_hp_3013_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x10; spec->autocfg.speaker_pins[1] = 0x11; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -6352,7 +6338,6 @@ static void alc260_hp_3012_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x0f; spec->autocfg.speaker_pins[1] = 0x11; spec->autocfg.speaker_pins[2] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -8887,7 +8872,6 @@ static void alc885_imac24_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8901,7 +8885,6 @@ static void alc885_mba21_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8913,7 +8896,6 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8924,7 +8906,6 @@ static void alc885_imac91_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x18; spec->autocfg.speaker_pins[1] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -8958,7 +8939,6 @@ static void alc882_targa_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9405,7 +9385,6 @@ static void alc883_medion_wim2160_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1a; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9559,7 +9538,6 @@ static void alc883_mitac_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9725,7 +9703,6 @@ static void alc888_3st_hp_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; spec->autocfg.speaker_pins[2] = 0x18; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9786,7 +9763,6 @@ static void alc888_lenovo_ms7195_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9797,7 +9773,6 @@ static void alc883_lenovo_nb0763_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9811,7 +9786,6 @@ static void alc883_clevo_m720_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9841,7 +9815,6 @@ static void alc883_2ch_fujitsu_pi2515_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9851,7 +9824,6 @@ static void alc883_haier_w66_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9862,7 +9834,6 @@ static void alc883_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->detect_line = 1; spec->automute_lines = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; @@ -9876,7 +9847,6 @@ static void alc883_acer_aspire_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9907,7 +9877,6 @@ static void alc888_6st_dell_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x15; spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9921,7 +9890,6 @@ static void alc888_lenovo_sky_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[2] = 0x16; spec->autocfg.speaker_pins[3] = 0x17; spec->autocfg.speaker_pins[4] = 0x1a; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9932,7 +9900,6 @@ static void alc883_vaiott_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -9959,7 +9926,6 @@ static void alc883_mode2_setup(struct hda_codec *codec) spec->ext_mic.mux_idx = 0; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11470,7 +11436,6 @@ static void alc262_hp_bpc_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11480,7 +11445,6 @@ static void alc262_hp_wildwest_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11560,7 +11524,6 @@ static void alc262_hp_t5735_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11674,7 +11637,6 @@ static void alc262_hippo_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11684,7 +11646,6 @@ static void alc262_hippo1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11748,7 +11709,6 @@ static void alc262_tyan_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -11893,7 +11853,6 @@ static void alc262_toshiba_s06_setup(struct hda_codec *codec) spec->int_mic.pin = 0x12; spec->int_mic.mux_idx = 9; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -11998,7 +11957,6 @@ static void alc262_fujitsu_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.hp_pins[1] = 0x1b; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -12045,7 +12003,6 @@ static void alc262_lenovo_3000_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[1] = 0x16; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -13255,7 +13212,6 @@ static void alc268_acer_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -13339,7 +13295,6 @@ static void alc268_acer_lc_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -13379,7 +13334,6 @@ static void alc268_dell_setup(struct hda_codec *codec) spec->int_mic.pin = 0x19; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -13411,7 +13365,6 @@ static void alc267_quanta_il1_setup(struct hda_codec *codec) spec->int_mic.pin = 0x19; spec->int_mic.mux_idx = 1; spec->auto_mic = 1; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; } @@ -14415,7 +14368,6 @@ static void alc269_quanta_fl1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14437,7 +14389,6 @@ static void alc269_lifebook_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x1a; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; } @@ -14510,7 +14461,6 @@ static void alc269_laptop_amic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14525,7 +14475,6 @@ static void alc269_laptop_dmic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14540,7 +14489,6 @@ static void alc269vb_laptop_amic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -14555,7 +14503,6 @@ static void alc269vb_laptop_dmic_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -16878,7 +16825,6 @@ static void alc861vd_lenovo_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -16953,7 +16899,6 @@ static void alc861vd_dallas_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -18222,7 +18167,6 @@ static void alc662_lenovo_101e_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.line_out_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x15; - spec->automute = 1; spec->detect_line = 1; spec->automute_lines = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; @@ -18246,7 +18190,6 @@ static void alc662_eeepc_ep20_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x1b; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } @@ -18256,7 +18199,6 @@ static void alc663_m51va_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18272,7 +18214,6 @@ static void alc663_mode1_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18287,7 +18228,6 @@ static void alc662_mode2_setup(struct hda_codec *codec) struct alc_spec *spec = codec->spec; spec->autocfg.hp_pins[0] = 0x1b; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18303,7 +18243,6 @@ static void alc663_mode3_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18321,7 +18260,6 @@ static void alc663_mode4_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x16; spec->automute_mixer_nid[0] = 0x0c; spec->automute_mixer_nid[1] = 0x0e; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18339,7 +18277,6 @@ static void alc663_mode5_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[1] = 0x16; spec->automute_mixer_nid[0] = 0x0c; spec->automute_mixer_nid[1] = 0x0e; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18356,7 +18293,6 @@ static void alc663_mode6_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->automute_mixer_nid[0] = 0x0c; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_MIXER; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18373,7 +18309,6 @@ static void alc663_mode7_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18390,7 +18325,6 @@ static void alc663_mode8_setup(struct hda_codec *codec) spec->autocfg.hp_pins[1] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; spec->autocfg.speaker_pins[0] = 0x17; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_PIN; spec->ext_mic.pin = 0x18; spec->ext_mic.mux_idx = 0; @@ -18405,7 +18339,6 @@ static void alc663_g71v_setup(struct hda_codec *codec) spec->autocfg.hp_pins[0] = 0x21; spec->autocfg.line_out_pins[0] = 0x15; spec->autocfg.speaker_pins[0] = 0x14; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; spec->detect_line = 1; spec->automute_lines = 1; @@ -19818,7 +19751,6 @@ static void alc680_base_setup(struct hda_codec *codec) spec->autocfg.inputs[0].type = AUTO_PIN_MIC; spec->autocfg.inputs[1].pin = 0x19; spec->autocfg.inputs[1].type = AUTO_PIN_LINE_IN; - spec->automute = 1; spec->automute_mode = ALC_AUTOMUTE_AMP; } -- 1.7.5.4

2 3

[alsa-devel] [PATCH] ASoC: Correct WM8994 MICBIAS supply widget hookup
by Mark Brown 14 Jul '11

14 Jul '11

The WM8994 and WM8958 series of devices have two MICBIAS supplies rather than one, the current widget actually manages the microphone detection control register bit (which is managed separately by the relevant API). Fix this, hooking the relevant supplies up to the MICBIAS1 and MICBIAS2 widgets. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 79dee89..414f817 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1206,7 +1206,6 @@ SND_SOC_DAPM_INPUT("DMIC1DAT"), SND_SOC_DAPM_INPUT("DMIC2DAT"), SND_SOC_DAPM_INPUT("Clock"), -SND_SOC_DAPM_MICBIAS("MICBIAS", WM8994_MICBIAS, 2, 0), SND_SOC_DAPM_SUPPLY_S("MICBIAS Supply", 1, SND_SOC_NOPM, 0, 0, micbias_ev, SND_SOC_DAPM_PRE_PMU), @@ -1517,8 +1516,10 @@ static const struct snd_soc_dapm_route wm8994_revd_intercon[] = { { "AIF2DACDAT", NULL, "AIF1DACDAT" }, { "AIF1ADCDAT", NULL, "AIF2ADCDAT" }, { "AIF2ADCDAT", NULL, "AIF1ADCDAT" }, - { "MICBIAS", NULL, "CLK_SYS" }, - { "MICBIAS", NULL, "MICBIAS Supply" }, + { "MICBIAS1", NULL, "CLK_SYS" }, + { "MICBIAS1", NULL, "MICBIAS Supply" }, + { "MICBIAS2", NULL, "CLK_SYS" }, + { "MICBIAS2", NULL, "MICBIAS Supply" }, }; static const struct snd_soc_dapm_route wm8994_intercon[] = { -- 1.7.5.4

1 0

[alsa-devel] [RFC PATCH] ASoC: codecs: add codec driver for MFD mc13783
by Jürgen Lambrecht 14 Jul '11

14 Jul '11

Sascha Hauer's driver from Pengutronix' imx-sound branch. Added some code found in a linux 2.6.33. Needs further work and cleanup. Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/mc13783.c | 781 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/mc13783.h | 29 ++ 4 files changed, 816 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/mc13783.c create mode 100644 sound/soc/codecs/mc13783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 98175a0..77fb7b2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX98095 if I2C select SND_SOC_MAX9850 if I2C select SND_SOC_MAX9877 if I2C + select SND_SOC_MC13783 if SPI select SND_SOC_PCM3008 select SND_SOC_SGTL5000 if I2C select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -200,6 +201,9 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate +config SND_SOC_MC13783 + tristate + config SND_SOC_PCM3008 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fd85584..a129af2 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -22,6 +22,7 @@ snd-soc-l3-objs := l3.o snd-soc-max98088-objs := max98088.o snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o +snd-soc-mc13783-objs := mc13783.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o @@ -115,6 +116,7 @@ obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o obj-$(CONFIG_SND_SOC_MAX98088) += snd-soc-max98088.o obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o +obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c new file mode 100644 index 0000000..af7603a --- /dev/null +++ b/sound/soc/codecs/mc13783.c @@ -0,0 +1,781 @@ +/* + * Copyright 2008 Juergen Beisert, kernel(a)pengutronix.de + * Copyright 2009 Sascha Hauer, s.hauer(a)pengutronix.de + * + * Initial development of this code was funded by + * Phytec Messtechnik GmbH, http://www.phytec.de + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * 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 Street, Fifth Floor, Boston, + * MA 02110-1301, USA. + */ +#include <linux/device.h> +#include <linux/mfd/mc13783.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <sound/soc-dapm.h> + +#include "mc13783.h" + +#define PMIC_AUDIO_RX_0 36 +#define PMIC_AUDIO_RX_1 37 +#define PMIC_AUDIO_TX 38 +#define PMIC_SSI_NETWORK 39 +#define PMIC_AUDIO_CODEC 40 +#define PMIC_AUDIO_DAC 41 + +#define SSI_NETWORK_CDCTXRXSLOT(x) (((x) & 0x3) << 2) +#define SSI_NETWORK_CDCTXSECSLOT(x) (((x) & 0x3) << 4) +#define SSI_NETWORK_CDCRXSECSLOT(x) (((x) & 0x3) << 6) +#define SSI_NETWORK_CDCRXSECGAIN(x) (((x) & 0x3) << 8) +#define SSI_NETWORK_CDCSUMGAIN(x) (1 << 10) +#define SSI_NETWORK_CDCFSDLY(x) (1 << 11) +#define SSI_NETWORK_STDCSLOT(x) (((x) & 0x3) << 12) +#define SSI_NETWORK_STDCRXSLOT(x) (((x) & 0x3) << 14) +#define SSI_NETWORK_STDCRXSECSLOT(x) (((x) & 0x3) << 16) +#define SSI_NETWORK_STDCRXSECGAIN(x) (((x) & 0x3) << 18) +#define SSI_NETWORK_STDCSUMGAIN (1 << 20) + +#define STEREO_DAC_STD_SSI_SEL (1 << 0) +#define STEREO_DAC_STD_CLK_SEL (1 << 1) +#define STEREO_DAC_STD_CSM (1 << 2) +#define STEREO_DAC_STD_BCL_INV (1 << 3) +#define STEREO_DAC_STD_CFS_INV (1 << 4) +#define STEREO_DAC_STD_CFS(x) (((x) & 0x3) << 5) +#define STEREO_DAC_STD_CLK(x) (((x) & 0x7) << 7) +#define STEREO_DAC_STD_CFS_DLY_B (1 << 10) +#define STEREO_DAC_STD_C_EN (1 << 11) +#define STEREO_DAC_STD_C_CLK_EN (1 << 12) +#define STEREO_DAC_STD_C_RESET (1 << 15) +#define STEREO_DAC_SPDIF (1 << 16) +#define STEREO_DAC_SR(x) (((x) & 0xf) << 17) + +#define RX_0_VAUDIOON 0 +#define RX_0_BIASEN 1 +#define RX_0_BIASSPEED 2 +#define RX_0_ASPEN 3 +#define RX_0_ASPSEL 4 +#define RX_0_ALSPEN 5 +#define RX_0_ALSPREF 6 +#define RX_0_ALSPSEL 7 +#define RX_0_LSPLEN 8 +#define RX_0_AHSREN 9 +#define RX_0_AHSLEN 10 +#define RX_0_AHSSEL 11 +#define RX_0_HSPGDIG 12 +#define RX_0_HSDETEN 13 +#define RX_0_HSDETAUTOB 14 +#define RX_0_ARXOUTREN 15 +#define RX_0_ARXOUTLEN 16 +#define RX_0_ARXOUTSEL 17 +#define RX_0_CDCOUTEN 18 +#define RX_0_HSLDETEN 19 +#define RX_0_ADDCDC 21 +#define RX_0_ADDSTDC 22 +#define RX_0_ADDRXIN 23 + +#define RX_1_PGASTEN 5 +#define RX_1_PGAST 6 +#define RX_1_ARXINEN 10 +#define RX_1_ARXIN 11 +#define RX_1_PGARXIN 12 + +#define MC13783_REG_CACHE_SIZE 42 + +struct mc13783_priv { + struct snd_soc_codec codec; + struct mc13783 *mc13783; + + u32 reg_cache[MC13783_REG_CACHE_SIZE]; + + int mc13783_asp_val; + int mc13783_alsp_val; + int mc13783_ahsx_val; +}; + +static unsigned int mc13783_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u32 ret; + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + if (reg >= MC13783_REG_CACHE_SIZE) { + ret = 0; + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); + } else { + ret = priv->reg_cache[reg]; + } + return ret; +} + +static int mc13783_write(struct snd_soc_codec *codec, + unsigned int reg, unsigned int value) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int ret; + + if (reg >= MC13783_REG_CACHE_SIZE) { + printk(KERN_WARNING "%s Invalid register R%u\n", __func__, reg); + ret = -1; + } else { + mc13783_lock(priv->mc13783); + priv->reg_cache[reg] = value; + + ret = mc13783_reg_write(priv->mc13783, reg, value); + + mc13783_unlock(priv->mc13783); + } + return ret; +} + +/* Mapping between sample rates and register value */ +static unsigned int mc13783_rates[] = { + 8000, 11025, 12000, 16000, + 22050, 24000, 32000, 44100, + 48000, 64000, 96000 +}; + +static int mc13783_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + unsigned int rate = params_rate(params); + int i; + unsigned int reg; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < ARRAY_SIZE(mc13783_rates); i++) { + if (rate == mc13783_rates[i]) { + snd_soc_update_bits(codec, PMIC_AUDIO_DAC, + 0xf << 17, i << 17); + return 0; + } + } + } else { + if (rate == 8000) + reg = 0; + else + reg = (1 << 10); + + snd_soc_update_bits(codec, PMIC_AUDIO_CODEC, + 0x1 << 10, reg); + + return 0; + } + + return -EINVAL; +} + +static int mc13783_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg; + + if (dai->id == MC13783_ID_STEREO_DAC) + reg = mc13783_read(codec, PMIC_AUDIO_DAC); + else + reg = mc13783_read(codec, PMIC_AUDIO_CODEC); + + reg &= ~STEREO_DAC_STD_CFS(3); + reg &= ~STEREO_DAC_STD_BCL_INV; + reg &= ~STEREO_DAC_STD_CFS_INV; + reg &= ~STEREO_DAC_STD_CSM; + reg &= ~STEREO_DAC_STD_C_CLK_EN; + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + reg |= STEREO_DAC_STD_CFS(2); + break; + case SND_SOC_DAIFMT_DSP_A: + reg |= STEREO_DAC_STD_CFS(1); + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + reg |= STEREO_DAC_STD_BCL_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + reg |= STEREO_DAC_STD_BCL_INV; + reg |= STEREO_DAC_STD_CFS_INV; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + reg |= STEREO_DAC_STD_CFS_INV; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + reg |= STEREO_DAC_STD_C_CLK_EN; + break; + case SND_SOC_DAIFMT_CBS_CFS: + reg |= STEREO_DAC_STD_CSM; + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFM: + return -EINVAL; + } + + reg |= STEREO_DAC_STD_C_EN; /* DAC power up */ + reg |= STEREO_DAC_STD_C_RESET; /* reset filter */ + + if (dai->id == MC13783_ID_STEREO_DAC) + mc13783_write(codec, PMIC_AUDIO_DAC, reg); + else + mc13783_write(codec, PMIC_AUDIO_CODEC, reg); + + return 0; +} + +static int mc13783_set_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + int clk; + unsigned int reg; + + if (dai->id == MC13783_ID_STEREO_DAC) + reg = mc13783_read(codec, PMIC_AUDIO_DAC); + else + reg = mc13783_read(codec, PMIC_AUDIO_CODEC); + + reg &= ~STEREO_DAC_STD_CLK(0x7); + reg &= ~STEREO_DAC_STD_CLK_SEL; + + switch (freq) { + case 13000000: + clk = 0; + break; + case 15360000: + clk = 1; + break; + case 16800000: + clk = 2; + break; + case 26000000: + clk = 4; + break; + case 12000000: + clk = 5; + break; + case 3686400: + clk = 6; + break; + case 33600000: + clk = 7; + break; + default: + return -EINVAL; + } + + if (clk_id == MC13783_CLK_CLIB) + reg |= STEREO_DAC_STD_CLK_SEL; + + reg |= STEREO_DAC_STD_CLK(clk); + + if (dai->id == MC13783_ID_STEREO_DAC) + mc13783_write(codec, PMIC_AUDIO_DAC, reg); + else + mc13783_write(codec, PMIC_AUDIO_CODEC, reg); + + return 0; +} + +static int mc13783_set_tdm_slot_codec(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + unsigned int reg; + + if (slots != 4) + return -EINVAL; + + reg = mc13783_read(codec, PMIC_SSI_NETWORK); + + reg &= ~(0xfff << 0); + reg |= (0x00 << 2); /* primary timeslot RX/TX(?) is 0 */ + reg |= (0x01 << 4); /* secondary timeslot TX is 1 */ + reg |= (0x01 << 6); /* secondary timeslot RX is 1 */ + + mc13783_write(codec, PMIC_SSI_NETWORK, reg); + + return 0; +} + +static int mc13783_get_alsp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = priv->mc13783_alsp_val; + + return 0; +} + +static int mc13783_put_alsp(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + priv->mc13783_alsp_val = ucontrol->value.integer.value[0]; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + reg &= ~((1 << RX_0_ALSPEN) | (1 << RX_0_ALSPSEL)); + + if (priv->mc13783_alsp_val) + reg |= 1 << RX_0_ALSPEN; + + if (priv->mc13783_alsp_val == 2) + reg |= 1 << RX_0_ALSPSEL; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; +} + +static int mc13783_get_ahsx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.integer.value[0] = priv->mc13783_ahsx_val; + + return 0; +} + +static int mc13783_put_ahsx(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + priv->mc13783_ahsx_val = ucontrol->value.integer.value[0]; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + reg &= ~((1 << RX_0_AHSREN) | (1 << RX_0_AHSLEN) | (1 << RX_0_AHSSEL)); + + if (priv->mc13783_ahsx_val) + reg |= (1 << RX_0_AHSREN) | (1 << RX_0_AHSLEN); + + if (priv->mc13783_ahsx_val == 2) + reg |= 1 << RX_0_AHSSEL; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; + } + +static int mc13783_pcm_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = mc13783_read(codec, PMIC_AUDIO_RX_0); + ucontrol->value.enumerated.item[0] = (val >> RX_0_ADDSTDC) & 1; + printk(KERN_DEBUG "pcm_get %d\n", ucontrol->value.enumerated.item[0]); + + return 0; +} + +static int mc13783_pcm_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r36, r37; + + r36 = mc13783_read(codec, PMIC_AUDIO_RX_0); + r37 = mc13783_read(codec, PMIC_AUDIO_RX_1); + + r36 &= ~(1 << RX_0_ADDSTDC); + r37 &= ~(1 << RX_1_PGASTEN); + + if (ucontrol->value.enumerated.item[0]) { + printk(KERN_DEBUG "pcm_put on %d\n", ucontrol->value.enumerated.item[0]); + r36 |= (1 << RX_0_ADDSTDC); + r37 |= (1 << RX_1_PGASTEN); + } else { + printk(KERN_DEBUG "pcm_put off\n"); + r36 &= ~(1 << RX_0_ADDSTDC); + r37 &= ~(1 << RX_1_PGASTEN); + } + + mc13783_write(codec, PMIC_AUDIO_RX_0, r36); + mc13783_write(codec, PMIC_AUDIO_RX_1, r37); + + return 0; +} + +static int mc13783_linein_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + int val; + + val = mc13783_read(codec, PMIC_AUDIO_RX_0); + ucontrol->value.enumerated.item[0] = (val >> RX_0_ADDRXIN) & 1; + + return 0; +} + +static int mc13783_linein_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r36, r37; + + r36 = mc13783_read(codec, PMIC_AUDIO_RX_0); + r37 = mc13783_read(codec, PMIC_AUDIO_RX_1); + + r36 &= ~(1 << RX_0_ADDRXIN); + r37 &= ~(1 << RX_1_ARXINEN); + + if (ucontrol->value.enumerated.item[0]) { + r36 |= (1 << RX_0_ADDRXIN); + r37 |= (1 << RX_1_ARXINEN); + } else { + r36 &= ~(1 << RX_0_ADDRXIN); + r37 &= ~(1 << RX_1_ARXINEN); + } + + mc13783_write(codec, PMIC_AUDIO_RX_0, r36); + mc13783_write(codec, PMIC_AUDIO_RX_1, r37); + + return 0; +} + +static int mc13783_capure_cache; + +static int mc13783_get_capture(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = mc13783_capure_cache; + return 0; +} + +#define AMC1REN (1 << 5) +#define AMC2EN (1 << 9) +#define ATXINEN (1 << 11) +#define RXINREC (1 << 13) +#define AMC1LEN (1 << 7) + +static int mc13783_put_capture(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int r38, change; + + r38 = mc13783_read(codec, PMIC_AUDIO_TX); + + change = (mc13783_capure_cache != ucontrol->value.enumerated.item[0]); + mc13783_capure_cache = ucontrol->value.enumerated.item[0]; + r38 &= ~(AMC1REN | AMC2EN | ATXINEN | RXINREC | AMC1LEN); + + switch (mc13783_capure_cache) { + case 0: + break; + case 1: + r38 |= RXINREC; + break; + case 2: + r38 |= AMC1REN | AMC1LEN; + break; + case 3: + r38 |= AMC1REN; + break; + case 4: + r38 |= AMC2EN; + break; + case 5: + r38 |= AMC1LEN | AMC2EN; + break; + case 6: + r38 |= ATXINEN; + break; + case 7: + r38 |= AMC1LEN | ATXINEN; + break; + case 8: + r38 |= AMC1LEN | RXINREC; + break; + case 9: + r38 |= AMC1LEN; + break; + default: + break; + } + + mc13783_write(codec, PMIC_AUDIO_TX, r38); + + return change; +} + +static const char *mc13783_asp[] = {"Off", "Codec", "Right"}; +static const char *mc13783_alsp[] = {"Off", "Codec", "Right"}; + +static const char *mc13783_ahs[] = {"Codec", "Mixer"}; + +//static const char *mc13783_ahsx[] = {"Off", "Codec", "Right"}; + +static const char *mc13783_arxout[] = {"Codec", "Mixer"}; + +static const char *mc13783_capture[] = {"off/off", "rxinl/rxinr", + "mc1lin/mc1rin", "off/mc1rin", "off/mc2in", "mc1lin/mc2in", + "off/txin", "mc1lin/txin", "mc1lin/rxinr", "mc1lin/off"}; + +static const char *mc13783_3d_mixer[] = {"Stereo", "Phase Mix", + "Mono", "Mono Mix"}; + +//static const char *mc13783_pcms[] = {"Off", "On"}; + +static const struct soc_enum mc13783_enum_asp = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 3, ARRAY_SIZE(mc13783_asp), mc13783_asp); + +static const struct soc_enum mc13783_enum_alsp = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mc13783_alsp), mc13783_alsp); + +static const struct soc_enum mc13783_enum_ahs = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 11, ARRAY_SIZE(mc13783_ahs), + mc13783_ahs); + +static const struct soc_enum mc13783_enum_arxout = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 17, ARRAY_SIZE(mc13783_arxout), + mc13783_arxout); + +static const struct soc_enum mc13783_enum_capture = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(mc13783_capture), mc13783_capture); + +static const struct soc_enum mc13783_enum_3d_mixer = + SOC_ENUM_SINGLE(PMIC_AUDIO_RX_1, 16, ARRAY_SIZE(mc13783_3d_mixer), + mc13783_3d_mixer); + +//static const struct soc_enum mc13783_enum_ahsx = +// SOC_ENUM_SINGLE(PMIC_AUDIO_RX_0, 9, ARRAY_SIZE(mc13783_ahsx), +// mc13783_ahsx); +// +//static const struct soc_enum mc13783_enum_pcms = +// SOC_ENUM_SINGLE(PMIC_AUDIO_RX_1, 5, ARRAY_SIZE(mc13783_pcms), +// mc13783_pcms); + +static struct snd_kcontrol_new mc13783_control_list[] = { + /* Output Routing */ + SOC_ENUM("Asp Source", mc13783_enum_asp), + SOC_ENUM_EXT("Alsp Source", mc13783_enum_alsp, mc13783_get_alsp, + mc13783_put_alsp), + SOC_ENUM("Ahs Source", mc13783_enum_ahs), + SOC_SINGLE("Ahsr enable", PMIC_AUDIO_RX_0, 9, 1, 0), + SOC_SINGLE("Ahsl enable", PMIC_AUDIO_RX_0, 10, 1, 0), +/* instead of the above 3 lines, could put this line: */ +// SOC_ENUM_EXT("Ahs Source", mc13783_enum_ahsx, mc13783_get_ahsx, +// mc13783_put_ahsx), +//// SOC_ENUM("Ahs Source", mc13783_enum_ahs), +//// SOC_SINGLE("Ahsr enable", PMIC_AUDIO_RX_0, 9, 2, 0), +//// SOC_SINGLE("Ahsl enable", PMIC_AUDIO_RX_0, 10, 2, 0), +//// SOC_ENUM("Ahsr enable", mc13783_enum_ahsr), +//// SOC_ENUM("Ahsl enable", mc13783_enum_ahsl), + SOC_ENUM("Arxout Source", mc13783_enum_arxout), + SOC_SINGLE("ArxoutR enable", PMIC_AUDIO_RX_0, 16, 1, 0), + SOC_SINGLE("ArxoutL enable", PMIC_AUDIO_RX_0, 15, 1, 0), + SOC_SINGLE_EXT("PCM Playback Switch", 0, 0, 1, 0, mc13783_pcm_get, +// SOC_ENUM_EXT("PCMS Playback Switch", mc13783_enum_pcms, mc13783_pcm_get, + mc13783_pcm_put), + SOC_SINGLE("PCM Playback Volume", PMIC_AUDIO_RX_1, 6, 15, 0), +// SOC_SINGLE("PCMP Playback Volume", PMIC_AUDIO_RX_1, RX_1_PGAST, 15, 0), + SOC_SINGLE_EXT("Line in Switch", 0, 0, 1, 0, mc13783_linein_get, + mc13783_linein_put), + SOC_SINGLE("Line in Volume", PMIC_AUDIO_RX_1, 12, 15, 0), +// SOC_SINGLE("Line in Volume", PMIC_AUDIO_RX_1, RX_1_PGARXIN, 15, 0), + SOC_ENUM_EXT("Capture Source", mc13783_enum_capture, mc13783_get_capture, + mc13783_put_capture), + SOC_DOUBLE("PCM Capture Volume", PMIC_AUDIO_TX, 19, 14, 31, 0), + SOC_ENUM("3D Control", mc13783_enum_3d_mixer), +}; + +static int mc13783_probe(struct snd_soc_codec *codec) +{ + struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); + int i, ret = 0; + + /* these are the reset values */ + priv->reg_cache[PMIC_AUDIO_RX_0] = (1 << RX_0_HSPGDIG); //0x001000; + priv->reg_cache[PMIC_AUDIO_RX_1] = 0x00d35A; + priv->reg_cache[PMIC_AUDIO_TX] = 0x420000; + priv->reg_cache[PMIC_SSI_NETWORK] = 0x013060; + priv->reg_cache[PMIC_AUDIO_CODEC] = 0x180027; + priv->reg_cache[PMIC_AUDIO_DAC] = 0x0e0004; + + /* VAUDIOON -> supply audio part, BIAS enable */ + priv->reg_cache[PMIC_AUDIO_RX_0] |= (1 << RX_0_VAUDIOON) | (1 << RX_0_BIASEN); //0x3; + + for (i = 36; i < 42; i++) + mc13783_write(codec, i, priv->reg_cache[i]); + + snd_soc_add_controls(codec, mc13783_control_list, + ARRAY_SIZE(mc13783_control_list)); + + return ret; +} + +static int mc13783_remove(struct snd_soc_codec *codec) +{ + unsigned int reg; + + reg = mc13783_read(codec, PMIC_AUDIO_RX_0); + + /* VAUDIOON -> switch off audio part, BIAS disable */ + reg &= ~0x3; + + mc13783_write(codec, PMIC_AUDIO_RX_0, reg); + + return 0; +} + +#define MC13783_RATES_RECORD (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000) + +#define MC13783_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ + SNDRV_PCM_FMTBIT_S24_LE) +/*Other functions (soc_dai.h): + * set_pll, set_clkdiv + * set_tdm_slot, set_channel_map, set_tristate + * digital_mute + * startup, shutdown, hw_free, prepare, trigger +*/ +static struct snd_soc_dai_ops mc13783_ops = { + .hw_params = mc13783_pcm_hw_params, + .set_fmt = mc13783_set_fmt, + .set_sysclk = mc13783_set_sysclk, + .set_tdm_slot = mc13783_set_tdm_slot_codec, +}; + +static struct snd_soc_dai_driver mc13783_dai = { + .name = "mc13783-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = MC13783_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = MC13783_RATES_RECORD, + .formats = MC13783_FORMATS, + }, + .ops = &mc13783_ops, +}; + +/* + * ASoC codec device structure + * + * Assign this variable to the codec_dev field of the machine driver's + * snd_soc_device structure. + */ +static struct snd_soc_codec_driver soc_codec_dev_mc13783 = { + .probe = mc13783_probe, + .remove = mc13783_remove, +// .suspend +// .resume + .read = mc13783_read, + .write = mc13783_write, +// .display_register +// .volatile_register +// .readable_register +// .reg_cache_size = MC13783_REG_CACHE_SIZE, TODO: use codec->reg_cache instead of privately +// reg_cache_step +// .reg_word_size = sizeof(u32), TODO: else: kmemdup error in soc-core.c::snd_soc_register_codec +// .reg_cache_default = wm8960_reg, +// .compress_type +// .set_bias_level +}; + +static int mc13783_codec_probe(struct platform_device *pdev) +{ + struct mc13783 *mc13783; + struct mc13783_priv *priv; + int ret; + + mc13783 = dev_get_drvdata(pdev->dev.parent); + + priv = kzalloc(sizeof(struct mc13783_priv), GFP_KERNEL); + if (priv == NULL) + return -ENOMEM; + + dev_set_drvdata(&pdev->dev, priv); + priv->mc13783 = mc13783; + + ret = snd_soc_register_codec(&pdev->dev, &soc_codec_dev_mc13783, &mc13783_dai, 1); + if (ret) + goto err_register_codec; + + return 0; + +err_register_codec: + dev_err(&pdev->dev, "register codec failed with %d\n", ret); + kfree(priv); + + return ret; +} + +static int mc13783_codec_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + + return 0; +} + +static struct platform_driver mc13783_codec_driver = { + .driver = { + .name = "mc13783-codec", + .owner = THIS_MODULE, + }, + .probe = mc13783_codec_probe, + .remove = __devexit_p(mc13783_codec_remove), +}; + +static __init int mc13783_init(void) +{ + return platform_driver_register(&mc13783_codec_driver); +} + +static __exit void mc13783_exit(void) +{ + platform_driver_unregister(&mc13783_codec_driver); +} + +module_init(mc13783_init); +module_exit(mc13783_exit); + +MODULE_DESCRIPTION("ASoC MC13783 driver"); +MODULE_AUTHOR("Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/mc13783.h b/sound/soc/codecs/mc13783.h new file mode 100644 index 0000000..7285370 --- /dev/null +++ b/sound/soc/codecs/mc13783.h @@ -0,0 +1,29 @@ +/* + * Copyright 2008 Juergen Beisert, kernel(a)pengutronix.de + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * 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 Street, Fifth Floor, Boston, MA 02110-1301, USA. + */ + +#ifndef MC13783_MIXER_H +#define MC13783_MIXER_H + +extern int mc13783_add_ctl(struct snd_card*, void *); + +#define MC13783_CLK_CLIA 1 +#define MC13783_CLK_CLIB 2 + +#define MC13783_ID_STEREO_DAC 1 +#define MC13783_ID_STEREO_CODEC 2 + +#endif /* MC13783_MIXER_H */ -- 1.7.1

4 3

[alsa-devel] [PATCH] ASoC: Reduce power consumption for idle DAIs in WM8994
by Mark Brown 14 Jul '11

14 Jul '11

If DAIs are idle but their clocks are in use for some reason (eg, as SYSCLK or for accessory detect) then set the clock dividers to the maximum to reduce slightly the power consumption of the unclocked circuits. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 29 +++++++++++++++++++++++++++++ 1 files changed, 29 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 31854dc..bc62b6e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2311,6 +2311,33 @@ static int wm8994_aif3_hw_params(struct snd_pcm_substream *substream, return snd_soc_update_bits(codec, aif1_reg, WM8994_AIF1_WL_MASK, aif1); } +static void wm8994_aif_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + int rate_reg = 0; + + switch (dai->id) { + case 1: + rate_reg = WM8994_AIF1_RATE; + break; + case 2: + rate_reg = WM8994_AIF1_RATE; + break; + default: + break; + } + + /* If the DAI is idle then configure the divider tree for the + * lowest output rate to save a little power if the clock is + * still active (eg, because it is system clock). + */ + if (rate_reg && !dai->playback_active && !dai->capture_active) + snd_soc_update_bits(codec, rate_reg, + WM8994_AIF1_SR_MASK | + WM8994_AIF1CLK_RATE_MASK, 0x9); +} + static int wm8994_aif_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -2377,6 +2404,7 @@ static struct snd_soc_dai_ops wm8994_aif1_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, + .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, @@ -2386,6 +2414,7 @@ static struct snd_soc_dai_ops wm8994_aif2_dai_ops = { .set_sysclk = wm8994_set_dai_sysclk, .set_fmt = wm8994_set_dai_fmt, .hw_params = wm8994_hw_params, + .shutdown = wm8994_aif_shutdown, .digital_mute = wm8994_aif_mute, .set_pll = wm8994_set_fll, .set_tristate = wm8994_set_tristate, -- 1.7.5.4

1 0

Re: [alsa-devel] [PATCH] ASoC: wm8900: fix a memory leak if wm8900_set_fll fails
by Mark Brown 14 Jul '11

14 Jul '11

On Thu, Jul 14, 2011 at 06:14:46PM +0800, Axel Lin wrote: > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> Applied, thanks

1 0

[alsa-devel] www.alsa-project.org down?
by David Henningsson 14 Jul '11

14 Jul '11

Hi, Is www.alsa-project.org down? Someone in a bug report he couldn't connect to it and here's what happens when I try: $ wget http://www.alsa-project.org --2011-07-08 11:41:36-- http://www.alsa-project.org/ Resolving www.alsa-project.org... 77.48.224.243 Connecting to www.alsa-project.org|77.48.224.243|:80... connected. HTTP request sent, awaiting response... (It's been awaiting response for a few minutes now...) -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

3 2

Re: [alsa-devel] Specifying the default buffer-size for snd-hda-intel
by Raymond Yau 14 Jul '11

14 Jul '11

> +config SND_HDA_PREALLOC_SIZE > + int "Pre-allocated buffer size for HD-audio driver" > + range 0 32768 > + default 64 > + help > + Speficies the default pre-allocated buffer-size in kB for > + HD-audio driver. A larger buffer (e.g. 2048) is preferred > + for systems with PulseAudio. The default 64 is chosen just > + from the compatibility reason. > + + range 0 32768 - range 64 32768 It seem that the minimum range cannot be zero Does current PA server still work with sound drivers with 64K buffer ?

2 1

[alsa-devel] [PATCH] ASoC: Log WM8994 FIFO errors from the interrupt
by Mark Brown 14 Jul '11

14 Jul '11

We should spot them anyway on state changes but logging them gives us better time information about when the misconfiguration happened. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 1 files changed, 14 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index fcd067f..d774fd7a 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2894,6 +2894,15 @@ out: return IRQ_HANDLED; } +static irqreturn_t wm8994_fifo_error(int irq, void *data) +{ + struct snd_soc_codec *codec = data; + + dev_err(codec->dev, "FIFO error\n"); + + return IRQ_HANDLED; +} + static int wm8994_codec_probe(struct snd_soc_codec *codec) { struct wm8994 *control; @@ -2973,6 +2982,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) break; } + wm8994_request_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, + wm8994_fifo_error, "FIFO error", codec); + ret = wm8994_request_irq(codec->control_data, WM8994_IRQ_DCS_DONE, wm_hubs_dcs_done, "DC servo done", &wm8994->hubs); @@ -3255,6 +3267,7 @@ err_irq: &wm8994->fll_locked[i]); wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); err: kfree(wm8994); return ret; @@ -3276,6 +3289,7 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); + wm8994_free_irq(codec->control_data, WM8994_IRQ_FIFOS_ERR, codec); switch (control->type) { case WM8994: -- 1.7.5.4

1 0

[alsa-devel] ASoC updates for 3.1
by Mark Brown 14 Jul '11

14 Jul '11

The following changes since commit e320bc42be746b8f006d163b6e340e5fb79ce7b2: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-09 11:43:04 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Giridhar Maruthy (1): ASoC: SAMSUNG: 24-bit audio playback on Exynos4210 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Johannes Stezenbach (2): ASoC: STA32x: Add mixer controls for biquad coefficients ASoC: STA32x: Preserve reserved register bits Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name Mark Brown (9): Merge branch 'for-3.0' into for-3.1 Merge branch 'for-3.0' into for-3.1 Merge branch 'topic/asoc' of git://git.kernel.org/.../tiwai/sound-2.6 into for-3.1 ASoC: Conditionalize the enable of WM8994 ADC TDM mode ASoC: Use late enable handling for direct voice, speaker and headphone ASoC: Implement DC servo completion IRQ handling for wm_hubs devices ASoC: Hook up DC servo completion IRQ for WM8994 and WM8958 ASoC: Use WM8994 FLL lock interrupt ASoC: Don't warn on low WM8994/58 AIFnCLKs Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration Takashi Iwai (4): Merge branch 'for-3.0' of git://git.kernel.org/.../lrg/asoc-2.6 into fix/asoc Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc ASoC: Don't set invalid name string to snd_card->driver field Merge branch 'for-3.0' of git://git.kernel.org/.../broonie/sound-2.6 into fix/asoc sound/soc/codecs/sta32x.c | 140 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic26.c | 14 +++- sound/soc/codecs/tlv320aic3x.c | 9 ++- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 101 +++++++++++++++++++++++----- sound/soc/codecs/wm8994.h | 3 + sound/soc/codecs/wm_hubs.c | 37 ++++++++-- sound/soc/codecs/wm_hubs.h | 8 ++ sound/soc/samsung/smdk_wm8994.c | 5 +- sound/soc/sh/fsi-ak4642.c | 16 ++-- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 +- sound/soc/soc-core.c | 5 +- 13 files changed, 301 insertions(+), 45 deletions(-)

1 0

Re: [alsa-devel] hda-intel front headphone (8086:27d8 1043:83bc)
by Raymond Yau 13 Jul '11

13 Jul '11

> My knowledge of audio/alsa programming is really poor, but if someone suggest > a patch I can setup the thing and do some debug. Try Takashi 's daily snapshot tarball you should able to get "Headphone Playback Volume" after this patch "ALSA: hda - Code consolidation for ALC88x and ALC662 auto-parsers" using hda-emu and it seem the driver support 6 channels even when the pink and blue jacks are not retasked

2 3

[alsa-devel] ASoC updates for 3.0
by Mark Brown 13 Jul '11

13 Jul '11

The following changes since commit 18361bbe3180eca62796188d62aefac1519f4c83: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into fix/asoc (2011-07-09 09:44:09 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Kuninori Morimoto (3): ASoC: sh: fsi-ak4642: fixup snd_soc_card name ASoC: sh: fsi-da7210: fixup snd_soc_card name ASoC: sh: fsi-hdmi: fixup snd_soc_card name sound/soc/sh/fsi-ak4642.c | 16 ++++++++-------- sound/soc/sh/fsi-da7210.c | 2 +- sound/soc/sh/fsi-hdmi.c | 4 ++-- 3 files changed, 11 insertions(+), 11 deletions(-)

2 1

[alsa-devel] [PATCH 0/2] ASoC: STA32x: filter coefficients, bug fix
by Johannes Stezenbach 13 Jul '11

13 Jul '11

Hi, here are two patches for the ST STA32x ASoC codec driver. The first one adds SNDRV_CTL_ELEM_TYPE_BYTES controls for access to the biquad filter coefficients. I haven't found any other driver using SNDRV_CTL_ELEM_TYPE_BYTES in this way, but it works nicely. The coefficients can be set with amixer (although amixer has a quirk in that it prints the values in hex but can read them only as decimal). # amixer cget name='Ch1 - Biquad 1' numid=43,iface=MIXER,name='Ch1 - Biquad 1' ; type=BYTES,access=rw------,values=15 : values=0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00 # amixer cset name='Ch1 - Biquad 1' 12,34,0,0,0,0,0,0,0,0,0,0,0,255,78 numid=43,iface=MIXER,name='Ch1 - Biquad 1' ; type=BYTES,access=rw------,values=15 : values=0x0c,0x22,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0x00,0xff,0x4e Due to the way the internal coeffcient RAM is accessed, the corresponding I2C registers are volatile, thus I added the .volatile_register function in this patch. The second patch preserves reset values as required by the chip documentation, it is mainly for future compatibility and was added during hardware testing to rule out issues from this side, although it did not make a difference in my tests. However, the chip spec insists it is important. Thanks Johannes

2 3

[alsa-devel] AC97 on S3C6410
by Marco Sinigaglia 13 Jul '11

13 Jul '11

Hi All, I am still trying to implement the AC97, on S3C6410, using the WM9714. I have done some big step forward but I am now hitting another issue. The kernel can see the WM9714: <6>ALSA device list: <6> #0: SMDK (WM9713) Anyway, after that, i cannot see anymore activity on the AC97 bus. I got, indeed, these errors: <3>s3c-ac97: req addr = 12, rep addr = 00 [...] <3>s3c-ac97: req addr = 12, rep addr = 00 When I try to play something, i got this (I enabled the DEBUG on dma.c): <7>asoc: AC97 HiFi <-> s3c-ac97 info: <7>asoc: rate mask 0xd6 <7>asoc: min ch 2 max ch 2 <7>asoc: min rate 8000 max rate 48000 <7>dma22: s3c2410_request_dma: client=AC97 PCMOut, dev=(null) <7>DMA8: 00000000->00000000 L 00000000 C 00000000,00000000 S 00000000 <7>s3c2410_dma_request: channel initialised, c06779c8 <7>s3c2410_dma_devconfig: channel 22, source 1, dev 7f001018, chan c06779c8 <7>s3c2410_dma_devconfig: peripheral 6 <7>s3c2410_dma_devconfig: config 0000c980 <7>DMA8: 00000000->00000000 L 00000000 C 00000000,00000000 S 0000c980 <7>s3c64xx_dma_flush: flushing channel <7>s3c2410_dma_enqueue: buff c9dcfac0, dp 5e740000 lli (ffcd6000, 57241000) 8192 <7>enquing onto empty channel <7>s3c64xx_lli_to_regs: LLI ffcd6000 => regs <7>LLI[ffcd6000] 5e740000->7f001018, NL 00000000 C 96480000,00000800 <7>LLI[ffcd6000] 5e740000->7f001018, NL 00000000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 00000000 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfac0, curr c9dcfac0, end c9dcfac0 <7>s3c2410_dma_enqueue: buff c9dcfa60, dp 5e742000 lli (ffcd6020, 57241020) 8192 <7>enquing onto channel <7>LLI[ffcd6000] 5e740000->7f001018, NL 57241020 C 96480000,00000800 <7>LLI[ffcd6020] 5e742000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcfa60 <7>s3c2410_dma_enqueue: buff c9dcfa40, dp 5e744000 lli (ffcd6040, 57241040) 8192 <7>enquing onto channel <7>LLI[ffcd6020] 5e742000->7f001018, NL 57241040 C 96480000,00000800 <7>LLI[ffcd6040] 5e744000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcfa40 <7>s3c2410_dma_enqueue: buff c9dcfa20, dp 5e746000 lli (ffcd6060, 57241060) 8192 <7>enquing onto channel <7>LLI[ffcd6040] 5e744000->7f001018, NL 57241060 C 96480000,00000800 <7>LLI[ffcd6060] 5e746000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcfa20 <7>s3c2410_dma_enqueue: buff c9dcfa00, dp 5e748000 lli (ffcd6080, 57241080) 8192 <7>enquing onto channel <7>LLI[ffcd6060] 5e746000->7f001018, NL 57241080 C 96480000,00000800 <7>LLI[ffcd6080] 5e748000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcfa00 <7>s3c2410_dma_enqueue: buff c9dcf9e0, dp 5e74a000 lli (ffcd60a0, 572410a0) 8192 <7>enquing onto channel <7>LLI[ffcd6080] 5e748000->7f001018, NL 572410a0 C 96480000,00000800 <7>LLI[ffcd60a0] 5e74a000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf9e0 <7>s3c2410_dma_enqueue: buff c9dcf9c0, dp 5e74c000 lli (ffcd60c0, 572410c0) 8192 <7>enquing onto channel <7>LLI[ffcd60a0] 5e74a000->7f001018, NL 572410c0 C 96480000,00000800 <7>LLI[ffcd60c0] 5e74c000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf9c0 <7>s3c2410_dma_enqueue: buff c9dcf9a0, dp 5e74e000 lli (ffcd60e0, 572410e0) 8192 <7>enquing onto channel <7>LLI[ffcd60c0] 5e74c000->7f001018, NL 572410e0 C 96480000,00000800 <7>LLI[ffcd60e0] 5e74e000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf9a0 <7>s3c2410_dma_enqueue: buff c9dcf980, dp 5e750000 lli (ffcd6100, 57241100) 8192 <7>enquing onto channel <7>LLI[ffcd60e0] 5e74e000->7f001018, NL 57241100 C 96480000,00000800 <7>LLI[ffcd6100] 5e750000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf980 <7>s3c2410_dma_enqueue: buff c9dcf960, dp 5e752000 lli (ffcd6120, 57241120) 8192 <7>enquing onto channel <7>LLI[ffcd6100] 5e750000->7f001018, NL 57241120 C 96480000,00000800 <7>LLI[ffcd6120] 5e752000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c9dcf960 <7>s3c2410_dma_enqueue: buff c7242940, dp 5e754000 lli (ffcd6140, 57241140) 8192 <7>enquing onto channel <7>LLI[ffcd6120] 5e752000->7f001018, NL 57241140 C 96480000,00000800 <7>LLI[ffcd6140] 5e754000->7f001018, NL 57241000 C 96480000,00000800 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <7>DMA8: buffs next c9dcfa60, curr c9dcfac0, end c7242940 <7>DMA8: 5e740000->7f001018 L 57241020 C 96480000,00000000 S 0000c980 <3> clearing interrupts <7>s3c64xx_dma_start: clearing interrupts <7>s3c64xx_dma_start: starting channel <7>s3c64xx_dma_start: writing config 0000c981 <7>ALSA sound/core/pcm_lib.c:1755: playback write error (DMA or IRQ trouble?) <7>ALSA sound/core/pcm_lib.c:1755: playback write error (DMA or IRQ trouble?) <7>s3c64xx_dma_stop: stopping channel <7>DMA8: 5e740010->7f001018 L 57241020 C 96480000,00000800 S 0002c981 I have the feeling that, for some reasons, the DMA is not doing its job. I have added the following line on dma.c but it does not fix the error: s3c64xx_dma_init1(8, DMACH_AC97_PCMOUT, IRQ_DMA1, 0x75100000); Can someone, please, give me some feedback? I am using kernel 2.6.35 and the head of Android. Cheers Marco Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom More information can be found at www.csr.com. Follow CSR on Twitter at http://twitter.com/CSR_PLC and read our blog at www.csr.com/blog

1 0

[alsa-devel] [PATCH 1/5] ALSA: snd-usb: don't fail hard if mixer creation fails
by Daniel Mack 13 Jul '11

13 Jul '11

There are cards that the mixer code fails to create controls for, and currently, the whole sound device creation fails in such cases. This patch changes the code so that it just spits out warnings and continues looking at other functions of the descriptor set which might work just fine. The worst thing that can happen now is that people see soundcards that don't have all funcions supported, but rejecting the whole device is not any better. Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/card.c | 5 ++--- sound/usb/mixer.c | 9 ++++++--- 2 files changed, 8 insertions(+), 6 deletions(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index 220c616..8990b49 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -512,9 +512,8 @@ static void *snd_usb_audio_probe(struct usb_device *dev, if (err > 0) { /* create normal USB audio interfaces */ if (snd_usb_create_streams(chip, ifnum) < 0 || - snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) { - goto __error; - } + snd_usb_create_mixer(chip, ifnum, ignore_ctl_error) < 0) + snd_printk(KERN_WARNING "snd_usb_create_mixer() failed\n"); } /* we are allowed to call snd_card_register() many times */ diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index c22fa76..c43dbea 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1967,7 +1967,8 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) - return err; + snd_printk(KERN_WARNING "Unable to parse terminal with bSourceID %d\n", + desc->bSourceID); } else { /* UAC_VERSION_2 */ struct uac2_output_terminal_descriptor *desc = p; @@ -1979,12 +1980,14 @@ static int snd_usb_mixer_controls(struct usb_mixer_interface *mixer) state.oterm.name = desc->iTerminal; err = parse_audio_unit(&state, desc->bSourceID); if (err < 0) - return err; + snd_printk(KERN_WARNING "Unable to parse terminal with bSourceID %d\n", + desc->bSourceID); /* for UAC2, use the same approach to also add the clock selectors */ err = parse_audio_unit(&state, desc->bCSourceID); if (err < 0) - return err; + snd_printk(KERN_WARNING "Unable to parse clock terminal with bCSourceID %d\n", + desc->bCSourceID); } } -- 1.7.5.4

2 7

Re: [alsa-devel] Drop 2.2/2.4 kernel support in alsa-driver
by Tom Watson 13 Jul '11

13 Jul '11

Jaroslav Kysela <perex(a)perex.cz> Wrote: >Date 12.7.2011 18:12, Takashi Iwai wrote: >> [This is resent since I haven't seen my original post due to >> alsa-project.org site down. If you have already read it, please >> disregard.] >> >> Hi, >> >> in upcoming 3.0-kernel era, I believe it's high time to drop the >> support of old 2.2/2.4 kernels in alsa-driver external builds. >> Recently I changed some PCI drivers to use KBUILD_MODNAME, and this >> isn't supported by the old kernels, i.e. it's currently broken for >> these kernels. But I have no more gut to patch alsa-driver/* codes. >> >> Is there any active user of such kernels with the latest alsa-driver? >> If not, let's clean up the whole things. >> 2.2/2.4 kernel users can still use 1.0.24, and we can release 1.0.24.x >> series if really needed. > > I agree. I believe that these kernels are using minimum users now. > Almost all embeded devices are using 2.6 kernels, too. > > Jaroslav While this IS a good idea, I would suggest that ALSA take a cue from the kernel people and bump the version up a major number. It would make lots of sense that ALSA 2.0 go with kernel version 3.0. An alternative might be to skip version 2.0 and go to ALSA 3.0 to match kernel version 3.0. This won't change any code, but it would be a little clearer. That way when you drop support for older kernels, you can easily tell what version should be used. "Just a suggestion" worth the paper it is written on. -- Tom Watson tsw(a)johana.com

1 0

[alsa-devel] [PATCH] ASoC: Don't warn on low WM8994/58 AIFnCLKs
by Mark Brown 13 Jul '11

13 Jul '11

We can have valid but very low clocks in accessory detection modes. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 4 ---- 1 files changed, 0 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index e5f3f50..5af1535 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -195,10 +195,6 @@ static int configure_aif_clock(struct snd_soc_codec *codec, int aif) aif + 1, rate); } - if (rate && rate < 3000000) - dev_warn(codec->dev, "AIF%dCLK is %dHz, should be >=3MHz for optimal performance\n", - aif + 1, rate); - wm8994->aifclk[aif] = rate; snd_soc_update_bits(codec, WM8994_AIF1_CLOCKING_1 + offset, -- 1.7.5.4

1 0

[alsa-devel] [PATCH 00/10] ARM: mxs: add audio support
by Dong Aisheng 13 Jul '11

13 Jul '11

This patch series is to add audio support based on saif for mxs platform. The driver only supports playback currently. For capturing, due to hardware limitation that we have to use two saif instances to implement full duplex (playback & recording), we need to figure out a good design to fit in ASoC as the following work. In addition, we simply just support master mode since saif tx can only work in master mode due to hw limitation(rx can work in both master & slave mode). Dong Aisheng (10): ASoc: mxs: add mxs-pcm driver ASoc: mxs: add mxs-saif driver ASoc: mxs: add mxs-sgtl5000 machine driver ASoc: mxs: add asoc configuration files ARM: mxs: add saif clock ARM: mxs: add saif device ARM: mxs: add sgtl5000 i2c device ARM: mxs: add mxs-sgtl5000 device ARM: mxs: correct the using of frac div for saif ARM: mxs-dma: include <linux/dmaengine.h> arch/arm/mach-mxs/Kconfig | 1 + arch/arm/mach-mxs/clock-mx28.c | 8 +- arch/arm/mach-mxs/devices-mx28.h | 7 + arch/arm/mach-mxs/devices/Kconfig | 3 + arch/arm/mach-mxs/devices/Makefile | 1 + arch/arm/mach-mxs/devices/platform-mxs-saif.c | 60 +++ arch/arm/mach-mxs/include/mach/audio.h | 18 + arch/arm/mach-mxs/include/mach/devices-common.h | 12 + arch/arm/mach-mxs/include/mach/dma.h | 2 + arch/arm/mach-mxs/mach-mx28evk.c | 39 ++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/mxs/Kconfig | 20 + sound/soc/mxs/Makefile | 10 + sound/soc/mxs/mxs-pcm.c | 377 ++++++++++++++++ sound/soc/mxs/mxs-pcm.h | 43 ++ sound/soc/mxs/mxs-saif.c | 523 +++++++++++++++++++++++ sound/soc/mxs/mxs-saif.h | 132 ++++++ sound/soc/mxs/mxs-sgtl5000.c | 181 ++++++++ 19 files changed, 1437 insertions(+), 2 deletions(-) create mode 100644 arch/arm/mach-mxs/devices/platform-mxs-saif.c create mode 100644 arch/arm/mach-mxs/include/mach/audio.h create mode 100644 sound/soc/mxs/Kconfig create mode 100644 sound/soc/mxs/Makefile create mode 100644 sound/soc/mxs/mxs-pcm.c create mode 100644 sound/soc/mxs/mxs-pcm.h create mode 100644 sound/soc/mxs/mxs-saif.c create mode 100644 sound/soc/mxs/mxs-saif.h create mode 100644 sound/soc/mxs/mxs-sgtl5000.c

6 29

Re: [alsa-devel] Bug#631963: linux-image-3.0.0-rc4-686-pae: no sound out the rear plugs
by Ben Hutchings 13 Jul '11

13 Jul '11

On Fri, 2011-07-01 at 23:41 +0200, Toni Mueller wrote: > Hi Ben, > > On Fri, 01.07.2011 at 22:22:01 +0100, Ben Hutchings <ben(a)decadent.org.uk> wrote: > > On Fri, Jul 01, 2011 at 11:06:56PM +0200, Toni Mueller wrote: > > > On Thu, 30.06.2011 at 07:59:19 +0200, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > I guess this must be the new feature, "Auto-Mute Mode". > > > it turns out that you're right - I just had the opportunity to verify > > > that. > > > > Can we consider this closed, then? Or should that feature be > > disabled by default? > > good question - I didn't see it. Don't know offhand where to easily > find it: Upstream's changelog isn't included in the package, so... > > I'd be fine either way, but think that the feature deserves more > advertising, or, alternatively (?), being disabled by default, to > prevent bogus bug reports like mine. What do you think of this? Ben. -- Ben Hutchings All extremists should be taken out and shot.

1 0

[alsa-devel] [PATCH 0/3] OMAP4/ASoC: New McPDM driver
by Peter Ujfalusi 13 Jul '11

13 Jul '11

Hello, The following series replaces the current driver for McPDM port. The current McPDM driver design is not suitable to support both the ABE and Legacy DMA operating modes. Therefore remove most of the old driver (and mcpdm.c) and replace it with a re-designed driver to support both Legacy and ABE modes of operation. As part of the work the McPDM driver has been converted to omap device, and from now on it is going to use hwmod. Patch 2, and 3 was originated from Jorge, and Margarita, but their @ti.com mail address does not work anymore, so I took those two patch under my name, but added them as s-o-b without e-mail address. For the McPDM driver replacement, I forced git to create a complete rewrite patch to help with the review. Regards, Peter --- Misael Lopez Cruz (1): ASoC: omap-mcpdm: Replace legacy driver Peter Ujfalusi (2): OMAP: McPDM: Convert McPDM device to omap_device OMAP4: hwmod: enable mcpdm hwmod device. arch/arm/mach-omap2/devices.c | 33 ++ arch/arm/mach-omap2/omap_hwmod_44xx_data.c | 2 +- arch/arm/plat-omap/devices.c | 36 -- sound/soc/omap/Makefile | 2 +- sound/soc/omap/mcpdm.c | 470 ---------------- sound/soc/omap/mcpdm.h | 153 ----- sound/soc/omap/omap-mcpdm.c | 828 +++++++++++++++++++--------- sound/soc/omap/omap-mcpdm.h | 109 ++++ sound/soc/omap/sdp4430.c | 4 +- 9 files changed, 702 insertions(+), 935 deletions(-) delete mode 100644 sound/soc/omap/mcpdm.c delete mode 100644 sound/soc/omap/mcpdm.h rewrite sound/soc/omap/omap-mcpdm.c (42%) create mode 100644 sound/soc/omap/omap-mcpdm.h -- 1.7.6

3 15

Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by David Henningsson 13 Jul '11

13 Jul '11

On 2011-07-08 12:36, Takashi Iwai wrote: > Hi, > > just wondering how about holding a sound-BoF in LinuxCon Europe in > October? > > I know ELCE will be held at the very same time, so it may need the > adjustment of schedule. But in general what do you think? > > I know some guys will attend LPC in September, too. If the topic is > redundant again after one month, it's fine, no much need for that. > OTOH, it's a conference in Europe, so people there may have more > chance to attend (like me :-) > > An alternative is a workshop in KS2011 day 1, but I think KS suits > better for more broader topics, e.g. the discussions on the media- > notification support or the jack implementation like a thread in the > last month. I like the initiative and I'm all for having a sound BoF! Unfortunately, the timing is bad. Ubuntu's biggest event, the Ubuntu Developer Summit, is in Florida 24-28 October and my attendance is required... -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic

2 2

[alsa-devel] Drop 2.2/2.4 kernel support in alsa-driver build?
by Takashi Iwai 12 Jul '11

12 Jul '11

[This is resent since I haven't seen my original post due to alsa-project.org site down. If you have already read it, please disregard.] Hi, in upcoming 3.0-kernel era, I believe it's high time to drop the support of old 2.2/2.4 kernels in alsa-driver external builds. Recently I changed some PCI drivers to use KBUILD_MODNAME, and this isn't supported by the old kernels, i.e. it's currently broken for these kernels. But I have no more gut to patch alsa-driver/* codes. Is there any active user of such kernels with the latest alsa-driver? If not, let's clean up the whole things. 2.2/2.4 kernel users can still use 1.0.24, and we can release 1.0.24.x series if really needed. thanks, Takashi

3 2

[alsa-devel] Playback through softvol locks mixer
by Martin Bailey 12 Jul '11

12 Jul '11

Hi, I setup an ALSA plugin chain to allow individual volume-controlled mono access to a 4-channel sound card. I'm including the asound.conf only for the first channel. Whenever I open "plug:playback1" and play audio, it appears the "playback1" softvol mixer goes locked and any attempt to change its value within the same or another application results in "Operation not permitted". Can you think of something I might be missing? pcm.playback8k { type dmix ipc_key 666 ipc_perm 0666 slave { pcm "hw:0" format S16_LE channels 4 rate 8000 buffer_size 8192 period_size 1024 } bindings.0 0 bindings.1 1 bindings.2 2 bindings.3 3 } pcm.playback8k1 { type route slave { pcm "playback8k" format S16_LE channels 4 } ttable.0.0 1 } pcm.playback1 { type softvol slave.pcm "playback8k1" control.name "playback1" control.card 0 control.count 1 min_dB -51.0 max_dB 0.0 resolution 256 } ctl.playback1 { type hw card 0 }

1 0

[alsa-devel] Specifying the default buffer-size for snd-hda-intel
by Takashi Iwai 12 Jul '11

12 Jul '11

Hi, this was a feature requested some time ago: changing the default buffer size of driver, specifically HD-audio. Now I applied the patch below as a start, found in topic/hda and master branches of sound git tree. In future, we may increase the default size once when all distros set larger values than now. Together with this change, I changed alsa-driver-build.git for allowing an integer kconfig option via --with-card-options. You can pass like --with-card-options=hda-prealloc-size=2048 Takashi --- From: Takashi Iwai <tiwai(a)suse.de> Subject: [PATCH] ALSA: hda - Add Kconfig for the default buffer size Add a Kconfig entry to specify the default buffer size. Distros using PulseAudio can choose a larger value here. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 13 +++++++++++++ sound/pci/hda/hda_intel.c | 8 +++++++- 2 files changed, 20 insertions(+), 1 deletions(-) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 70762fc..1f1a4ae 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -14,6 +14,19 @@ menuconfig SND_HDA_INTEL if SND_HDA_INTEL +config SND_HDA_PREALLOC_SIZE + int "Pre-allocated buffer size for HD-audio driver" + range 0 32768 + default 64 + help + Speficies the default pre-allocated buffer-size in kB for + HD-audio driver. A larger buffer (e.g. 2048) is preferred + for systems with PulseAudio. The default 64 is chosen just + from the compatibility reason. + + Note that the pre-allocation size can be changed dynamically + via a proc file (/proc/asound/card*/pcm*/sub*/prealloc), too. + config SND_HDA_HWDEP bool "Build hwdep interface for HD-audio driver" select SND_HWDEP diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 25619cd..5ce9531 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2075,6 +2075,8 @@ static void azx_pcm_free(struct snd_pcm *pcm) } } +#define MAX_PREALLOC_SIZE (32 * 1024 * 1024) + static int azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, struct hda_pcm *cpcm) @@ -2083,6 +2085,7 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, struct snd_pcm *pcm; struct azx_pcm *apcm; int pcm_dev = cpcm->device; + unsigned int size; int s, err; if (pcm_dev >= HDA_MAX_PCMS) { @@ -2118,9 +2121,12 @@ azx_attach_pcm_stream(struct hda_bus *bus, struct hda_codec *codec, snd_pcm_set_ops(pcm, s, &azx_pcm_ops); } /* buffer pre-allocation */ + size = CONFIG_SND_HDA_PREALLOC_SIZE * 1024; + if (size > MAX_PREALLOC_SIZE) + size = MAX_PREALLOC_SIZE; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV_SG, snd_dma_pci_data(chip->pci), - 1024 * 64, 32 * 1024 * 1024); + size, MAX_PREALLOC_SIZE); return 0; } -- 1.7.6

2 2

[alsa-devel] [PATCH][INFO]: linux/sound/usb/ : add special settings for M-Audio Fast Track Pro and Quattro (24 bits, etc...)
by Guillaume Pellerin 12 Jul '11

12 Jul '11

!! ALREADY APPLIED BY TAKASHI IWAI ON LINUX 3.1 !! This patch built against Linux 2.6.39 gives M-Audio Fast Track Pro and M-Audio Quattro quirks and endpoints to boot and setup those devices with special options (digital inputs and outputs, 24 bits mode, etc...). M-Audio Audiophile quirks are just adapted to match the new global M-Audio parameters. Special configurations can be then loaded through a modprobe conf file. For example, to set the 24 bits mode on the Fast Track Pro add /etc/modprobe.d/fast_track_pro.conf : options snd_usb_audio vid=0x763 pid=0x2012 device_setup=0x08 index=1 enable=1 Here is a list of the possibilities in this example : http://files.parisson.com/debian/fast-track-pro.conf Enjoy, Cheers Alsa Team ;) Guillaume

1 0

[alsa-devel] [PATCH] ASoC: Use WM8994 FLL lock interrupt
by Mark Brown 12 Jul '11

12 Jul '11

If we have interrupts then wait for the FLL lock interrupt rather than using dead reckoning when waiting for the FLL to start. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 40 +++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm8994.h | 3 +++ 2 files changed, 42 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index d2dcaa2..f89ae47 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1635,6 +1635,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, int reg_offset, ret; struct fll_div fll; u16 reg, aif1, aif2; + unsigned long timeout; aif1 = snd_soc_read(codec, WM8994_AIF1_CLOCKING_1) & WM8994_AIF1CLK_ENA; @@ -1726,7 +1727,15 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, WM8994_FLL1_ENA | WM8994_FLL1_FRAC, reg); - msleep(5); + if (wm8994->fll_locked_irq) { + timeout = wait_for_completion_timeout(&wm8994->fll_locked[id], + msecs_to_jiffies(10)); + if (timeout == 0) + dev_warn(codec->dev, + "Timed out waiting for FLL lock\n"); + } else { + msleep(5); + } } wm8994->fll[id].in = freq_in; @@ -1744,6 +1753,14 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, return 0; } +static irqreturn_t wm8994_fll_locked_irq(int irq, void *data) +{ + struct completion *completion = data; + + complete(completion); + + return IRQ_HANDLED; +} static int opclk_divs[] = { 10, 20, 30, 40, 55, 60, 80, 120, 160 }; @@ -2879,6 +2896,9 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->pdata = dev_get_platdata(codec->dev->parent); wm8994->codec = codec; + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + init_completion(&wm8994->fll_locked[i]); + if (wm8994->pdata && wm8994->pdata->micdet_irq) wm8994->micdet_irq = wm8994->pdata->micdet_irq; else if (wm8994->pdata && wm8994->pdata->irq_base) @@ -2994,6 +3014,16 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } } + wm8994->fll_locked_irq = true; + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) { + ret = wm8994_request_irq(codec->control_data, + WM8994_IRQ_FLL1_LOCK + i, + wm8994_fll_locked_irq, "FLL lock", + &wm8994->fll_locked[i]); + if (ret != 0) + wm8994->fll_locked_irq = false; + } + /* Remember if AIFnLRCLK is configured as a GPIO. This should be * configured on init - if a system wants to do this dynamically * at runtime we can deal with that then. @@ -3179,6 +3209,9 @@ err_irq: wm8994_free_irq(codec->control_data, WM8994_IRQ_MIC1_SHRT, wm8994); if (wm8994->micdet_irq) free_irq(wm8994->micdet_irq, wm8994); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i, + &wm8994->fll_locked[i]); wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); err: @@ -3190,11 +3223,16 @@ static int wm8994_codec_remove(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); struct wm8994 *control = codec->control_data; + int i; wm8994_set_bias_level(codec, SND_SOC_BIAS_OFF); pm_runtime_disable(codec->dev); + for (i = 0; i < ARRAY_SIZE(wm8994->fll_locked); i++) + wm8994_free_irq(codec->control_data, WM8994_IRQ_FLL1_LOCK + i, + &wm8994->fll_locked[i]); + wm8994_free_irq(codec->control_data, WM8994_IRQ_DCS_DONE, &wm8994->hubs); diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 0a1db04..1ab2266 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -11,6 +11,7 @@ #include <sound/soc.h> #include <linux/firmware.h> +#include <linux/completion.h> #include "wm_hubs.h" @@ -79,6 +80,8 @@ struct wm8994_priv { int mclk[2]; int aifclk[2]; struct wm8994_fll_config fll[2], fll_suspend[2]; + struct completion fll_locked[2]; + bool fll_locked_irq; int dac_rates[2]; int lrclk_shared[2]; -- 1.7.5.4

1 0

[alsa-devel] [PATCH v4 3/4] ASoC: SAMSUNG: Modify I2S driver to support idma
by Sangbeom Kim 12 Jul '11

12 Jul '11

Previously, I2S driver only can support system dma. In this patch, i2s driver can support internal dma too. IDMA h/w configuration is initialized on idma.c Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 1568eea..c086b78 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -21,6 +21,7 @@ #include <plat/audio.h> #include "dma.h" +#include "idma.h" #include "i2s.h" #include "i2s-regs.h" @@ -60,6 +61,7 @@ struct i2s_dai { /* DMA parameters */ struct s3c_dma_params dma_playback; struct s3c_dma_params dma_capture; + struct s3c_dma_params idma_playback; u32 quirks; u32 suspend_i2smod; u32 suspend_i2scon; @@ -877,6 +879,10 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) if (i2s->quirks & QUIRK_NEED_RSTCLR) writel(CON_RSTCLR, i2s->addr + I2SCON); + if (i2s->quirks & QUIRK_SEC_DAI) + idma_reg_addr_init((void *)i2s->addr, + i2s->sec_dai->idma_playback.dma_addr); + probe_exit: /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; @@ -1077,6 +1083,7 @@ static __devinit int samsung_i2s_probe(struct platform_device *pdev) sec_dai->dma_playback.dma_size = 4; sec_dai->base = regs_base; sec_dai->quirks = quirks; + sec_dai->idma_playback.dma_addr = i2s_cfg->idma_addr; sec_dai->pri_dai = pri_dai; pri_dai->sec_dai = sec_dai; } -- 1.7.1

4 7

[alsa-devel] [PATCH 1/2] ASoC: Implement DC servo completion IRQ handling for wm_hubs devices
by Mark Brown 12 Jul '11

12 Jul '11

The individual devices should set the flag dcs_done_irq in the hubs shared data structure to indicate that they will flag the interrupt by calling wm_hubs_dcs_done(). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 34 +++++++++++++++++++++++++++++----- sound/soc/codecs/wm_hubs.h | 8 ++++++++ 2 files changed, 37 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 7e60e22..5c2d565 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -63,9 +63,11 @@ static const struct soc_enum speaker_mode = static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); unsigned int reg; int count = 0; unsigned int val; + unsigned long timeout; val = op | WM8993_DCS_ENA_CHAN_0 | WM8993_DCS_ENA_CHAN_1; @@ -74,18 +76,37 @@ static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) dev_dbg(codec->dev, "Waiting for DC servo...\n"); - do { - count++; - msleep(1); + if (hubs->dcs_done_irq) { + timeout = wait_for_completion_timeout(&hubs->dcs_done, + msecs_to_jiffies(500)); + if (timeout == 0) + dev_warn(codec->dev, "No DC servo interrupt\n"); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); - dev_dbg(codec->dev, "DC servo: %x\n", reg); - } while (reg & op && count < 400); + } else { + do { + count++; + msleep(1); + reg = snd_soc_read(codec, WM8993_DC_SERVO_0); + dev_dbg(codec->dev, "DC servo: %x\n", reg); + } while (reg & op && count < 400); + } if (reg & op) dev_err(codec->dev, "Timed out waiting for DC Servo %x\n", op); } +irqreturn_t wm_hubs_dcs_done(int irq, void *data) +{ + struct wm_hubs_data *hubs = data; + + complete(&hubs->dcs_done); + + return IRQ_HANDLED; +} +EXPORT_SYMBOL_GPL(wm_hubs_dcs_done); + /* * Startup calibration of the DC servo */ @@ -863,8 +884,11 @@ EXPORT_SYMBOL_GPL(wm_hubs_add_analogue_controls); int wm_hubs_add_analogue_routes(struct snd_soc_codec *codec, int lineout1_diff, int lineout2_diff) { + struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; + init_completion(&hubs->dcs_done); + snd_soc_dapm_add_routes(dapm, analogue_routes, ARRAY_SIZE(analogue_routes)); diff --git a/sound/soc/codecs/wm_hubs.h b/sound/soc/codecs/wm_hubs.h index 0d290d2..676b125 100644 --- a/sound/soc/codecs/wm_hubs.h +++ b/sound/soc/codecs/wm_hubs.h @@ -14,6 +14,9 @@ #ifndef _WM_HUBS_H #define _WM_HUBS_H +#include <linux/completion.h> +#include <linux/interrupt.h> + struct snd_soc_codec; extern const unsigned int wm_hubs_spkmix_tlv[]; @@ -28,6 +31,9 @@ struct wm_hubs_data { bool class_w; u16 class_w_dcs; + + bool dcs_done_irq; + struct completion dcs_done; }; extern int wm_hubs_add_analogue_controls(struct snd_soc_codec *); @@ -38,4 +44,6 @@ extern int wm_hubs_handle_analogue_pdata(struct snd_soc_codec *, int jd_scthr, int jd_thr, int micbias1_lvl, int micbias2_lvl); +extern irqreturn_t wm_hubs_dcs_done(int irq, void *data); + #endif -- 1.7.5.4

1 1

Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by Liam Girdwood 12 Jul '11

12 Jul '11

On Fri, 2011-07-08 at 12:36 +0200, Takashi Iwai wrote: > Hi, > > just wondering how about holding a sound-BoF in LinuxCon Europe in > October? > > I know ELCE will be held at the very same time, so it may need the > adjustment of schedule. But in general what do you think? > I don't think I'll be at ELCE but Peter will be attending from TI. > I know some guys will attend LPC in September, too. If the topic is > redundant again after one month, it's fine, no much need for that. > OTOH, it's a conference in Europe, so people there may have more > chance to attend (like me :-) > Mark and I are looking at another ASoC/Audio workshop at LPC, to go over progress and planning for the next development phase. Imo, this would likely complement a BOF at LinuxCon since a good proportion of the BOF audience will probably be different from LPC with different hardware, requirements, users, etc. > An alternative is a workshop in KS2011 day 1, but I think KS suits > better for more broader topics, e.g. the discussions on the media- > notification support or the jack implementation like a thread in the > last month. > Regards Liam

3 2

[alsa-devel] What is correct way to put conditional stuff in ASoC codec driver?
by Ashish Chavan 11 Jul '11

11 Jul '11

Hi, I am working on updating ASoC codec driver for Dialog's DA7210 codec (sound/soc/codec/da7210.c). This update would be a major functionality update in nature. In fact we already have a feature complete driver for this codec but there are two main issues with that driver, (1) It was developed(and tested) for a quite old kernel version (2.6.28) (2) It was written in custom(non standard) way and not suitable for direct submission to kernel Our final goal is to pull in all missing features from this custom driver in to existing mainline driver. I have already created few patches for basic functions but I am bit confused at one point. Some of the features supported by this codec have inter dependency, e.g. ALC can be used only if NOISE SUPRESSION is disabled. So ideally, all controls related to ALC should be either disabled or not added at all, if NOISE SUPRESSION to be used. There are few other features having similar kind of dependency. I just want to know what is the correct way to handle this in ASoC codec driver? Looking at the existing codec drivers, it seems that having conditional defines is not common here. As We just want to support static configuration of such features, is it a good idea to add sub menu options in driver's Kconfig to enable/disable such features and use them within code? Any pointer(s) to existing code are most welcome. Thanks, -- Ashish ---------------------------------------------------------------------------- || Linux Is User Friendly, It's Just Selective About Who It's Friends Are || ----------------------------------------------------------------------------

2 11

[alsa-devel] [PATCH] Acer aspire 3830TG and Conexant 506c/20588
by David Henningsson 11 Jul '11

11 Jul '11

The new Conexant 5066 auto-parser is barely finished and already we might need a quirk system for it... Anyway, in this bug, the internal speaker is not working (and was not working before the auto-parser either). I'm attaching two patches that are quite simple, and at least the first one (add ID 506c) I think should be applied right away. The problem: The internal speaker is at node 0x1f and you need to turn on EAPD on node 0x1b for the speaker to sound. Now I'm not sure how to fix this in the best way. An "init verb" quirk, or a "use EAPD from this pin" quirk? Try to copy-paste realtek's quirk system, and if so, refactor it into hda_codec.c, or write something new? What are your ideas? Alsa-info reference: https://launchpadlibrarian.net/73966502/alsa-info Bug reference: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/783582 -- David Henningsson http://launchpad.net/~diwic

2 10

[alsa-devel] [RFC PATCH] ARM ASoC: add sound driver for imx27pdk using mc13783 codec
by Jürgen Lambrecht 11 Jul '11

11 Jul '11

- Started from sound/soc/imx/phycore-mc13783.c - Renamed to general "imx" because also valid for imx27pdk (3ds) and other 3ds platforms. Or should there be a file per platform? - Fixed some issues (add names, add read/write functions, changed late_initcall to module_init, configure clock input pin) - Add debug printing and comments. - Result: * soundcard detection; * after creating /dev entries, aplay runs; * both 'aplay song.wav' and 'madplay song.mp3' give the same error: a time-out waiting for DMA (ALSA pcm_lib.c:1802: playback write error (DMA or IRQ trouble?)) * this driver uses the DMA method; the FIQ method gives the same error Signed-off-by: J�rgen Lambrecht <J.Lambrecht(a)televic.com> --- sound/soc/imx/Kconfig | 20 +++-- sound/soc/imx/Makefile | 2 + sound/soc/imx/imx-mc13783.c | 175 +++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 190 insertions(+), 7 deletions(-) create mode 100644 sound/soc/imx/imx-mc13783.c diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index d8f130d..2f57f12 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -11,20 +11,19 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC -config SND_MXC_SOC_SSI - tristate - config SND_MXC_SOC_FIQ tristate config SND_MXC_SOC_MX2 tristate +config SND_SOC_IMX_MC13783 + tristate + config SND_MXC_SOC_WM1133_EV1 tristate "Audio on the the i.MX31ADS with WM1133-EV1 fitted" depends on MACH_MX31ADS_WM1133_EV1 && EXPERIMENTAL select SND_SOC_WM8350 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable support for audio on the i.MX31ADS with the WM1133-EV1 @@ -34,7 +33,6 @@ config SND_SOC_MX27VIS_AIC32X4 tristate "SoC audio support for Visstrim M10 boards" depends on MACH_IMX27_VISSTRIM_M10 select SND_SOC_TVL320AIC32X4 - select SND_MXC_SOC_SSI select SND_MXC_SOC_MX2 help Say Y if you want to add support for SoC audio on Visstrim SM10 @@ -44,7 +42,6 @@ config SND_SOC_PHYCORE_AC97 tristate "SoC Audio support for Phytec phyCORE (and phyCARD) boards" depends on MACH_PCM043 || MACH_PCA100 select SND_SOC_WM9712 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Say Y if you want to add support for SoC audio on Phytec phyCORE @@ -57,10 +54,19 @@ config SND_SOC_EUKREA_TLV320 || MACH_EUKREA_MBIMXSD35_BASEBOARD \ || MACH_EUKREA_MBIMXSD51_BASEBOARD select SND_SOC_TLV320AIC23 - select SND_MXC_SOC_SSI select SND_MXC_SOC_FIQ help Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX27_MC13783 + tristate "SoC Audio support for i.MX27 platforms with a MC13783 codec" + depends on MACH_MX27_3DS + select SND_SOC_MC13783 + select SND_MXC_SOC_MX2 + select SND_SOC_IMX_MC13783 + help + Say Y if you want to add support for SoC audio on i.MX27 platforms + with a MC13783 codec (based on Freescale's imx27pdk kit) + endif # SND_IMX_SOC diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index d6d609b..d86e5b8 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -12,8 +12,10 @@ snd-soc-eukrea-tlv320-objs := eukrea-tlv320.o snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o +snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o +obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/imx/imx-mc13783.c b/sound/soc/imx/imx-mc13783.c new file mode 100644 index 0000000..4274f84 --- /dev/null +++ b/sound/soc/imx/imx-mc13783.c @@ -0,0 +1,175 @@ +/* + * imx-mc13783.c -- SoC audio for imx processor with mx13783 co-processor + * + * Copyright 2009 Sascha Hauer, Pengutronix <s.hauer(a)pengutronix.de> + * Copyright 2011 J�rgen Lambrecht, Televic <J.Lambrecht(a)televic.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dai.h> +#include <asm/mach-types.h> +#include "../codecs/mc13783.h" +#include "imx-ssi.h" + +static struct snd_soc_card imx_mc13783_audio; + +#define FMT_PLAYBACK (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) +#define FMT_CAPTURE (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ + SND_SOC_DAIFMT_CBM_CFM) + +static int imx_mc13783_audio_hifi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + int ret; + + /* set codec and cpu DAI configuration */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = snd_soc_dai_set_fmt(codec_dai, FMT_PLAYBACK); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, FMT_PLAYBACK); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + } else { + ret = snd_soc_dai_set_fmt(codec_dai, FMT_CAPTURE); + if (ret) { + pr_err("%s: failed set codec dai format\n", __func__); + return ret; + } + ret = snd_soc_dai_set_fmt(cpu_dai, FMT_CAPTURE); + if (ret) { + pr_err("%s: failed set cpu dai format\n", __func__); + return ret; + } + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0, 0, 4, 32); /* args are not used in mc13783 */ + if (ret) + return ret; + } +//ok: checked; we use clia input pin; last arg is not used + ret = snd_soc_dai_set_sysclk(codec_dai, MC13783_CLK_CLIA, 26000000, SND_SOC_CLOCK_OUT); + if (ret) { + pr_err("%s: failed setting codec sysclk\n", __func__); + return ret; + } + + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x0, 0xfffffffc, 4, 32); + if (ret) + return ret; + +//from eukrea-tlv320.c and mx27vis-aic32x4.c + ret = snd_soc_dai_set_sysclk(cpu_dai, IMX_SSP_SYS_CLK, 0, + SND_SOC_CLOCK_IN); + if (ret) { + pr_err("can't set CPU system clock IMX_SSP_SYS_CLK\n"); + return ret; + } + + return 0; +} + +static int imx_mc13783_audio_hifi_hw_free(struct snd_pcm_substream *substream) +{ + return 0; +} + +static struct snd_soc_ops imx_mc13783_audio_hifi_ops = { + .hw_params = imx_mc13783_audio_hifi_hw_params, + .hw_free = imx_mc13783_audio_hifi_hw_free, +}; + +static int imx_mc13783_audio_probe(struct platform_device *pdev) +{ + return 0; +} + +static int imx_mc13783_audio_remove(struct platform_device *pdev) +{ + return 0; +} + +static struct snd_soc_dai_link imx_dai_mc13783[] = { + { + .name = "MC13783 Playback", + .stream_name = "Playback", + .codec_dai_name = "mc13783-hifi", + .platform_name = "imx-pcm-audio.0", + .codec_name = "mc13783-codec", + .cpu_dai_name = "imx-ssi.0", + .ops = &imx_mc13783_audio_hifi_ops, + }, { + .name = "MC13783 Capture", + .stream_name = "Capture", + .codec_dai_name = "mc13783-hifi", + .platform_name = "imx-pcm-audio.0", + .codec_name = "mc13783-codec", + .cpu_dai_name = "imx-ssi.0", + .ops = &imx_mc13783_audio_hifi_ops, + }, +}; + +static struct snd_soc_card imx_mc13783_audio = { + .name = "iMX-mc13783-audio", + .probe = imx_mc13783_audio_probe, + .remove = imx_mc13783_audio_remove, + .dai_link = imx_dai_mc13783, + .num_links = ARRAY_SIZE(imx_dai_mc13783), +}; + +static struct platform_device *imx_mc13783_snd_device; + +static int __init imx_mc13783_audio_init(void) +{ + int ret; + + pr_info("IMX MC13783 Sound Card\n"); +/* Possibly add a check for IMX and MC13783 */ +/* if (!machine_is_pcm038() && !machine_is_pcm037()) */ +/* * return happy. We might run on a totally different machine * */ +/* return 0; */ + + imx_mc13783_snd_device = platform_device_alloc("soc-audio", -1); + if (!imx_mc13783_snd_device) + return -ENOMEM; + + platform_set_drvdata(imx_mc13783_snd_device, &imx_mc13783_audio); + ret = platform_device_add(imx_mc13783_snd_device); + + if (ret) { + printk(KERN_ERR "ASoC: Platform device allocation failed\n"); + platform_device_put(imx_mc13783_snd_device); + } + + return ret; +} + +static void __exit imx_mc13783_audio_exit(void) +{ + platform_device_unregister(imx_mc13783_snd_device); +} + +//late_initcall(imx_mc13783_audio_init); +module_init(imx_mc13783_audio_init); +module_exit(imx_mc13783_audio_exit); + +MODULE_AUTHOR("Sascha Hauer <s.hauer(a)pengutronix.de>"); +MODULE_DESCRIPTION("iMX ALSA SoC driver"); +MODULE_LICENSE("GPL"); -- 1.7.1

4 4

[PATCH 1/1] sgtl5000: fix i2c r/w issue add neglect VDDIO and VDDI=
by Dong Aisheng 11 Jul '11

11 Jul '11

O This is caused by sgtl5000 using register address step is 2, and snd-soc-core can't handle this as we expect, so we have to fill the register cache by reading register out when initialization instead of providing a default value array. Signed-off-by: Dong Aisheng <b29396(a)freescale.com> --- sound/soc/codecs/sgtl5000.c | 159 ++++++++++++++-------------------------= ---- 1 files changed, 52 insertions(+), 107 deletions(-) diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index ff29380..c29f153 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -33,87 +33,18 @@ #define SGTL5000_DAP_REG_OFFSET 0x0100 #define SGTL5000_MAX_REG_OFFSET 0x013A =20 -/* default value of sgtl5000 registers except DAP */ -static const u16 sgtl5000_regs[SGTL5000_MAX_REG_OFFSET >> 1] =3D { - 0xa011, /* 0x0000, CHIP_ID. 11 stand for revison 17 */ - 0x0000, /* 0x0002, CHIP_DIG_POWER. */ - 0x0008, /* 0x0004, CHIP_CKL_CTRL */ - 0x0010, /* 0x0006, CHIP_I2S_CTRL */ - 0x0000, /* 0x0008, reserved */ - 0x0008, /* 0x000A, CHIP_SSS_CTRL */ - 0x0000, /* 0x000C, reserved */ - 0x020c, /* 0x000E, CHIP_ADCDAC_CTRL */ - 0x3c3c, /* 0x0010, CHIP_DAC_VOL */ - 0x0000, /* 0x0012, reserved */ - 0x015f, /* 0x0014, CHIP_PAD_STRENGTH */ - 0x0000, /* 0x0016, reserved */ - 0x0000, /* 0x0018, reserved */ - 0x0000, /* 0x001A, reserved */ - 0x0000, /* 0x001E, reserved */ - 0x0000, /* 0x0020, CHIP_ANA_ADC_CTRL */ - 0x1818, /* 0x0022, CHIP_ANA_HP_CTRL */ - 0x0111, /* 0x0024, CHIP_ANN_CTRL */ - 0x0000, /* 0x0026, CHIP_LINREG_CTRL */ - 0x0000, /* 0x0028, CHIP_REF_CTRL */ - 0x0000, /* 0x002A, CHIP_MIC_CTRL */ - 0x0000, /* 0x002C, CHIP_LINE_OUT_CTRL */ - 0x0404, /* 0x002E, CHIP_LINE_OUT_VOL */ - 0x7060, /* 0x0030, CHIP_ANA_POWER */ - 0x5000, /* 0x0032, CHIP_PLL_CTRL */ - 0x0000, /* 0x0034, CHIP_CLK_TOP_CTRL */ - 0x0000, /* 0x0036, CHIP_ANA_STATUS */ - 0x0000, /* 0x0038, reserved */ - 0x0000, /* 0x003A, CHIP_ANA_TEST2 */ - 0x0000, /* 0x003C, CHIP_SHORT_CTRL */ - 0x0000, /* reserved */ -}; - -/* default value of dap registers */ -static const u16 sgtl5000_dap_regs[] =3D { - 0x0000, /* 0x0100, DAP_CONTROL */ - 0x0000, /* 0x0102, DAP_PEQ */ - 0x0040, /* 0x0104, DAP_BASS_ENHANCE */ - 0x051f, /* 0x0106, DAP_BASS_ENHANCE_CTRL */ - 0x0000, /* 0x0108, DAP_AUDIO_EQ */ - 0x0040, /* 0x010A, DAP_SGTL_SURROUND */ - 0x0000, /* 0x010C, DAP_FILTER_COEF_ACCESS */ - 0x0000, /* 0x010E, DAP_COEF_WR_B0_MSB */ - 0x0000, /* 0x0110, DAP_COEF_WR_B0_LSB */ - 0x0000, /* 0x0112, reserved */ - 0x0000, /* 0x0114, reserved */ - 0x002f, /* 0x0116, DAP_AUDIO_EQ_BASS_BAND0 */ - 0x002f, /* 0x0118, DAP_AUDIO_EQ_BAND0 */ - 0x002f, /* 0x011A, DAP_AUDIO_EQ_BAND2 */ - 0x002f, /* 0x011C, DAP_AUDIO_EQ_BAND3 */ - 0x002f, /* 0x011E, DAP_AUDIO_EQ_TREBLE_BAND4 */ - 0x8000, /* 0x0120, DAP_MAIN_CHAN */ - 0x0000, /* 0x0122, DAP_MIX_CHAN */ - 0x0510, /* 0x0124, DAP_AVC_CTRL */ - 0x1473, /* 0x0126, DAP_AVC_THRESHOLD */ - 0x0028, /* 0x0128, DAP_AVC_ATTACK */ - 0x0050, /* 0x012A, DAP_AVC_DECAY */ - 0x0000, /* 0x012C, DAP_COEF_WR_B1_MSB */ - 0x0000, /* 0x012E, DAP_COEF_WR_B1_LSB */ - 0x0000, /* 0x0130, DAP_COEF_WR_B2_MSB */ - 0x0000, /* 0x0132, DAP_COEF_WR_B2_LSB */ - 0x0000, /* 0x0134, DAP_COEF_WR_A1_MSB */ - 0x0000, /* 0x0136, DAP_COEF_WR_A1_LSB */ - 0x0000, /* 0x0138, DAP_COEF_WR_A2_MSB */ - 0x0000, /* 0x013A, DAP_COEF_WR_A2_LSB */ -}; - /* regulator supplies for sgtl5000, VDDD is an optional external supply */ enum sgtl5000_regulator_supplies { - VDDA, - VDDIO, +// VDDA, +// VDDIO, VDDD, SGTL5000_SUPPLY_NUM }; =20 /* vddd is optional supply */ static const char *supply_names[SGTL5000_SUPPLY_NUM] =3D { - "VDDA", - "VDDIO", +// "VDDA", +// "VDDIO", "VDDD" }; =20 @@ -780,6 +711,11 @@ static int ldo_regulator_is_enabled(struct regulator_d= ev *dev) return ldo->enabled; } =20 +/* + * enable internal VDDD power supply. Since register + * cache not fill yet, we have to use hw_read and write + * instead of snd_soc_read and snd_soc_write. + */ static int ldo_regulator_enable(struct regulator_dev *dev) { struct ldo_regulator *ldo =3D rdev_get_drvdata(dev); @@ -797,17 +733,17 @@ static int ldo_regulator_enable(struct regulator_dev = *dev) ldo->voltage =3D (1600 - reg * 50) * 1000; =20 /* set voltage to register */ - snd_soc_update_bits(codec, SGTL5000_CHIP_LINREG_CTRL, - (0x1 << 4) - 1, reg); + codec->write(codec, SGTL5000_CHIP_LINREG_CTRL, reg); =20 - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINEREG_D_POWERUP, - SGTL5000_LINEREG_D_POWERUP); + reg =3D codec->hw_read(codec, SGTL5000_CHIP_ANA_POWER); + reg |=3D SGTL5000_LINEREG_D_POWERUP; + codec->write(codec, SGTL5000_CHIP_ANA_POWER, reg); =20 + reg &=3D ~SGTL5000_LINREG_SIMPLE_POWERUP; /* when internal ldo enabled, simple digital power can be disabled */ - snd_soc_update_bits(codec, SGTL5000_CHIP_ANA_POWER, - SGTL5000_LINREG_SIMPLE_POWERUP, - 0); + codec->write(codec, SGTL5000_CHIP_ANA_POWER, reg); + + udelay(10); =20 ldo->enabled =3D 1; return 0; @@ -1022,13 +958,11 @@ static int sgtl5000_suspend(struct snd_soc_codec *co= dec, pm_message_t state) static int sgtl5000_restore_regs(struct snd_soc_codec *codec) { u16 *cache =3D codec->reg_cache; - int i; - int regular_regs =3D SGTL5000_CHIP_SHORT_CTRL >> 1; + int reg; + int step =3D codec->driver->reg_cache_step; =20 /* restore regular registers */ - for (i =3D 0; i < regular_regs; i++) { - int reg =3D i << 1; - + for (reg =3D 0; reg < SGTL5000_CHIP_SHORT_CTRL; reg +=3D step) { /* this regs depends on the others */ if (reg =3D=3D SGTL5000_CHIP_ANA_POWER || reg =3D=3D SGTL5000_CHIP_CLK_CTRL || @@ -1037,35 +971,32 @@ static int sgtl5000_restore_regs(struct snd_soc_code= c *codec) reg =3D=3D SGTL5000_CHIP_CLK_CTRL) continue; =20 - snd_soc_write(codec, reg, cache[i]); + snd_soc_write(codec, reg, cache[reg]); } =20 /* restore dap registers */ - for (i =3D SGTL5000_DAP_REG_OFFSET >> 1; - i < SGTL5000_MAX_REG_OFFSET >> 1; i++) { - int reg =3D i << 1; - - snd_soc_write(codec, reg, cache[i]); - } + for (reg =3D SGTL5000_DAP_REG_OFFSET; + reg < SGTL5000_MAX_REG_OFFSET; reg +=3D step) + snd_soc_write(codec, reg, cache[reg]); =20 /* * restore power and other regs according * to set_power() and set_clock() */ snd_soc_write(codec, SGTL5000_CHIP_LINREG_CTRL, - cache[SGTL5000_CHIP_LINREG_CTRL >> 1]); + cache[SGTL5000_CHIP_LINREG_CTRL]); =20 snd_soc_write(codec, SGTL5000_CHIP_ANA_POWER, - cache[SGTL5000_CHIP_ANA_POWER >> 1]); + cache[SGTL5000_CHIP_ANA_POWER]); =20 snd_soc_write(codec, SGTL5000_CHIP_CLK_CTRL, - cache[SGTL5000_CHIP_CLK_CTRL >> 1]); + cache[SGTL5000_CHIP_CLK_CTRL]); =20 snd_soc_write(codec, SGTL5000_CHIP_REF_CTRL, - cache[SGTL5000_CHIP_REF_CTRL >> 1]); + cache[SGTL5000_CHIP_REF_CTRL]); =20 snd_soc_write(codec, SGTL5000_CHIP_LINE_OUT_CTRL, - cache[SGTL5000_CHIP_LINE_OUT_CTRL >> 1]); + cache[SGTL5000_CHIP_LINE_OUT_CTRL]); return 0; } =20 @@ -1105,12 +1036,12 @@ static int sgtl5000_set_power_regs(struct snd_soc_c= odec *codec) int vag; struct sgtl5000_priv *sgtl5000 =3D snd_soc_codec_get_drvdata(codec); =20 - vdda =3D regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); - vddio =3D regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); +// vdda =3D regulator_get_voltage(sgtl5000->supplies[VDDA].consumer); +// vddio =3D regulator_get_voltage(sgtl5000->supplies[VDDIO].consumer); vddd =3D regulator_get_voltage(sgtl5000->supplies[VDDD].consumer); =20 - vdda =3D vdda / 1000; - vddio =3D vddio / 1000; + vdda =3D 3300; + vddio =3D 3300; vddd =3D vddd / 1000; =20 if (vdda <=3D 0 || vddio <=3D 0 || vddd < 0) { @@ -1335,6 +1266,22 @@ err_regulator_free: =20 } =20 +static int sgtl5000_fill_reg_cache(struct snd_soc_codec *codec) +{ + int reg; + int step =3D codec->driver->reg_cache_step; + u16 *cache =3D codec->reg_cache; + + for (reg =3D SGTL5000_DAP_REG_OFFSET; + reg <=3D SGTL5000_MAX_REG_OFFSET; reg +=3D step) + cache[reg] =3D codec->hw_read(codec, reg); + + for (reg =3D 0; reg <=3D SGTL5000_CHIP_SHORT_CTRL; reg +=3D step) + cache[reg] =3D codec->hw_read(codec, reg); + + return 0; +} + static int sgtl5000_probe(struct snd_soc_codec *codec) { int ret; @@ -1351,6 +1298,8 @@ static int sgtl5000_probe(struct snd_soc_codec *codec= ) if (ret) return ret; =20 + sgtl5000_fill_reg_cache(codec); + /* power up sgtl5000 */ ret =3D sgtl5000_set_power_regs(codec); if (ret) @@ -1443,10 +1392,9 @@ static struct snd_soc_codec_driver sgtl5000_driver = =3D { .suspend =3D sgtl5000_suspend, .resume =3D sgtl5000_resume, .set_bias_level =3D sgtl5000_set_bias_level, - .reg_cache_size =3D ARRAY_SIZE(sgtl5000_regs), + .reg_cache_size =3D SGTL5000_MAX_REG_OFFSET, .reg_word_size =3D sizeof(u16), .reg_cache_step =3D 2, - .reg_cache_default =3D sgtl5000_regs, .volatile_register =3D sgtl5000_volatile_register, }; =20 @@ -1466,9 +1414,6 @@ static __devinit int sgtl5000_i2c_probe(struct i2c_cl= ient *client, * at init phase makes life easy. * FIXME: should we drop 'const' of sgtl5000_regs? */ - memcpy((void *)(&sgtl5000_regs[0] + (SGTL5000_DAP_REG_OFFSET >> 1)), - sgtl5000_dap_regs, - SGTL5000_MAX_REG_OFFSET - SGTL5000_DAP_REG_OFFSET); =20 i2c_set_clientdata(client, sgtl5000); =20 --=20 1.7.0.4

1 0

[alsa-devel] [PATCH] ASoC: Use late enable handling for direct voice, speaker and headphone
by Mark Brown 11 Jul '11

11 Jul '11

This ensures appropriate clocking for bypass paths to speaker and headphone and direct voice paths on affected revisions. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8993.c | 2 +- sound/soc/codecs/wm8994.c | 30 +++++++++++++++++++++--------- sound/soc/codecs/wm_hubs.c | 3 --- 3 files changed, 22 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 03af7ad..6e85b88 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -876,7 +876,7 @@ SND_SOC_DAPM_MIXER("SPKL", WM8993_POWER_MANAGEMENT_3, 8, 0, left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), SND_SOC_DAPM_MIXER("SPKR", WM8993_POWER_MANAGEMENT_3, 9, 0, right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), - +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), }; static const struct snd_soc_dapm_route routes[] = { diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 8d1258c..96ea69b 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1146,13 +1146,33 @@ SND_SOC_DAPM_PGA_E("Late DAC2L Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_PGA_E("Late DAC2R Enable PGA", SND_SOC_NOPM, 0, 0, NULL, 0, late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_PGA_E("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), + +SND_SOC_DAPM_MIXER_E("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer), + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MIXER_E("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer), + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), +SND_SOC_DAPM_MUX_E("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux, + late_enable_ev, SND_SOC_DAPM_PRE_PMU), SND_SOC_DAPM_POST("Late Disable PGA", late_disable_ev) }; static const struct snd_soc_dapm_widget wm8994_lateclk_widgets[] = { SND_SOC_DAPM_SUPPLY("AIF1CLK", WM8994_AIF1_CLOCKING_1, 0, 0, NULL, 0), -SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0) +SND_SOC_DAPM_SUPPLY("AIF2CLK", WM8994_AIF2_CLOCKING_1, 0, 0, NULL, 0), +SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), +SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, + left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), +SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, + right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), +SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), +SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), }; static const struct snd_soc_dapm_widget wm8994_dac_revd_widgets[] = { @@ -1283,14 +1303,6 @@ SND_SOC_DAPM_ADC("DMIC1R", NULL, WM8994_POWER_MANAGEMENT_4, 2, 0), SND_SOC_DAPM_ADC("ADCL", NULL, SND_SOC_NOPM, 1, 0), SND_SOC_DAPM_ADC("ADCR", NULL, SND_SOC_NOPM, 0, 0), -SND_SOC_DAPM_MUX("Left Headphone Mux", SND_SOC_NOPM, 0, 0, &hpl_mux), -SND_SOC_DAPM_MUX("Right Headphone Mux", SND_SOC_NOPM, 0, 0, &hpr_mux), - -SND_SOC_DAPM_MIXER("SPKL", WM8994_POWER_MANAGEMENT_3, 8, 0, - left_speaker_mixer, ARRAY_SIZE(left_speaker_mixer)), -SND_SOC_DAPM_MIXER("SPKR", WM8994_POWER_MANAGEMENT_3, 9, 0, - right_speaker_mixer, ARRAY_SIZE(right_speaker_mixer)), - SND_SOC_DAPM_POST("Debug log", post_ev), }; diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 2d6c88b..7e60e22 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -598,9 +598,6 @@ SND_SOC_DAPM_MIXER("IN2L PGA", WM8993_POWER_MANAGEMENT_2, 7, 0, SND_SOC_DAPM_MIXER("IN2R PGA", WM8993_POWER_MANAGEMENT_2, 5, 0, in2r_pga, ARRAY_SIZE(in2r_pga)), -/* Dummy widgets to represent differential paths */ -SND_SOC_DAPM_PGA("Direct Voice", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("MIXINL", WM8993_POWER_MANAGEMENT_2, 9, 0, mixinl, ARRAY_SIZE(mixinl)), SND_SOC_DAPM_MIXER("MIXINR", WM8993_POWER_MANAGEMENT_2, 8, 0, -- 1.7.5.4

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[alsa-devel] [PATCH 0/5] treewide: Add missing breaks in switch/case
by Joe Perches 10 Jul '11

10 Jul '11

Found by grep. Uncompiled/untested. Joe Perches (5): gpio-tps65910.c: Add missing breaks in switch/case tps65910-irq: Add missing breaks in switch/case ath5k: Add missing breaks in switch/case fusb300_udc: Add missing breaks in switch/case dbri: Add missing breaks in switch/case drivers/gpio/gpio-tps65910.c | 2 ++ drivers/mfd/tps65910-irq.c | 2 ++ drivers/net/wireless/ath/ath5k/desc.c | 3 +++ drivers/usb/gadget/fusb300_udc.c | 3 +++ sound/sparc/dbri.c | 5 +++++ 5 files changed, 15 insertions(+), 0 deletions(-) -- 1.7.6.131.g99019

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[alsa-devel] max channels of 3 jacks 6 channels motherboard using model=auto realtek codec
by Raymond Yau 10 Jul '11

10 Jul '11

Refer to commit ce764ab22e40a046065c9417ee5f04ad2a816ac1 ALSA: hda - Add channel-mode support to Realtek auto-parser It seem that max channel is six instead of two when the channel mode is 2ch for those 3 jacks motherboard using hda-emu and alsa-info from http://thread.gmane.org/gmane.linux.alsa.devel/86816 > PCM 0 p 44100 2 16 Open PCM ALC887 Analog for play Available PCM parameters: channels: 2/6 formats: S16_LE S32_LE rates: 44100 48000 96000 192000 Prepare PCM, rate=44100, channels=2, format=16 bits because spec->multiout.max_channels = spec->multiout.num_dacs * 2; is assigned after alc_auto_add_multi_channel_mode() increase spec->multiout.num_dacs from 1 to 3 spec->multiout.num_dacs = num_pins + 1;

2 2

[alsa-devel] [PATCH] ASoC: OMAP4: HDMI: Make stricter dependency on OMAP2_DSS
by Ricardo Neri 09 Jul '11

09 Jul '11

When building DSS HDMI driver with audio support, there is dependency on SND_SOC to register the ASoC audio codec. Also, SND_OMAP_SOC_OMAP4_HDMI has dependency on OMAP2_DSS. Then, to avoid build breakages, SND_OMAP_SOC_OMAP4_HDMI, is restricted to be available only when OMAP2_DSS AND SND_SOC have the same build type. Also, dependency on ARCH_OMAP4 is removed, as it is checked indirectly by OMAP4_DSS_HDMI. Reported-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Ricardo Neri <ricardo.neri(a)ti.com> --- sound/soc/omap/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index e8ca5be..ffbd83f 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -113,7 +113,7 @@ config SND_OMAP_SOC_SDP4430 config SND_OMAP_SOC_OMAP4_HDMI tristate "SoC Audio support for Texas Instruments OMAP4 HDMI" - depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS && ARCH_OMAP4 + depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && (OMAP2_DSS = SND_SOC) select SND_OMAP_SOC_HDMI help Say Y if you want to add support for SoC HDMI audio on Texas Instruments -- 1.7.4.1

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[alsa-devel] ALSA become "Segmentation fault" on current linus tree
by kuninori.morimoto.gx＠renesas.com 09 Jul '11

09 Jul '11

Dear Liam, Mark Currect linus tree become "Segmentation fault" when I use aplay. # aplay xxx.wav Segmentation fault I used "git bisect" and 1st bad commit was below. I'm using sh7372 mackerel board. commit 22de71ba03311cdc1063757c50a1488cb90a1fca Author: Liam Girdwood <lrg(a)ti.com> Date: Thu May 12 16:14:04 2011 +0100 ASoC: core - allow ASoC more flexible machine name Allow ASoC machine drivers to register a driver name and a longname. This allows user space to determine the flavour of machine driver. Signed-off-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Best regards --- Kuninori Morimoto

5 19

[alsa-devel] ASoC updates for 3.1
by Mark Brown 09 Jul '11

09 Jul '11

The following changes since commit f187700c2d769bd3160b52f99cc4e747d5c5014b: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6 into topic/asoc (2011-07-05 08:20:19 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Axel Lin (1): ASoC: pxa2xx-pcm: remove unused variable 'dai' Kuninori Morimoto (1): ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 Liam Girdwood (4): ASoC: core - Add API call to register platform kcontrols. ASoC: core - Add platform widget IO ASoC: core - Add convenience register for platform kcontrol and DAPM ASoC: core - Add platform IO tracing Mark Brown (3): ASoC: Manage WM8731 ACTIVE bit as a supply widget ASoC: Manage WM8731 ACTIVE bit as a supply widget Merge branch 'for-3.0' into for-3.1 Stephen Warren (1): ASoC: Tegra: Implement SPDIF CPU DAI include/sound/soc-dapm.h | 2 + include/sound/soc.h | 12 + include/trace/events/asoc.h | 45 ++++ sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/wm8731.c | 29 +--- sound/soc/pxa/pxa2xx-pcm.c | 1 - sound/soc/soc-core.c | 47 ++++ sound/soc/soc-dapm.c | 13 +- sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 12 files changed, 976 insertions(+), 30 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h

1 0

[alsa-devel] Jack event API - decision needed
by David Henningsson 09 Jul '11

09 Jul '11

Hi, I'm still new in the community in that sense that I'm not sure how decisions are made. But I could use the outcome of such a decision. Background: I'm trying to pull together the missing pieces of jack detection on both kernel / plumbing / application layers, so that when you plug something in (headset, microphone etc), userspace is notified and can take appropriate actions (e g routing decisions). As part of that I wrote a udev patch a few days ago, which nobody commented on in alsa-devel [1], but was somewhat disliked by at least Kay Sievers who maintains udev [2], who preferred we would rewrite our input layer to do something else within ALSA. So before I proceed further I'd like to know if 1) We're continuing the path with /dev/input devices 2a) We'll rewrite these devices to be read-only ALSA mixer controls 2b) We'll rewrite these devices to be something within ALSA, but not exactly mixer controls. For options 2a) and 2b) I guess the existing /dev/input thing should be deprecated and/or removed. So part of decision should maybe be based on information about how widespread the usage of these devices are currently...? -- David Henningsson, Canonical Ltd. http://launchpad.net/~diwic [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2011-June/040916.html [2] e g http://www.spinics.net/lists/hotplug/msg04949.html

10 42

Re: [alsa-devel] Sound BoF in LinuxCon Europe?
by Mark Brown 09 Jul '11

09 Jul '11

On Fri, Jul 08, 2011 at 12:36:34PM +0200, Takashi Iwai wrote: > just wondering how about holding a sound-BoF in LinuxCon Europe in > October? > I know ELCE will be held at the very same time, so it may need the > adjustment of schedule. But in general what do you think? I have no current plans to attend these conferences, I'm planning to go to LPC instead. Not sure if I'd be able to arrange it unless I get a KS invite. > An alternative is a workshop in KS2011 day 1, but I think KS suits > better for more broader topics, e.g. the discussions on the media- > notification support or the jack implementation like a thread in the > last month. Yes, I don't think it's good for covering at KS as it's far too detailed. If we're going to do something probably LinuxCon/ELC Europe make most sense.

1 0

[alsa-devel] ASoC updates for 3.0
by Mark Brown 09 Jul '11

09 Jul '11

The following changes since commit 873bd4cb4fbba6a3e99f750e17ef2ba6ef96e9d3: ASoC: Don't set invalid name string to snd_card->driver field (2011-07-05 14:39:27 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 The WM8731 patch fixes resume when we suspend with an active playback or record. Kuninori Morimoto (1): ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2 Mark Brown (1): ASoC: Manage WM8731 ACTIVE bit as a supply widget sound/soc/codecs/ak4642.c | 2 +- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 2 files changed, 4 insertions(+), 27 deletions(-)

1 0

Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Takashi Iwai 09 Jul '11

09 Jul '11

At Thu, 7 Jul 2011 21:35:17 +0900, Toshiharu Okada wrote: > > Date: Thu, 07 Jul 2011 11:00:47 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > >> > >> Therefore, I am not considering about ASoC. > >> Do I need to fit this driver to an ASoC framework for an upstream? > > > >Yes, I guess this would be a better choice in your case. > > Thank you for your comment. > >From now on, We study about ASoC. > If there are a document, a website, a driver, etc. which are referred, > please tell us. Check Documentation/sound/alsa/soc/* files. > Why is choosing ASoS better? > Is it better to correspond to ASoC, when Audio Codec is unfixed? It's not clear what you mean exactly as "unfixed", but in general, the decision rather depends on the usage of the device. If it's designed for a use as a desktop PC component (such as a PCI card or onboard-audio on PC), it'd make sense to implement as a stand-alone, self-contained driver. OTOH, if it's targeted mainly for embedded area, ASoC is the right answer. It's more modularized, and can be more flexibly configured in the end. Takashi

2 1

Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 08 Jul '11

08 Jul '11

On Thu, Jul 7, 2011 at 3:08 PM, Oliver Neukum <oliver(a)neukum.org> wrote: > Am Donnerstag, 7. Juli 2011, 14:38:51 schrieb Daniel Mack: >> On Thu, Jul 7, 2011 at 2:33 PM, Oliver Neukum <oliver(a)neukum.org> wrote: >> >> > PS: Do you still see this if you enable 64bit DMA for EHCI? >> >> The problem is that I personally don't see that issue at all. I even >> installed 4GB of RAM to my development machine last year to be able to >> reproduce this, but I can't, even when the memory allocator is under >> heavy load. The only people who see this effect are Pedro Ribeiro and >> William Light (both in Cc:), and both have been very helpful in trying >> patches and reporting back in detail. Which instructions could we >> probably give to these people to finally hunt this issue down? > > Do they have an IOMMU or are they using bounce buffers? > If the latter, you might be able to reproduce it if you disable your > IOMMU. Ok, Pedro and William - can I ask you to dump the following information to some paste bin: - your .config - the dmesg output of a booted kernel that shows the effect - the output of "sudo lspci -v" Also, the attached patch might give you at least one warning to your kernel logs once the audio stream starts and outputs noise. I'm curious whether it really does. Thanks, Daniel

1 0

[alsa-devel] [RFC PATCH] ARM: imx27pdk (3ds): enable m13783 codec over SSI
by Jürgen Lambrecht 08 Jul '11

08 Jul '11

- add SSI in Kconfig - enable audio in MFD mc13783 - configure audmux-v1 - configure SSI4 Signed-off-by: Jürgen Lambrecht <J.Lambrecht(a)televic.com> --- arch/arm/mach-imx/Kconfig | 1 + arch/arm/mach-imx/mach-mx27_3ds.c | 40 ++++++++++++++++++++++++++++++++++++- 2 files changed, 40 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-imx/Kconfig b/arch/arm/mach-imx/Kconfig index 0a3512b..2fe5e70 100644 --- a/arch/arm/mach-imx/Kconfig +++ b/arch/arm/mach-imx/Kconfig @@ -286,6 +286,7 @@ config MACH_MX27_3DS select IMX_HAVE_PLATFORM_MXC_MMC select IMX_HAVE_PLATFORM_MXC_NAND select IMX_HAVE_PLATFORM_SPI_IMX + select IMX_HAVE_PLATFORM_IMX_SSI select MXC_DEBUG_BOARD select MXC_ULPI if USB_ULPI help diff --git a/arch/arm/mach-imx/mach-mx27_3ds.c b/arch/arm/mach-imx/mach-mx27_3ds.c index 2787cf3..c0f7dcc 100644 --- a/arch/arm/mach-imx/mach-mx27_3ds.c +++ b/arch/arm/mach-imx/mach-mx27_3ds.c @@ -41,6 +41,7 @@ #include <mach/irqs.h> #include <mach/3ds_debugboard.h> #include <mach/mxc_nand.h> +#include <mach/audmux.h> #include "devices-imx27.h" @@ -101,6 +102,11 @@ static const int mx27pdk_pins[] __initconst = { PE2_PF_USBOTG_DIR, PE24_PF_USBOTG_CLK, PE25_PF_USBOTG_DATA7, + /* SSI4 for audio to MC13783 */ + PC16_PF_SSI4_FS, + PC17_PF_SSI4_RXD, + PC18_PF_SSI4_TXD, + PC19_PF_SSI4_CLK, /* CSPI1 */ PD31_PF_CSPI1_MOSI, PD30_PF_CSPI1_MISO, @@ -271,6 +277,13 @@ static struct regulator_init_data vgen_init = { .consumer_supplies = vgen_consumers, }; +static struct regulator_init_data vaudio_init = { + .constraints = { + .boot_on = 1, + .always_on = 1, + }, +}; + static struct mc13xxx_regulator_init_data mx27_3ds_regulators[] = { { .id = MC13783_REG_VMMC1, @@ -284,6 +297,9 @@ static struct mc13xxx_regulator_init_data mx27_3ds_regulators[] = { }, { .id = MC13783_REG_GPO3, /* Turn on 3.3V */ .init_data = &gpo_init, + }, { + .id = MC13783_REG_VAUDIO, /* Power codec */ + .init_data = &vaudio_init, }, }; @@ -295,7 +311,7 @@ static struct mc13xxx_platform_data mc13783_pdata = { }, .flags = MC13783_USE_REGULATOR | MC13783_USE_TOUCHSCREEN | - MC13783_USE_RTC, + MC13783_USE_RTC | MC13783_USE_CODEC, }; /* SPI */ @@ -385,6 +401,11 @@ static const struct imxi2c_platform_data mx27_3ds_i2c0_data __initconst = { .bitrate = 100000, }; +/* 4-wire is synchronous; SSI-slave and MC13783-master; DMA instead of FIQ */ +static const struct imx_ssi_platform_data mx27pdk_ssi4_pdata __initconst = { + .flags = IMX_SSI_SYN | IMX_SSI_DMA | IMX_SSI_USE_I2S_SLAVE, +}; + static void __init mx27pdk_init(void) { imx27_soc_init(); @@ -420,6 +441,23 @@ static void __init mx27pdk_init(void) pr_warn("Init of the debugboard failed, all devices on the debugboard are unusable.\n"); imx27_add_imx_i2c(0, &mx27_3ds_i2c0_data); imx27_add_imx_fb(&mx27_3ds_fb_data); + + /* SSI unit master I2S codec connected to SSI_PINS_4*/ + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR1_SSI0, + MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V1_PCR_TFSDIR | /* Transmit Frame Sync is output */ + MXC_AUDMUX_V1_PCR_TCLKDIR | /* Transmit Clock is output */ + MXC_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | /*Transmit Frame Sync and Clock select source port to be HPCR3 or external SSI4*/ + MXC_AUDMUX_V1_PCR_RFSDIR | /* Receive Frame Sync is output */ + MXC_AUDMUX_V1_PCR_RCLKDIR | /* Receive Clock is output */ + MXC_AUDMUX_V1_PCR_RFCSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) | /*Receive Frame Sync and Clock select source port to be HPCR3 or external SSI4*/ + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR3_SSI_PINS_4) /*Receive Data select source port to be HPCR3 or external SSI4*/ + ); + mxc_audmux_v1_configure_port(MX27_AUDMUX_HPCR3_SSI_PINS_4, + MXC_AUDMUX_V1_PCR_SYN | /* 4wire mode */ + MXC_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0) /*Receive Data select source port to be HPCR1 or internal SSI1*/ + ); + imx27_add_imx_ssi(0, &mx27pdk_ssi4_pdata); } static void __init mx27pdk_timer_init(void) -- 1.7.1

1 0

Re: [alsa-devel] [PATCH] ASoC: dapm - Add API calls to query valid DAPM paths.
by Mark Brown 07 Jul '11

07 Jul '11

Funny formatting due to phone, sorry... Indeed, but I'd have expected that anything that needed to do that would be able to (and want to) ask by name rather than type? Liam Girdwood <lrg(a)ti.com> wrote: >On 07/07/11 17:38, Mark Brown wrote: >> On Thu, Jul 07, 2011 at 05:18:47PM +0100, Liam Girdwood wrote: >> >>> The reason for using a separate walk here was because this code is >>> taken from my old auto-router and it was to avoid the extra logic >>> introduced by the auto-router. The auto-router did need to check for >>> loops and always select the shortest path when > 1 path was viable >>> (interesting as WM9713 has both loops and multiple paths). Let me see >>> how cleanly I can merge it into the current walk. >> >> Alternatively could we replace the current walk? It's O(n^2) or so in >> the number of widgets so it's not the greatest treasure ever. So long >> as we only have one walk and it works I'm happy. >> >>>> It does also occur to me that perhaps what we want to do here is allow >>>> widgets to find out about paths when they're being activated anyway? >> >>> For dynamic PCM, we need to also work out which DAIs are active based >>> on the graph. Hence for playback we can supply the DAC/AIF and find >>> out all the connected output AIF/Speakers/Pins/etc. This is required >>> at the start of open() in order that we can call the other PCM ops for >>> each DAI, codec and platform. >> >> Right, that's not really what I'm saying though - I'm saying that >> widgets in general could be interested in this information. For >> example, some charge pumps can save additional power with some input >> signal paths (which is currently handled but this would make that code >> more general). > >Ok, but I'm not stopping this sort of functionality from being added >later. Right now this is not a priority (for me anyway). > >> >>>>> + * snd_soc_dapm_get_connected_widgets_type - query audio path and it's widgets. >>>>> + * @dapm: the dapm context. >>>>> + * @stream_name: stream name. >>>>> + * @list: list of active widgets for this stream. >>>>> + * @stream: stream direction. >>>>> + * @type: Initial widget type. >>>>> + * >>>>> + * Queries DAPM graph as to whether an valid audio stream path exists for >>>>> + * the DAPM stream and initial widget type specified. This takes into account >>>>> + * current mixer and mux kcontrol settings. Creates list of valid widgets. >>>> >>>> Why would someone want to query by type? That seems surprising. Name >>>> and/or direction yes but type seems hard to find a use for. >> >>> We query by type so that we can qualify the root widget for a shared >>> common stream name. i.e. Dynamic DSP needs to find the AIF rather than >>> a DAC for stream X. >> >> It does? Hrm. Perhaps it would make life simpler if we just mandated >> the use of explicit AIF widgets? > >But we may also want to know what was after a DAC. I'm thinking back >from the conference discussions where we could use this information to >selectively enable/disable kcontrols. However, this is not going to be >used at the moment either so I'll stick by the AIF widgets. > >Liam > >

2 1

[alsa-devel] [PATCH] ASoC: dapm - Add API calls to query valid DAPM paths.
by Liam Girdwood 07 Jul '11

07 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Add DAPM API calls to determine whether a DAPM audio path is valid between source and sink widgets and return a lits of active widgets in that path. This also takes into account all kcontrol mux and mixer settings in between the source and sink widgets to validate the audio path. This will be used by the DSP core to determine the runtime DAI mappings between FE and BE DAIs in order to run PCM operations. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dapm.h | 14 ++ sound/soc/soc-dapm.c | 371 ++++++++++++++++++++++++++++++++++++++++++++- 2 files changed, 376 insertions(+), 9 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index e09505c..5a7dae6 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -310,6 +310,7 @@ struct snd_soc_dapm_path; struct snd_soc_dapm_pin; struct snd_soc_dapm_route; struct snd_soc_dapm_context; +struct snd_soc_dapm_widget_list; int dapm_reg_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); @@ -375,6 +376,13 @@ int snd_soc_dapm_force_enable_pin(struct snd_soc_dapm_context *dapm, int snd_soc_dapm_ignore_suspend(struct snd_soc_dapm_context *dapm, const char *pin); +/* dapm path query */ +int snd_soc_dapm_get_connected_widgets_type(struct snd_soc_dapm_context *dapm, + const char *stream_name, struct snd_soc_dapm_widget_list **list, + int stream, enum snd_soc_dapm_type type); +int snd_soc_dapm_get_connected_widgets_name(struct snd_soc_dapm_context *dapm, + const char *name, struct snd_soc_dapm_widget_list **list, int stream); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ @@ -432,6 +440,8 @@ struct snd_soc_dapm_path { u32 connect:1; /* source and sink widgets are connected */ u32 walked:1; /* path has been walked */ u32 weak:1; /* path ignored for power management */ + u32 length:6; /* path length - used by route checker */ + int (*connected)(struct snd_soc_dapm_widget *source, struct snd_soc_dapm_widget *sink); @@ -456,6 +466,8 @@ struct snd_soc_dapm_widget { unsigned char shift; /* bits to shift */ unsigned int saved_value; /* widget saved value */ unsigned int value; /* widget current value */ + unsigned int path_idx; + unsigned int hops; unsigned int mask; /* non-shifted mask */ unsigned int on_val; /* on state value */ unsigned int off_val; /* off state value */ @@ -518,6 +530,8 @@ struct snd_soc_dapm_context { enum snd_soc_bias_level target_bias_level; struct list_head list; + int num_valid_paths; /* valid paths for route checker */ + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0a78482..800af10 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -48,6 +48,8 @@ #include <trace/events/asoc.h> +#define DAPM_MAX_HOPS 16 + /* dapm power sequences - make this per codec in the future */ static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, @@ -169,6 +171,366 @@ static int soc_widget_update_bits(struct snd_soc_dapm_widget *w, return change; } +/* get snd_card from DAPM context */ +static inline struct snd_card *dapm_get_snd_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card->snd_card; + else if (dapm->platform) + return dapm->platform->card->snd_card; + + dev_err(dapm->dev, "no snd_card for dapm context\n"); + return NULL; +} + +/* get soc_card from DAPM context */ +static inline struct snd_soc_card *dapm_get_soc_card( + struct snd_soc_dapm_context *dapm) +{ + if (dapm->codec) + return dapm->codec->card; + else if (dapm->platform) + return dapm->platform->card; + + dev_err(dapm->dev, "no soc_card for dapm context\n"); + return NULL; +} + +/* reset 'walked' bit for each dapm path */ +static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_dapm_path *p; + + list_for_each_entry(p, &dapm->card->paths, list) + p->walked = 0; +} + +/* clear all 'walked' path data for each widget */ +static void dapm_clear_paths(struct snd_soc_dapm_context *dapm) +{ + struct snd_soc_dapm_path *p; + struct snd_soc_dapm_widget *w; + struct list_head *l; + + list_for_each(l, &dapm->card->paths) { + p = list_entry(l, struct snd_soc_dapm_path, list); + p->length = 0; + } + list_for_each(l, &dapm->card->widgets) { + w = list_entry(l, struct snd_soc_dapm_widget, list); + w->hops = 0; + } + dapm_clear_walk(dapm); +} + +/* add widget to list if it's not already in the list */ +static int dapm_add_unique_widget(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget_list **list, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_dapm_widget_list *wlist; + int wlistsize, wlistentries, i; + + /* is the list empty ? */ + if (*list == NULL) { + + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + sizeof(struct snd_soc_dapm_widget *); + *list = kzalloc(wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + } else { + wlist = *list; + + /* is this widget already in the list */ + for (i = 0; i < wlist->num_widgets; i++) { + if (wlist->widgets[i] == w) + return 0; + } + + wlistentries = wlist->num_widgets + 1; + wlistsize = sizeof(struct snd_soc_dapm_widget_list) + + wlistentries * sizeof(struct snd_soc_dapm_widget *); + *list = krealloc(wlist, wlistsize, GFP_KERNEL); + if (*list == NULL) { + dev_err(dapm->dev, "can't allocate widget list for %s\n", + w->name); + return -ENOMEM; + } + } + wlist = *list; + + /* insert the widget */ + dev_dbg(dapm->dev, "added %s in widget list pos %d\n", + w->name, wlist->num_widgets); + wlist->widgets[wlist->num_widgets] = w; + wlist->num_widgets++; + return 1; +} + +/* is widget an output endpoint */ +static int is_output_widget_ep(struct snd_soc_dapm_widget *widget) +{ + switch (widget->id) { + case snd_soc_dapm_adc: + case snd_soc_dapm_aif_out: + return 1; + case snd_soc_dapm_output: + if (widget->connected && !widget->ext) + return 1; + else + return 0; + case snd_soc_dapm_hp: + case snd_soc_dapm_spk: + case snd_soc_dapm_line: + return !list_empty(&widget->sources); + default: + return 0; + } +} + +/* is widget an input endpoint */ +static int is_input_widget_ep(struct snd_soc_dapm_widget *widget) +{ + switch (widget->id) { + case snd_soc_dapm_dac: + case snd_soc_dapm_aif_in: + return 1; + case snd_soc_dapm_input: + if (widget->connected && !widget->ext) + return 1; + else + return 0; + case snd_soc_dapm_mic: + return !list_empty(&widget->sources); + default: + return 0; + } +} + +/* + * Find all the playback paths between the root source widget and + * sink widgets. Store all path widgets in the list. + */ +static int dapm_find_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list, int hops) +{ + struct list_head *lp; + struct snd_soc_dapm_path *path; + int dist = 0; + + if (hops > DAPM_MAX_HOPS) + return 0; + + /* are we at an endpoint widget */ + if (is_output_widget_ep(root)) { + dev_dbg(dapm->dev," ! %d: valid playback route found\n", hops); + dapm->num_valid_paths++; + return 1; + } + + /* have we visited this widget before */ + if (root->hops && root->hops <= hops) + return 0; + root->hops = hops; + + /* check all the output paths on this source widget by walking + * from source to sink */ + list_for_each(lp, &root->sinks) { + path = list_entry(lp, struct snd_soc_dapm_path, list_source); + + dev_dbg(dapm->dev," %c %d: %s -> %s -> %s\n", + path->connect ? '*' : ' ', hops, + root->name, path->name, path->sink->name); + + /* have we been down this path before ? */ + if (path->length && path->length <= hops) + continue; + + /* check down the next path if connected */ + if (path->sink && path->connect && + dapm_find_playback_paths(dapm, path->sink, list, hops + 1)) { + path->length = hops; + + /* add widget to list */ + dapm_add_unique_widget(dapm, list, path->sink); + + if (!dist || dist > path->length) + dist = path->length; + } + } + + return dist; +} + +/* + * Find all the capture paths between the root sink widget and + * source widgets. Store all path widgets in the list. + */ +static int dapm_find_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list, int hops) +{ + struct list_head *lp; + struct snd_soc_dapm_path *path; + int dist = 0; + + if (hops > DAPM_MAX_HOPS) + return 0; + + /* are we at an endpoint widget */ + if (is_input_widget_ep(root)) { + dev_dbg(dapm->dev," ! %d: valid capture route found\n", hops); + dapm->num_valid_paths++; + return 1; + } + + /* have we visited this widget before */ + if (root->hops && root->hops <= hops) + return 0; + root->hops = hops; + + /* check all the output paths on this source widget by walking from + * sink to source */ + list_for_each(lp, &root->sources) { + path = list_entry(lp, struct snd_soc_dapm_path, list_sink); + + dev_dbg(dapm->dev," %c %d: %s <- %s <- %s\n", + path->connect ? '*' : ' ', hops, + root->name, path->name, path->source->name); + + /* have we been here before ? */ + if (path->length && path->length <= hops) + continue; + + /* check down the next path if connected */ + if (path->source && path->connect && + dapm_find_capture_paths(dapm, path->source, list, hops + 1)) { + path->length = hops; + + /* add widget to list */ + dapm_add_unique_widget(dapm, list, path->source); + + if (!dist || dist > path->length) + dist = path->length; + } + } + + return dist; +} + +static int dapm_get_playback_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + dev_dbg(dapm->dev, "Playback: checking paths from %s\n",root->name); + dapm_find_playback_paths(dapm, root, list, 0); + return dapm->num_valid_paths; +} + +static int dapm_get_capture_paths(struct snd_soc_dapm_context *dapm, + struct snd_soc_dapm_widget *root, + struct snd_soc_dapm_widget_list **list) +{ + dev_dbg(dapm->dev, "Capture: checking paths to %s\n", root->name); + dapm_find_capture_paths(dapm, root, list, 0); + return dapm->num_valid_paths; +} + +/** + * snd_soc_dapm_get_connected_widgets_type - query audio path and it's widgets. + * @dapm: the dapm context. + * @stream_name: stream name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * @type: Initial widget type. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the DAPM stream and initial widget type specified. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets_type( + struct snd_soc_dapm_context *dapm, const char *stream_name, + struct snd_soc_dapm_widget_list **list, int stream, + enum snd_soc_dapm_type type) +{ + struct snd_soc_dapm_widget *w; + int paths; + + /* get stream root widget AIF, DAC or ADC from stream string + * and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (!w->sname) + continue; + + if (w->id != type) + continue; + + if (strstr(w->sname, stream_name)) + goto found; + } + dev_err(dapm->dev, "root widget not found\n"); + return 0; + +found: + dapm->num_valid_paths = 0; + *list = NULL; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_paths(dapm); + return paths; +} + +/** + * snd_soc_dapm_get_connected_widgets_name - query audio path and it's widgets. + * @dapm: the dapm context. + * @name: initial widget name. + * @list: list of active widgets for this stream. + * @stream: stream direction. + * + * Queries DAPM graph as to whether an valid audio stream path exists for + * the initial widget specified by name. This takes into account + * current mixer and mux kcontrol settings. Creates list of valid widgets. + * + * Returns the number of valid paths or negative error. + */ +int snd_soc_dapm_get_connected_widgets_name(struct snd_soc_dapm_context *dapm, + const char *name, struct snd_soc_dapm_widget_list **list, int stream) +{ + struct snd_soc_dapm_widget *w; + int paths; + + /* get stream root widget AIF, DAC or ADC from stream string + * and direction */ + list_for_each_entry(w, &dapm->card->widgets, list) { + + if (strstr(w->name, name)) + goto found; + } + dev_err(dapm->dev, "root widget %s not found\n", name); + return 0; + +found: + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + paths = dapm_get_playback_paths(dapm, w, list); + else + paths = dapm_get_capture_paths(dapm, w, list); + + dapm_clear_paths(dapm); + return paths; +} + /** * snd_soc_dapm_set_bias_level - set the bias level for the system * @dapm: DAPM context @@ -576,15 +938,6 @@ static int dapm_new_pga(struct snd_soc_dapm_widget *w) return 0; } -/* reset 'walked' bit for each dapm path */ -static inline void dapm_clear_walk(struct snd_soc_dapm_context *dapm) -{ - struct snd_soc_dapm_path *p; - - list_for_each_entry(p, &dapm->card->paths, list) - p->walked = 0; -} - /* We implement power down on suspend by checking the power state of * the ALSA card - when we are suspending the ALSA state for the card * is set to D3. -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: ak4642: fixup snd_soc_update_bits mask for PW_MGMT2
by Kuninori Morimoto 07 Jul '11

07 Jul '11

mask didn't cover update-data Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- This patch based on asoc/for-3.1 sound/soc/codecs/ak4642.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 4be0570..65f4604 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -357,7 +357,7 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) default: return -EINVAL; } - snd_soc_update_bits(codec, PW_MGMT2, MS, data); + snd_soc_update_bits(codec, PW_MGMT2, MS | MCKO | PMPLL, data); snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko); /* format type */ -- 1.7.4.1

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Re: [alsa-devel] [PATCH] ASoC: pxa2xx-pcm: remove unused variable 'dai'
by Mark Brown 07 Jul '11

07 Jul '11

On Wed, Jul 06, 2011 at 09:20:42PM +0800, Axel Lin wrote: > Remove unused variable 'dai' to eliminate below warning. Applied, thanks.

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[alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 07 Jul '11

07 Jul '11

I have to revisit an issue we've discussed in length around a year ago and which still remains unsolved. I've been getting feedback from more users of my driver snd-usb-caiaq who report the issue found in bug #15580 in the kernel bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=15580 Let me quickly summarize the current state of this issue as I see it. The problem seems to be that certain 64bit chipsets can't deal with the fact that an URB's transfer_buffer is allocated with kmalloc(GFP_KERNEL). The effect is that kmalloc() is very likely to hand out memory which is not addressable by devices that are connected via 32bit PCI busses, such as EHCI controllers. In theory, DMA bounce buffers should be installed in such cases, or the IOMMU would be in charge to re-map these buffers to suitable locations, but for at least two people who have reported the issue, this obviously fails. So my approach was to force the driver using memory that is DMA coherent, which leads to 32bit addressable memory if the device's DMA_MASK is set accordingly. Other drivers do the same, and the patch for doing this is found here: http://lkml.org/lkml/2010/5/7/61. Users that see the bug all report that this patch fixes their problem with my driver. However, it was rejected back then as other developers said it was fixing the wrong end, and that the driver doesn't necessarily need coherent memory. Takashi recently posted a patch to the bugzilla entry which uses a different approach: it introduces a function to determine suitable GFP flags for USB devices, and passes __DMA32 to kmalloc() eventually. However, using this flags directly with the SLUB allocator is illegal and causes a BUG() in mm/slub.c, cache_grow(). The question now is how to proceed. I see three possible ways. 1. Find a way to allocate 32bit-aware memory with kmalloc(), following Takashi's idea 2. Find out exactly why these machines fail to install bounce buffers or set up their IOMMU correctly 3. re-activate the currently disabled functions usb_buffer_{map,unmap,sync} functions and let the USB stack do the memory mapping I'm not really deep into all the memory management code and even less into how the IOMMU and bounce buffers work, so I need some help here. Can anyone elaborate what would be the best option? I really want to get this fixed now, and provide a driver that works for everyone. What really puzzles me is that this doesn't hit a lot more people with other drivers. Thanks, Daniel

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Re: [alsa-devel] Allocating buffers for USB transfers (again)
by Daniel Mack 07 Jul '11

07 Jul '11

On Thu, Jul 7, 2011 at 2:33 PM, Oliver Neukum <oliver(a)neukum.org> wrote: > PS: Do you still see this if you enable 64bit DMA for EHCI? The problem is that I personally don't see that issue at all. I even installed 4GB of RAM to my development machine last year to be able to reproduce this, but I can't, even when the memory allocator is under heavy load. The only people who see this effect are Pedro Ribeiro and William Light (both in Cc:), and both have been very helpful in trying patches and reporting back in detail. Which instructions could we probably give to these people to finally hunt this issue down? Daniel

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Re: [alsa-devel] [PATCH] ASoC: pxa2xx-pcm: remove unused variable 'dai'
by Liam Girdwood 07 Jul '11

07 Jul '11

On 06/07/11 14:20, Axel Lin wrote: > Remove unused variable 'dai' to eliminate below warning. > > CC sound/soc/pxa/pxa2xx-pcm.o > sound/soc/pxa/pxa2xx-pcm.c: In function 'pxa2xx_soc_pcm_new': > sound/soc/pxa/pxa2xx-pcm.c:91: warning: unused variable 'dai' > > Signed-off-by: Axel Lin <axel.lin(a)gmail.com> > --- > sound/soc/pxa/pxa2xx-pcm.c | 1 - > 1 files changed, 0 insertions(+), 1 deletions(-) > > diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c > index da28394..c430600 100644 > --- a/sound/soc/pxa/pxa2xx-pcm.c > +++ b/sound/soc/pxa/pxa2xx-pcm.c > @@ -88,7 +88,6 @@ static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); > static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) > { > struct snd_card *card = rtd->card->snd_card; > - struct snd_soc_dai *dai = rtd->cpu_dai; > struct snd_pcm *pcm = rtd->pcm; > int ret = 0; > Acked-by: Liam Girdwood <lrg(a)ti.com>

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Re: [alsa-devel] [PATCH] Add SoundCard driver for OKI SEMICONDUCTOR ML7213 IOH
by Takashi Iwai 07 Jul '11

07 Jul '11

At Thu, 7 Jul 2011 16:54:18 +0900, Toshiharu Okada wrote: > > Date: Wed, 06 Jul 2011 13:06:15 +0200 > From: Takashi Iwai <tiwai(a)suse.de> > > > >I just took a quick glance over the code, and wonder whether this kind > >of driver would fit better with ASoC framework. > >Have you considered the implementation on ASoC? > > Thank you for your comment. > > I referred to the following files and implemented this driver. > > http://www.mirrorservice.org/sites/ftp.sourceforge.net/pub/sourceforge/s/pr… > sound/drivers/sta2x11i2s.c > > Therefore, I am not considering about ASoC. > Do I need to fit this driver to an ASoC framework for an upstream? Yes, I guess this would be a better choice in your case. Takashi

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[alsa-devel] hda-intel front headphone (8086:27d8 1043:83bc)
by Alex 07 Jul '11

07 Jul '11

Hi, I run a Debian/sid on an Asus P5KPL-SE mainboard, the snd-hda-intel seem to work fine, ever if it run with guess method: Unknown hardware: "HDA-Intel" "Realtek ALC887" "HDA:10ec0887,104383bc,00100202" "0x1043" "0x83bc" Hardware is initialized using a guess method The only annoying thing is about the front headphone, every mixer application show the front headphone item but doesn't have adjustable level and I hear nothing on headphone. Any hint? http://www.alsa-project.org/db/?f=d6bc3005fab4683f2606e48f2bc909258a4ee3e3 Thanks in advance Regards

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[alsa-devel] [PATCH] ASoC: twl6040: Add back support for legacy mode
by Peter Ujfalusi 07 Jul '11

07 Jul '11

The legacy mode has been accidentaly removed by commit: ASoC: twl6040: add all ABE DAIs Add back the twl6040-hifi dai. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi Tony, Mark, Liam, this patch fixes the accidental removal of the twl6040-hifi dai from the codec driver. The series, which this patch depends on not yet pulled (I have the branch ready), since I'm still waiting for few acks for it. The MFD series is mostly OK, but the threaded interrupt related followup waits for acks, and also the PLL restructuring series is in waiting state. Is it OK, if I queue up this patch as well for the same pull request, and goes with the rest through l-o (since it is on top of those changes)? I could try to squash it down, if you prefer to the offending commit - I hope it is not breaking the rebase too much... Thank you, Peter sound/soc/codecs/twl6040.c | 18 ++++++++++++++++++ 1 files changed, 18 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index d4b944c..443032b 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1464,6 +1464,24 @@ static struct snd_soc_dai_ops twl6040_dai_ops = { static struct snd_soc_dai_driver twl6040_dai[] = { { + .name = "twl6040-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = TWL6040_RATES, + .formats = TWL6040_FORMATS, + }, + .ops = &twl6040_dai_ops, +}, +{ .name = "twl6040-ul", .capture = { .stream_name = "Capture", -- 1.7.6

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[alsa-devel] [PATCH v3 0/5] MFD/input/ASoC: twl6040: irq registration changes
by Peter Ujfalusi 06 Jul '11

06 Jul '11

Hello, Changes since v2: - Fix the check in the slaves driver for the validity of the received irq - Mark Brown's ack added to patch 3 Intro mail froim v1: As Samuel, and Felipe requested for the twl6040 MFD series: http://marc.info/?l=linux-kernel&m=130978512225430&w=2 http://marc.info/?l=linux-omap&m=130978403224218&w=2 This is the update to remove the obfuscation around the threaded irq request in the twl6040 driver and it's slaves. With this change the slaves uses directly the threaded irq request, and the core driver passes the irq number for the slaves to use (configured runtime, so machine drivers do not need to be updated). Regards, Peter PS: for the remaining comments from Samuel I have changed the series, since the requested change was small. I'm not intended to send the series again, unless it is requested (again the change from v6 is minimal - few lines has been removed/replaced). --- Peter Ujfalusi (5): MFD: twl6040: Use resource to provide irq number for slaves input: twl6040-vibra: Do not use wrapper for irq request ASoC: twl6040: Do not use wrapper for irq request MFD: twl6040: Demand valid interrupt configuration MFD: twl6040: Remove wrapper for threaded irq request drivers/input/misc/twl6040-vibra.c | 17 ++++++--- drivers/mfd/twl6040-core.c | 69 ++++++++++++++++++++++++----------- drivers/mfd/twl6040-irq.c | 16 +-------- include/linux/mfd/twl6040.h | 22 ----------- sound/soc/codecs/twl6040.c | 19 +++++++--- 5 files changed, 74 insertions(+), 69 deletions(-) -- 1.7.6

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Re: [alsa-devel] [RFC] ASoC: Add machine driver for i.MX SGTL500 based boards
by Mark Brown 06 Jul '11

06 Jul '11

On Mon, Jul 04, 2011 at 03:41:55PM -0300, Fabio Estevam wrote: > sgtl5000 1-000a: sgtl5000 revision 17 > sgtl5000 1-000a: asoc: failed to probe CODEC sgtl5000.1-000a: -22 > asoc: failed to instantiate card sgtl5000: -22 Have you looked to see where the error is coming from? > + ret = snd_soc_dai_set_sysclk(codec_dai, SGTL5000_LRCLK, > + params_rate(params), 0); > + > + if (ret) { > + pr_err("%s: failed setting codec sysclk\n", __func__); > + return ret; > + } Fix this in the CODEC driver, copying this code into the machine drivers is clearly not useful. > +static int __init imx_sgtl5000_init(void) > +{ > + int ret; > + > + imx_sgtl5000_snd_device = platform_device_alloc("soc-audio", -1); > + if (!imx_sgtl5000_snd_device) > + return -ENOMEM; We're not accepting new drivers using soc-audio, you should use a regular platform device and register the card with snd_soc_register_card().

2 4

[alsa-devel] [PATCH] ARM i.MX dma: Fix burstsize settings
by Sascha Hauer 06 Jul '11

06 Jul '11

dmaengine expects the maxburst parameter in words, not bytes. The imxdma driver and its users do this wrong. Fix this. As a side note the imx-pcm-dma-mx2 driver was 'fixed' to work with imx-dma. This broke the driver with imx-sdma support which correctly takes the maxburst parameter in words. This patch puts the sdma based sound back to work. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> --- I made this as a cross subsystem patch to maintain bisectability. I suggest to push it either via my tree or the dmaengine tree. drivers/dma/imx-dma.c | 3 ++- drivers/mmc/host/mxcmmc.c | 8 ++++---- sound/soc/imx/imx-pcm-dma-mx2.c | 4 ++-- 3 files changed, 8 insertions(+), 7 deletions(-) diff --git a/drivers/dma/imx-dma.c b/drivers/dma/imx-dma.c index e18eaab..d99f71c 100644 --- a/drivers/dma/imx-dma.c +++ b/drivers/dma/imx-dma.c @@ -135,7 +135,8 @@ static int imxdma_control(struct dma_chan *chan, enum dma_ctrl_cmd cmd, if (ret) return ret; - imx_dma_config_burstlen(imxdmac->imxdma_channel, imxdmac->watermark_level); + imx_dma_config_burstlen(imxdmac->imxdma_channel, + imxdmac->watermark_level * imxdmac->word_size); return 0; default: diff --git a/drivers/mmc/host/mxcmmc.c b/drivers/mmc/host/mxcmmc.c index cc20e02..14aa213 100644 --- a/drivers/mmc/host/mxcmmc.c +++ b/drivers/mmc/host/mxcmmc.c @@ -715,13 +715,13 @@ static void mxcmci_set_ios(struct mmc_host *mmc, struct mmc_ios *ios) int burstlen, ret; /* - * use burstlen of 64 in 4 bit mode (--> reg value 0) - * use burstlen of 16 in 1 bit mode (--> reg value 16) + * use burstlen of 64 (16 words) in 4 bit mode (--> reg value 0) + * use burstlen of 16 (4 words) in 1 bit mode (--> reg value 16) */ if (ios->bus_width == MMC_BUS_WIDTH_4) - burstlen = 64; - else burstlen = 16; + else + burstlen = 4; if (mxcmci_use_dma(host) && burstlen != host->burstlen) { host->burstlen = burstlen; diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 4173b3d..43fdc24f 100644 --- a/sound/soc/imx/imx-pcm-dma-mx2.c +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -110,12 +110,12 @@ static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream, slave_config.direction = DMA_TO_DEVICE; slave_config.dst_addr = dma_params->dma_addr; slave_config.dst_addr_width = buswidth; - slave_config.dst_maxburst = dma_params->burstsize * buswidth; + slave_config.dst_maxburst = dma_params->burstsize; } else { slave_config.direction = DMA_FROM_DEVICE; slave_config.src_addr = dma_params->dma_addr; slave_config.src_addr_width = buswidth; - slave_config.src_maxburst = dma_params->burstsize * buswidth; + slave_config.src_maxburst = dma_params->burstsize; } ret = dmaengine_slave_config(iprtd->dma_chan, &slave_config); -- 1.7.5.3 -- Pengutronix e.K. | | Industrial Linux Solutions | http://www.pengutronix.de/ | Peiner Str. 6-8, 31137 Hildesheim, Germany | Phone: +49-5121-206917-0 | Amtsgericht Hildesheim, HRA 2686 | Fax: +49-5121-206917-5555 |

2 1

[alsa-devel] [PATCH] ASoC: Conditionalize the enable of WM8994 ADC TDM mode
by Mark Brown 06 Jul '11

06 Jul '11

Future devices will not benefit from this. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 16 ++++++++++++---- 1 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c0d8686..299f6eb 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3114,10 +3114,18 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT, 1 << WM8994_AIF2DAC_3D_GAIN_SHIFT); - /* Unconditionally enable AIF1 ADC TDM mode; it only affects - * behaviour on idle TDM clock cycles. */ - snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, - WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + /* Unconditionally enable AIF1 ADC TDM mode on chips which can + * use this; it only affects behaviour on idle TDM clock + * cycles. */ + switch (control->type) { + case WM8994: + case WM8958: + snd_soc_update_bits(codec, WM8994_AIF1_CONTROL_1, + WM8994_AIF1ADC_TDM, WM8994_AIF1ADC_TDM); + break; + default: + break; + } wm8994_update_class_w(codec); -- 1.7.5.4

2 1

Re: [alsa-devel] Wrong bitrate from usb microphone
by Takashi Iwai 06 Jul '11

06 Jul '11

At Tue, 05 Jul 2011 15:34:06 +0200, Pali Rohár wrote: > > Hello, > > I have USB webcam with integrated USB microphone. Microphone does not working > correctly because kernel module snd-usb-audio set wrong bitrate. > > Problem appered after this commit: > http://git.kernel.org/?p=linux/kernel/git/stable/linux-2.6.39.y.git;a=commi… > > When I reverted patch in above commit microphone works properly. > This commit set bitrate to 8000Hz for 041e:4068 but my webcam (propably) > produce 16000Hz. So what to do with my webcam? It needs to revert above patch! > > Here is output from lsusb of my webcam: > Bus 002 Device 003: ID 041e:4068 Creative Technology, Ltd Live! Cam Notebook > [VF0470] > Device Descriptor: > bLength 18 > bDescriptorType 1 > bcdUSB 1.10 > bDeviceClass 0 (Defined at Interface level) > bDeviceSubClass 0 > bDeviceProtocol 0 > bMaxPacketSize0 8 > idVendor 0x041e Creative Technology, Ltd > idProduct 0x4068 Live! Cam Notebook [VF0470] > bcdDevice 1.00 > iManufacturer 1 > iProduct 2 > iSerial 0 > bNumConfigurations 1 Could you give the lsusb -v output? Arseniy, also check whether the current code is really OK? thanks, Takashi

1 0

[alsa-devel] [PATCH] Cirrus Logic CS4210-rev-A1 support.
by Tim Howe 06 Jul '11

06 Jul '11

From: Tim Howe <timothyc.howe(a)gmail.com> Signed-off-by: Tim Howe <tim.howe(a)cirrus.com> --- sound/pci/hda/patch_cirrus.c | 841 ++++++++++++++++++++++++++++++++++++++++-- 1 files changed, 805 insertions(+), 36 deletions(-) diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index c7b5ca2..7e09722 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -51,7 +51,7 @@ struct cs_spec { unsigned int cur_adc_format; hda_nid_t dig_in; - const struct hda_bind_ctls *capture_bind[2]; + struct hda_bind_ctls *capture_bind[2]; unsigned int gpio_mask; unsigned int gpio_dir; @@ -61,6 +61,12 @@ struct cs_spec { unsigned int hp_detect:1; unsigned int mic_detect:1; + /* CS421x */ + unsigned int spdif_detect:1; + unsigned int sense_b:1; + hda_nid_t vendor_nid; + struct hda_input_mux input_mux; + unsigned int last_input; }; /* available models */ @@ -114,18 +120,20 @@ enum { static inline int cs_vendor_coef_get(struct hda_codec *codec, unsigned int idx) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - return snd_hda_codec_read(codec, CS420X_VENDOR_NID, 0, + return snd_hda_codec_read(codec, spec->vendor_nid, 0, AC_VERB_GET_PROC_COEF, 0); } static inline void cs_vendor_coef_set(struct hda_codec *codec, unsigned int idx, unsigned int coef) { - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + struct cs_spec *spec = codec->spec; + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_COEF_INDEX, idx); - snd_hda_codec_write(codec, CS420X_VENDOR_NID, 0, + snd_hda_codec_write(codec, spec->vendor_nid, 0, AC_VERB_SET_PROC_COEF, coef); } @@ -231,7 +239,7 @@ static int cs_capture_pcm_cleanup(struct hda_pcm_stream *hinfo, /* */ -static const struct hda_pcm_stream cs_pcm_analog_playback = { +static struct hda_pcm_stream cs_pcm_analog_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -242,7 +250,7 @@ static const struct hda_pcm_stream cs_pcm_analog_playback = { }, }; -static const struct hda_pcm_stream cs_pcm_analog_capture = { +static struct hda_pcm_stream cs_pcm_analog_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -252,7 +260,7 @@ static const struct hda_pcm_stream cs_pcm_analog_capture = { }, }; -static const struct hda_pcm_stream cs_pcm_digital_playback = { +static struct hda_pcm_stream cs_pcm_digital_playback = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -264,7 +272,7 @@ static const struct hda_pcm_stream cs_pcm_digital_playback = { }, }; -static const struct hda_pcm_stream cs_pcm_digital_capture = { +static struct hda_pcm_stream cs_pcm_digital_capture = { .substreams = 1, .channels_min = 2, .channels_max = 2, @@ -331,8 +339,8 @@ static int is_ext_mic(struct hda_codec *codec, unsigned int idx) struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; hda_nid_t pin = cfg->inputs[idx].pin; - unsigned int val; - if (!is_jack_detectable(codec, pin)) + unsigned int val = snd_hda_query_pin_caps(codec, pin); + if (!(val & AC_PINCAP_PRES_DETECT)) return 0; val = snd_hda_codec_get_pincfg(codec, pin); return (snd_hda_get_input_pin_attr(val) != INPUT_PIN_ATTR_INT); @@ -346,10 +354,11 @@ static hda_nid_t get_adc(struct hda_codec *codec, hda_nid_t pin, nid = codec->start_nid; for (i = 0; i < codec->num_nodes; i++, nid++) { - hda_nid_t pins[2]; + hda_nid_t pins[HDA_MAX_NUM_INPUTS]; unsigned int type; int j, nums; - type = get_wcaps_type(get_wcaps(codec, nid)); + type = (get_wcaps(codec, nid) & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; if (type != AC_WID_AUD_IN) continue; nums = snd_hda_get_connections(codec, nid, pins, @@ -558,10 +567,10 @@ static int add_output(struct hda_codec *codec, hda_nid_t dac, int idx, const char *name; int err, index; struct snd_kcontrol *kctl; - static const char * const speakers[] = { + static char *speakers[] = { "Front Speaker", "Surround Speaker", "Bass Speaker" }; - static const char * const line_outs[] = { + static char *line_outs[] = { "Front Line-Out", "Surround Line-Out", "Bass Line-Out" }; @@ -641,7 +650,7 @@ static int build_output(struct hda_codec *codec) /* */ -static const struct snd_kcontrol_new cs_capture_ctls[] = { +static struct snd_kcontrol_new cs_capture_ctls[] = { HDA_BIND_SW("Capture Switch", 0), HDA_BIND_VOL("Capture Volume", 0), }; @@ -709,7 +718,7 @@ static int cs_capture_source_put(struct snd_kcontrol *kcontrol, return change_cur_input(codec, idx, 0); } -static const struct snd_kcontrol_new cs_capture_source = { +static struct snd_kcontrol_new cs_capture_source = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, @@ -718,7 +727,7 @@ static const struct snd_kcontrol_new cs_capture_source = { .put = cs_capture_source_put, }; -static const struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, +static struct hda_bind_ctls *make_bind_capture(struct hda_codec *codec, struct hda_ctl_ops *ops) { struct cs_spec *spec = codec->spec; @@ -821,8 +830,7 @@ static int build_digital_output(struct hda_codec *codec) if (!spec->multiout.dig_out_nid) return 0; - err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid, - spec->multiout.dig_out_nid); + err = snd_hda_create_spdif_out_ctls(codec, spec->multiout.dig_out_nid); if (err < 0) return err; err = snd_hda_create_spdif_share_sw(codec, &spec->multiout); @@ -847,14 +855,15 @@ static void cs_automute(struct hda_codec *codec) { struct cs_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; - unsigned int hp_present; + unsigned int caps, hp_present; hda_nid_t nid; int i; hp_present = 0; for (i = 0; i < cfg->hp_outs; i++) { nid = cfg->hp_pins[i]; - if (!is_jack_detectable(codec, nid)) + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) continue; hp_present = snd_hda_jack_detect(codec, nid); if (hp_present) @@ -923,7 +932,7 @@ static void init_output(struct hda_codec *codec) AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); if (!cfg->speaker_outs) continue; - if (is_jack_detectable(codec, nid)) { + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_UNSOLICITED_ENABLE, AC_USRSP_EN | HP_EVENT); @@ -982,7 +991,7 @@ static void init_input(struct hda_codec *codec) cs_vendor_coef_set(codec, IDX_ADC_CFG, coef); } -static const struct hda_verb cs_coef_init_verbs[] = { +static struct hda_verb cs_coef_init_verbs[] = { {0x11, AC_VERB_SET_PROC_STATE, 1}, {0x11, AC_VERB_SET_COEF_INDEX, IDX_DAC_CFG}, {0x11, AC_VERB_SET_PROC_COEF, @@ -1016,7 +1025,7 @@ static const struct hda_verb cs_coef_init_verbs[] = { * blocks, which will alleviate the issue. */ -static const struct hda_verb cs_errata_init_verbs[] = { +static struct hda_verb cs_errata_init_verbs[] = { {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ {0x11, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ @@ -1125,7 +1134,7 @@ static void cs_unsol_event(struct hda_codec *codec, unsigned int res) } } -static const struct hda_codec_ops cs_patch_ops = { +static struct hda_codec_ops cs_patch_ops = { .build_controls = cs_build_controls, .build_pcms = cs_build_pcms, .init = cs_init, @@ -1165,7 +1174,7 @@ static const char * const cs420x_models[CS420X_MODELS] = { }; -static const struct snd_pci_quirk cs420x_cfg_tbl[] = { +static struct snd_pci_quirk cs420x_cfg_tbl[] = { SND_PCI_QUIRK(0x10de, 0x0ac0, "MacBookPro 5,3", CS420X_MBP53), SND_PCI_QUIRK(0x10de, 0x0d94, "MacBookAir 3,1(2)", CS420X_MBP55), SND_PCI_QUIRK(0x10de, 0xcb79, "MacBookPro 5,5", CS420X_MBP55), @@ -1179,7 +1188,7 @@ struct cs_pincfg { u32 val; }; -static const struct cs_pincfg mbp53_pincfgs[] = { +static struct cs_pincfg mbp53_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100141 }, { 0x0b, 0x90100140 }, @@ -1193,7 +1202,7 @@ static const struct cs_pincfg mbp53_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg mbp55_pincfgs[] = { +static struct cs_pincfg mbp55_pincfgs[] = { { 0x09, 0x012b4030 }, { 0x0a, 0x90100121 }, { 0x0b, 0x90100120 }, @@ -1207,7 +1216,7 @@ static const struct cs_pincfg mbp55_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg imac27_pincfgs[] = { +static struct cs_pincfg imac27_pincfgs[] = { { 0x09, 0x012b4050 }, { 0x0a, 0x90100140 }, { 0x0b, 0x90100142 }, @@ -1221,22 +1230,22 @@ static const struct cs_pincfg imac27_pincfgs[] = { {} /* terminator */ }; -static const struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { +static struct cs_pincfg *cs_pincfgs[CS420X_MODELS] = { [CS420X_MBP53] = mbp53_pincfgs, [CS420X_MBP55] = mbp55_pincfgs, [CS420X_IMAC27] = imac27_pincfgs, }; -static void fix_pincfg(struct hda_codec *codec, int model) +static void fix_pincfg(struct hda_codec *codec, int model, + struct cs_pincfg **pin_configs) { - const struct cs_pincfg *cfg = cs_pincfgs[model]; + const struct cs_pincfg *cfg = pin_configs[model]; if (!cfg) return; for (; cfg->nid; cfg++) snd_hda_codec_set_pincfg(codec, cfg->nid, cfg->val); } - static int patch_cs420x(struct hda_codec *codec) { struct cs_spec *spec; @@ -1247,11 +1256,13 @@ static int patch_cs420x(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + spec->vendor_nid = CS420X_VENDOR_NID; + spec->board_config = snd_hda_check_board_config(codec, CS420X_MODELS, cs420x_models, cs420x_cfg_tbl); if (spec->board_config >= 0) - fix_pincfg(codec, spec->board_config); + fix_pincfg(codec, spec->board_config, cs_pincfgs); switch (spec->board_config) { case CS420X_IMAC27: @@ -1278,18 +1289,776 @@ static int patch_cs420x(struct hda_codec *codec) return err; } +/* + Cirrus Logic CS4210 + + 1 DAC => HP(sense) / Speakers, + 1 ADC <= LineIn(sense) / MicIn / DMicIn, + 1 SPDIF OUT => SPDIF Trasmitter(sense) +*/ +#define CS4210_DAC_NID 0x02 +#define CS4210_ADC_NID 0x03 +#define CS421X_VENDOR_NID 0x0B +#define CS421X_DMIC_PIN_NID 0x09 /* Port E */ +#define CS421X_SPDIF_PIN_NID 0x0A /* Port H */ + +#define CS421X_IDX_DEV_CFG 0x01 +#define CS421X_IDX_ADC_CFG 0x02 +#define CS421X_IDX_DAC_CFG 0x03 +#define CS421X_IDX_SPK_CTL 0x04 + +#define SPDIF_EVENT 0x04 + +/* CS4210 boards */ +enum { + CS421X_CDB4210, + CS421X_MODELS +}; + +/* CS4210 board names */ +static const char *cs421x_models[CS421X_MODELS] = { + [CS421X_CDB4210] = "cdb4210", +}; + +static struct snd_pci_quirk cs421x_cfg_tbl[] = { + /* Test Intel board + CDB2410 */ + SND_PCI_QUIRK(0x8086, 0x5001, "DP45SG/CDB4210", CS421X_CDB4210), + {} /* terminator */ +}; + +/* CS4210 board pinconfigs */ +/* Default CS4210 (CDB4210)*/ +static struct cs_pincfg cdb4210_pincfgs[] = { + { 0x05, 0x0321401f }, + { 0x06, 0x90170010 }, + { 0x07, 0x03813031 }, + { 0x08, 0xb7a70037 }, + { 0x09, 0xb7a6003e }, + { 0x0a, 0x034510f0 }, + {} /* terminator */ +}; + +static struct cs_pincfg *cs421x_pincfgs[CS421X_MODELS] = { + [CS421X_CDB4210] = cdb4210_pincfgs, +}; + +static struct hda_verb cs421x_coef_init_verbs[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 1}, + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DEV_CFG}, + /* + Disable Coefficient Index Auto-Increment(DAI)=1, + PDREF=0 + */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0001 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_ADC_CFG}, + /* ADC SZCMode = Digital Soft Ramp */ + {0x0B, AC_VERB_SET_PROC_COEF, 0x0002 }, + + {0x0B, AC_VERB_SET_COEF_INDEX, CS421X_IDX_DAC_CFG}, + {0x0B, AC_VERB_SET_PROC_COEF, + (0x0002 /* DAC SZCMode = Digital Soft Ramp */ + | 0x0004 /* Mute DAC on FIFO error */ + | 0x0008 /* Enable DAC High Pass Filter */ + )}, + {} /* terminator */ +}; + +/* Errata: CS4210 rev A1 Silicon + * + * http://www.cirrus.com/en/pubs/errata/ + * + * Description: 1. Performance degredation is present in the ADC. + * 2. Speaker output is not completely muted upon HP detect. + * 3. Noise is present when clipping occurs on the amplified + * speaker outputs. + * + * Workaround: The following verb sequence written to the registers during + * initialization will correct the issues listed above. + */ + +static struct hda_verb cs421x_coef_init_verbs_A1_silicon_fixes[] = { + {0x0B, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0006}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x9999}, /* Test mode: on */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x000A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x14CB}, /* Chop double */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x0011}, + {0x0B, AC_VERB_SET_PROC_COEF, 0xA2D0}, /* Increase ADC current */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001A}, + {0x0B, AC_VERB_SET_PROC_COEF, 0x02A9}, /* Mute speaker */ + + {0x0B, AC_VERB_SET_COEF_INDEX, 0x001B}, + {0x0B, AC_VERB_SET_PROC_COEF, 0X1006}, /* Remove noise */ + + {} /* terminator */ +}; + +#include <sound/tlv.h> +/* Speaker Amp Gain is controlled by the vendor widget's coef 4 */ +static const DECLARE_TLV_DB_SCALE(cs421x_speaker_boost_db_scale, 900, 300, 0); + +static int cs421x_boost_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 3; + return 0; +} + +static int cs421x_boost_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL) & 0x0003; + return 0; +} + +static int cs421x_boost_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + + unsigned int vol = ucontrol->value.integer.value[0]; + unsigned int coef = + cs_vendor_coef_get(codec, CS421X_IDX_SPK_CTL); + + coef &= ~0x0003; + coef |= (vol & 0x0003); + cs_vendor_coef_set(codec, CS421X_IDX_SPK_CTL, coef); + + return 1; +} + +static const struct snd_kcontrol_new cs421x_speaker_bost_ctl = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .access = (SNDRV_CTL_ELEM_ACCESS_READWRITE | + SNDRV_CTL_ELEM_ACCESS_TLV_READ), + .name = "Speaker Boost Playback Volume", + .info = cs421x_boost_vol_info, + .get = cs421x_boost_vol_get, + .put = cs421x_boost_vol_put, + .tlv = { .p = cs421x_speaker_boost_db_scale }, +}; + +/* + CS4210 auto-output redirect + HP/SPK/SPDIF +*/ +static void cs421x_auto_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + unsigned int caps, hp_present, spdif_present; + hda_nid_t nid; + int i; + + spdif_present = 0; + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + caps = snd_hda_query_pin_caps(codec, nid); + if (caps & AC_PINCAP_PRES_DETECT) { + /* + TODO: SPDIF output redirect when SENSE_B is enabled. + Shared (SENSE_A) jack (e.g HP/mini-TOSLINK) + assumed. + */ + if (snd_hda_jack_detect(codec, nid) + /* && spec->sense_b */) + spdif_present = 1; + } + } + + hp_present = 0; + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + caps = snd_hda_query_pin_caps(codec, nid); + if (!(caps & AC_PINCAP_PRES_DETECT)) + continue; + hp_present = snd_hda_jack_detect(codec, nid); + if (hp_present) + break; + } + + /* mute speakers is spdif or hp jack is plugged in */ + for (i = 0; i < cfg->speaker_outs; i++) { + nid = cfg->speaker_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (hp_present | spdif_present) ? 0 : PIN_OUT); + } + + /* mute HPs if spdif jack (SENSE_B) is present */ + for (i = 0; i < cfg->hp_outs; i++) { + nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + (spdif_present & spec->sense_b)? 0 : PIN_HP); + } + + /* SPDIF TX on/off */ + if (cfg->dig_outs) { + nid = cfg->dig_out_pins[0]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, + spdif_present ? PIN_OUT:0); + + } + /* Update board GPIOs if neccessary ... */ +} + +/* + CS4210 auto-input redirect + Switch max 3 inputs of a single ADC (nid 3) +*/ +static void cs421x_auto_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + hda_nid_t nid; + unsigned int present; + + nid = cfg->inputs[spec->automic_idx].pin; + present = snd_hda_jack_detect(codec, nid); + + if (present) { + spec->last_input = spec->cur_input; + spec->cur_input = spec->automic_idx; + } else { + spec->cur_input = spec->last_input; + } + + snd_hda_codec_write_cache(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); +} + +static void cs421x_pinmux_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + unsigned int def_conf, coef; + + /* GPIO, DMIC_SCL, DMIC_SDA and SENSE_B are multiplexed */ + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + + if (spec->gpio_mask) + coef |= 0x0008; /* B1,B2 are GPIOs */ + else + coef &= ~0x0008; + + if (spec->sense_b) + coef |= 0x0010; /* B2 is SENSE_B, not inverted */ + else + coef &= ~0x0010; + + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + if ((spec->gpio_mask || spec->sense_b) && + is_active_pin(codec, CS421X_DMIC_PIN_NID)) + { + /* + GPIO or SENSE_B forced - disconnect the DMIC pin. + */ + def_conf = snd_hda_codec_get_pincfg(codec, CS421X_DMIC_PIN_NID); + def_conf &= ~AC_DEFCFG_PORT_CONN; + def_conf |= (AC_JACK_PORT_NONE << AC_DEFCFG_PORT_CONN_SHIFT); + snd_hda_codec_set_pincfg(codec, CS421X_DMIC_PIN_NID, def_conf); + } +} + +/* */ +static void init_cs421x_output(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + /* mute first */ + for (i = 0; i < spec->multiout.num_dacs; i++) + snd_hda_codec_write(codec, spec->multiout.dac_nids[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + if (spec->multiout.hp_nid) + snd_hda_codec_write(codec, spec->multiout.hp_nid, 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + for (i = 0; i < ARRAY_SIZE(spec->multiout.extra_out_nid); i++) { + if (!spec->multiout.extra_out_nid[i]) + break; + snd_hda_codec_write(codec, spec->multiout.extra_out_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); + } + + /* set appropriate pin controls */ + for (i = 0; i < cfg->line_outs; i++) + snd_hda_codec_write(codec, cfg->line_out_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + /* HP */ + for (i = 0; i < cfg->hp_outs; i++) { + hda_nid_t nid = cfg->hp_pins[i]; + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP); + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | HP_EVENT); + spec->hp_detect = 1; + } + } + + /* Speaker */ + for (i = 0; i < cfg->speaker_outs; i++) + snd_hda_codec_write(codec, cfg->speaker_pins[i], 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); + + /* SPDIF is enabled on presence detect */ + if (spec->hp_detect || spec->spdif_detect) + cs421x_auto_output(codec); +} + +static void init_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + unsigned int ctl; + hda_nid_t pin = cfg->inputs[i].pin; + + if (!spec->adc_nid[i]) + continue; + /* set appropriate pin control and mute first */ + ctl = PIN_IN; + if (cfg->inputs[i].type == AUTO_PIN_MIC) { + unsigned int caps = snd_hda_query_pin_caps(codec, pin); + caps >>= AC_PINCAP_VREF_SHIFT; + if (caps & AC_PINCAP_VREF_80) + ctl = PIN_VREF80; + } + + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_PIN_WIDGET_CONTROL, ctl); + snd_hda_codec_write(codec, spec->adc_nid[i], 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(spec->adc_idx[i])); + + if (spec->mic_detect && spec->automic_idx == i) { + snd_hda_codec_write(codec, pin, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | MIC_EVENT); + } + } + + if (spec->mic_detect) + cs421x_auto_input(codec); + else { + spec->cur_adc = spec->adc_nid[spec->cur_input]; + snd_hda_codec_write(codec, spec->cur_adc, 0, + AC_VERB_SET_CONNECT_SEL, + spec->adc_idx[spec->cur_input]); + } + +} + +static void init_cs421x_digital(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + + for (i = 0; i < cfg->dig_outs; i++) { + hda_nid_t nid = cfg->dig_out_pins[i]; + if (!cfg->speaker_outs) + continue; + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + AC_USRSP_EN | SPDIF_EVENT); + spec->spdif_detect = 1; + } + } +} + +static int cs421x_init(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + + snd_hda_sequence_write(codec, cs421x_coef_init_verbs); + snd_hda_sequence_write(codec, cs421x_coef_init_verbs_A1_silicon_fixes); + + cs421x_pinmux_init(codec); + + if (spec->gpio_mask) { + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_MASK, + spec->gpio_mask); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DIRECTION, + spec->gpio_dir); + snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_GPIO_DATA, + spec->gpio_data); + } + + init_cs421x_output(codec); + init_cs421x_input(codec); + init_cs421x_digital(codec); + + return 0; +} + +/* + * CS4210 Input MUX (1 ADC) + */ +static int cs421x_mux_enum_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_info(&spec->input_mux, uinfo); +} + +static int cs421x_mux_enum_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + ucontrol->value.enumerated.item[0] = spec->cur_input; + return 0; +} + +static int cs421x_mux_enum_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct cs_spec *spec = codec->spec; + + return snd_hda_input_mux_put(codec, &spec->input_mux, ucontrol, + spec->adc_nid[0], &spec->cur_input); + +} + +static struct snd_kcontrol_new cs421x_capture_source = +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = cs421x_mux_enum_info, + .get = cs421x_mux_enum_get, + .put = cs421x_mux_enum_put, +}; + +static int cs421x_add_input_volume_control(struct hda_codec *codec, int item) +{ + struct cs_spec* spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + const struct hda_input_mux *imux = &spec->input_mux; + hda_nid_t pin = cfg->inputs[item].pin; + struct snd_kcontrol *kctl; + u32 caps; + + if (!(get_wcaps(codec, pin) & AC_WCAP_IN_AMP)) + return 0; + + caps = query_amp_caps(codec, pin, HDA_INPUT); + caps = (caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT; + if (caps <= 1) + return 0; + + return add_volume(codec, imux->items[item].label, 0, + HDA_COMPOSE_AMP_VAL(pin, 3, 0, HDA_INPUT), 1, &kctl); +} + +/* add a (input-boost) volume control to the given input pin */ +static int build_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct hda_input_mux *imux = &spec->input_mux; + int i, err, type_idx; + const char *label; + + if (!spec->num_inputs) + return 0; + + /* make bind-capture */ + spec->capture_bind[0] = make_bind_capture(codec, &snd_hda_bind_sw); + spec->capture_bind[1] = make_bind_capture(codec, &snd_hda_bind_vol); + for (i = 0; i < 2; i++) { + struct snd_kcontrol *kctl; + int n; + if (!spec->capture_bind[i]) + return -ENOMEM; + kctl = snd_ctl_new1(&cs_capture_ctls[i], codec); + if (!kctl) + return -ENOMEM; + kctl->private_value = (long)spec->capture_bind[i]; + err = snd_hda_ctl_add(codec, 0, kctl); + if (err < 0) + return err; + for (n = 0; n < AUTO_PIN_LAST; n++) { + if (!spec->adc_nid[n]) + continue; + err = snd_hda_add_nid(codec, kctl, 0, spec->adc_nid[n]); + if (err < 0) + return err; + } + } + + /* Add Input MUX Items + Capture Volume/Switch */ + for (i = 0; i < spec->num_inputs; i++) { + label = hda_get_autocfg_input_label(codec, cfg, i); + snd_hda_add_imux_item(imux, label, spec->adc_idx[i], &type_idx); + + err = cs421x_add_input_volume_control(codec, i); + if (err < 0) + return err; + } + + /* + Add 'Capture Source' Switch if + * 2 inputs and no mic detec + * 3 inputs + */ + if ((spec->num_inputs == 2 && !spec->mic_detect) || + (spec->num_inputs == 3) ) + { + err = snd_hda_ctl_add(codec, spec->adc_nid[0], + snd_ctl_new1(&cs421x_capture_source, codec)); + if (err < 0) + return err; + } + + return 0; +} + +/* Single DAC (Mute/Gain) */ +static int build_cs421x_output(struct hda_codec *codec) +{ + hda_nid_t dac = CS4210_DAC_NID; + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + struct snd_kcontrol *kctl; + int err; + char* name = "HP/Speakers"; + + fix_volume_caps(codec, dac); + if (!spec->vmaster_sw) { + err = add_vmaster(codec, dac); + if (err < 0) + return err; + } + + err = add_mute(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_sw, kctl); + if (err < 0) + return err; + + err = add_volume(codec, name, 0, + HDA_COMPOSE_AMP_VAL(dac, 3, 0, HDA_OUTPUT), 0, &kctl); + if (err < 0) + return err; + err = snd_ctl_add_slave(spec->vmaster_vol, kctl); + if (err < 0) + return err; + + if (cfg->speaker_outs) { + err = snd_hda_ctl_add(codec, 0, + snd_ctl_new1(&cs421x_speaker_bost_ctl, codec)); + if (err < 0) + return err; + } + return err; +} + +static int cs421x_build_controls(struct hda_codec *codec) +{ + int err; + + err = build_cs421x_output(codec); + if (err < 0) + return err; + err = build_cs421x_input(codec); + if (err < 0) + return err; + err = build_digital_output(codec); + if (err < 0) + return err; + return cs421x_init(codec); +} + +static void cs421x_unsol_event(struct hda_codec *codec, unsigned int res) +{ + switch ((res >> 26) & 0x3f) { + case HP_EVENT: + case SPDIF_EVENT: + cs421x_auto_output(codec); + break; + + case MIC_EVENT: + cs421x_auto_input(codec); + break; + } +} + +static int parse_cs421x_input(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) { + hda_nid_t pin = cfg->inputs[i].pin; + spec->adc_nid[i] = get_adc(codec, pin, &spec->adc_idx[i]); + spec->cur_input = spec->last_input = i; + spec->num_inputs ++; + + /* check whether the automatic mic switch is available */ + if (is_ext_mic(codec, i) && cfg->num_inputs >= 2) { + spec->mic_detect = 1; + spec->automic_idx = i; + } + } + return 0; +} + +static int cs421x_parse_auto_config(struct hda_codec *codec) +{ + struct cs_spec *spec = codec->spec; + int err; + + err = snd_hda_parse_pin_def_config(codec, &spec->autocfg, NULL); + if (err < 0) + return err; + err = parse_output(codec); + if (err < 0) + return err; + err = parse_cs421x_input(codec); + if (err < 0) + return err; + err = parse_digital_output(codec); + if (err < 0) + return err; + return 0; +} + +#ifdef SND_HDA_NEEDS_RESUME +/* + Manage PDREF, when transitioning to D3hot + (DAC,ADC) -> D3, PDREF=1, AFG->D3 +*/ +static int cs421x_suspend(struct hda_codec *codec, pm_message_t state) +{ + unsigned int coef; + + snd_hda_shutup_pins(codec); + + snd_hda_codec_write(codec, CS4210_DAC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + snd_hda_codec_write(codec, CS4210_ADC_NID, 0, + AC_VERB_SET_POWER_STATE, AC_PWRST_D3); + + coef = cs_vendor_coef_get(codec, CS421X_IDX_DEV_CFG); + coef |= 0x0004; /* PDREF */ + cs_vendor_coef_set(codec, CS421X_IDX_DEV_CFG, coef); + + return 0; +} +#endif + +static struct hda_codec_ops cs4210_patch_ops = { + .build_controls = cs421x_build_controls, + .build_pcms = cs_build_pcms, + .init = cs421x_init, + .free = cs_free, + .unsol_event = cs421x_unsol_event, +#ifdef SND_HDA_NEEDS_RESUME + .suspend = cs421x_suspend, +#endif +}; + +static int patch_cs421x(struct hda_codec *codec) +{ + struct cs_spec *spec; + int err; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (!spec) + return -ENOMEM; + codec->spec = spec; + + spec->vendor_nid = CS421X_VENDOR_NID; + + spec->board_config = + snd_hda_check_board_config(codec, CS421X_MODELS, + cs421x_models, cs421x_cfg_tbl); + if (spec->board_config >= 0) + fix_pincfg(codec, spec->board_config, cs421x_pincfgs); + /* + Setup GPIO/SENSE for each board (if used) + */ + switch (spec->board_config) { + case CS421X_CDB4210: + snd_printd("CS4210 board: %s\n", cs421x_models[spec->board_config]); + +/* spec->gpio_mask = 3; + spec->gpio_dir = 3; + spec->gpio_data = 3; +*/ + spec->sense_b = 1; + + break; + } + + /* + Update the GPIO/DMIC/SENSE_B pinmux before the configuration + is auto-parsed. If GPIO or SENSE_B is forced, DMIC input + is disabled. + */ + cs421x_pinmux_init(codec); + + err = cs421x_parse_auto_config(codec); + if (err < 0) + goto error; + + codec->patch_ops = cs4210_patch_ops; + + return 0; + + error: + kfree(codec->spec); + codec->spec = NULL; + return err; +} + /* * patch entries */ -static const struct hda_codec_preset snd_hda_preset_cirrus[] = { +static struct hda_codec_preset snd_hda_preset_cirrus[] = { { .id = 0x10134206, .name = "CS4206", .patch = patch_cs420x }, { .id = 0x10134207, .name = "CS4207", .patch = patch_cs420x }, + { .id = 0x10134210, .name = "CS4210", .patch = patch_cs421x }, {} /* terminator */ }; MODULE_ALIAS("snd-hda-codec-id:10134206"); MODULE_ALIAS("snd-hda-codec-id:10134207"); +MODULE_ALIAS("snd-hda-codec-id:10134210"); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Cirrus Logic HD-audio codec"); -- 1.7.0.4

2 1

[alsa-devel] [PATCH 0/6] ASoC/MFD: twl6040: PLL handling changes
by Peter Ujfalusi 06 Jul '11

06 Jul '11

Hello, The following series depends on the v6 of twl6040 MFD series. The aim is to simplify the PLL handling for twl6040: - Move the PLL selection control to the codec driver from machine driver - Sample rate constraint simplification for selected PLL - Avoid multiple, duplicated PLL configuration, which can slow down the audio startup. - Harmonize the PLL ID usage among the drivers, so we avoid redundancy. I have generated this series on top of l-o/devel-cleanup branch + my previous twl6040 MFD/Input/ASoC v6 series, since it depends on the changes introduced there. Is it OK, if Tony will be taking this as well to avoid cross tree merge issues? I can put the series on top of my previous series, and offer Tony a pull link. Regards, Peter --- Peter Ujfalusi (6): ASoC: twl6040: Use neutral name for power mode text/enum ASoC: twl6040: Move PLL selection to codec driver ASoC: twl6040: Simplify sample rate constraint handling ASoC: twl6040: Configure PLL only once MFD: twl6040: Remove enum for PLL tracking ASoC: twl6040: No need to convert the PLL ID drivers/mfd/twl6040-core.c | 26 +++--- include/linux/mfd/twl6040.h | 17 ++--- sound/soc/codecs/twl6040.c | 182 ++++++++++++++++++++++--------------------- sound/soc/codecs/twl6040.h | 1 + sound/soc/omap/sdp4430.c | 47 +---------- 5 files changed, 118 insertions(+), 155 deletions(-) -- 1.7.6

2 8

[alsa-devel] [PATCH] ASoC: wm8731: set the ACTIVE bit if bias ON entered
by Nicolas Ferre 05 Jul '11

05 Jul '11

Resuming from a suspend/resume cycle, the ACTIVE bit was not set even if a stream had actually been processing. Setting this bit at entering SND_SOC_BIAS_ON level will ensure this. Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> --- Supersed the wrong patch: "[PATCH 1/5] ASoC: wm8731: rework power management" now based on broonie/sound-2.6 git tree branch: for-3.1. sound/soc/codecs/wm8731.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2dc964b..ee4ec31 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -453,6 +453,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, switch (level) { case SND_SOC_BIAS_ON: + snd_soc_write(codec, WM8731_ACTIVE, 0x0001); break; case SND_SOC_BIAS_PREPARE: break; -- 1.7.3

3 7

[alsa-devel] [PATCH v2] ASoC: Tegra: Implement SPDIF CPU DAI
by Stephen Warren 05 Jul '11

05 Jul '11

This is a minimal driver for the Tegra SPDIF controller. In hardware, the SPDIF output signal is always routed to any active HDMI display controllers, and may also be routed to external pins on Tegra using the pinmux. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> Acked-by: Liam Girdwood <lrg(a)ti.com> --- v2: Fixed tegra_spdif_dai.name; removed trailing ".0". This worked previously because the spdif platform device's ID was -1, which caused fmt_single_name to throw away the "." and anything after it. sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 4 files changed, 855 insertions(+), 0 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h diff --git a/sound/soc/tegra/Kconfig b/sound/soc/tegra/Kconfig index 035d39a..c6af1fd 100644 --- a/sound/soc/tegra/Kconfig +++ b/sound/soc/tegra/Kconfig @@ -12,6 +12,15 @@ config SND_SOC_TEGRA_I2S Tegra I2S interface. You will also need to select the individual machine drivers to support below. +config SND_SOC_TEGRA_SPDIF + tristate + depends on SND_SOC_TEGRA + default m + help + Say Y or M if you want to add support for the SPDIF interface. + You will also need to select the individual machine drivers to support + below. + config MACH_HAS_SND_SOC_TEGRA_WM8903 bool help diff --git a/sound/soc/tegra/Makefile b/sound/soc/tegra/Makefile index fa6574d..4d943b3 100644 --- a/sound/soc/tegra/Makefile +++ b/sound/soc/tegra/Makefile @@ -2,12 +2,14 @@ snd-soc-tegra-das-objs := tegra_das.o snd-soc-tegra-pcm-objs := tegra_pcm.o snd-soc-tegra-i2s-objs := tegra_i2s.o +snd-soc-tegra-spdif-objs := tegra_spdif.o snd-soc-tegra-utils-objs += tegra_asoc_utils.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-utils.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-das.o obj-$(CONFIG_SND_SOC_TEGRA) += snd-soc-tegra-pcm.o obj-$(CONFIG_SND_SOC_TEGRA_I2S) += snd-soc-tegra-i2s.o +obj-$(CONFIG_SND_SOC_TEGRA_SPDIF) += snd-soc-tegra-spdif.o # Tegra machine Support snd-soc-tegra-wm8903-objs := tegra_wm8903.o diff --git a/sound/soc/tegra/tegra_spdif.c b/sound/soc/tegra/tegra_spdif.c new file mode 100644 index 0000000..abe606b --- /dev/null +++ b/sound/soc/tegra/tegra_spdif.c @@ -0,0 +1,371 @@ +/* + * tegra_spdif.c - Tegra SPDIF driver + * + * Author: Stephen Warren <swarren(a)nvidia.com> + * Copyright (C) 2011 - NVIDIA, Inc. + * + * 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 <linux/clk.h> +#include <linux/module.h> +#include <linux/debugfs.h> +#include <linux/device.h> +#include <linux/platform_device.h> +#include <linux/seq_file.h> +#include <linux/slab.h> +#include <linux/io.h> +#include <mach/iomap.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include "tegra_spdif.h" + +#define DRV_NAME "tegra-spdif" + +static inline void tegra_spdif_write(struct tegra_spdif *spdif, u32 reg, + u32 val) +{ + __raw_writel(val, spdif->regs + reg); +} + +static inline u32 tegra_spdif_read(struct tegra_spdif *spdif, u32 reg) +{ + return __raw_readl(spdif->regs + reg); +} + +#ifdef CONFIG_DEBUG_FS +static int tegra_spdif_show(struct seq_file *s, void *unused) +{ +#define REG(r) { r, #r } + static const struct { + int offset; + const char *name; + } regs[] = { + REG(TEGRA_SPDIF_CTRL), + REG(TEGRA_SPDIF_STATUS), + REG(TEGRA_SPDIF_STROBE_CTRL), + REG(TEGRA_SPDIF_DATA_FIFO_CSR), + REG(TEGRA_SPDIF_CH_STA_RX_A), + REG(TEGRA_SPDIF_CH_STA_RX_B), + REG(TEGRA_SPDIF_CH_STA_RX_C), + REG(TEGRA_SPDIF_CH_STA_RX_D), + REG(TEGRA_SPDIF_CH_STA_RX_E), + REG(TEGRA_SPDIF_CH_STA_RX_F), + REG(TEGRA_SPDIF_CH_STA_TX_A), + REG(TEGRA_SPDIF_CH_STA_TX_B), + REG(TEGRA_SPDIF_CH_STA_TX_C), + REG(TEGRA_SPDIF_CH_STA_TX_D), + REG(TEGRA_SPDIF_CH_STA_TX_E), + REG(TEGRA_SPDIF_CH_STA_TX_F), + }; +#undef REG + + struct tegra_spdif *spdif = s->private; + int i; + + for (i = 0; i < ARRAY_SIZE(regs); i++) { + u32 val = tegra_spdif_read(spdif, regs[i].offset); + seq_printf(s, "%s = %08x\n", regs[i].name, val); + } + + return 0; +} + +static int tegra_spdif_debug_open(struct inode *inode, struct file *file) +{ + return single_open(file, tegra_spdif_show, inode->i_private); +} + +static const struct file_operations tegra_spdif_debug_fops = { + .open = tegra_spdif_debug_open, + .read = seq_read, + .llseek = seq_lseek, + .release = single_release, +}; + +static void tegra_spdif_debug_add(struct tegra_spdif *spdif) +{ + spdif->debug = debugfs_create_file(DRV_NAME, S_IRUGO, + snd_soc_debugfs_root, spdif, + &tegra_spdif_debug_fops); +} + +static void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +{ + if (spdif->debug) + debugfs_remove(spdif->debug); +} +#else +static inline void tegra_spdif_debug_add(struct tegra_spdif *spdif) +{ +} + +static inline void tegra_spdif_debug_remove(struct tegra_spdif *spdif) +{ +} +#endif + +static int tegra_spdif_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct device *dev = substream->pcm->card->dev; + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + int ret, srate, spdifclock; + + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_PACK; + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_BIT_MODE_MASK; + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_PACK; + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_BIT_MODE_16BIT; + break; + default: + return -EINVAL; + } + + srate = params_rate(params); + switch (params_rate(params)) { + case 32000: + spdifclock = 4096000; + break; + case 44100: + spdifclock = 5644800; + break; + case 48000: + spdifclock = 6144000; + break; + case 88200: + spdifclock = 11289600; + break; + case 96000: + spdifclock = 12288000; + break; + case 176400: + spdifclock = 22579200; + break; + case 192000: + spdifclock = 24576000; + break; + default: + return -EINVAL; + } + + ret = clk_set_rate(spdif->clk_spdif_out, spdifclock); + if (ret) { + dev_err(dev, "Can't set SPDIF clock rate: %d\n", ret); + return ret; + } + + return 0; +} + +static void tegra_spdif_start_playback(struct tegra_spdif *spdif) +{ + spdif->reg_ctrl |= TEGRA_SPDIF_CTRL_TX_EN; + tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); +} + +static void tegra_spdif_stop_playback(struct tegra_spdif *spdif) +{ + spdif->reg_ctrl &= ~TEGRA_SPDIF_CTRL_TX_EN; + tegra_spdif_write(spdif, TEGRA_SPDIF_CTRL, spdif->reg_ctrl); +} + +static int tegra_spdif_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + if (!spdif->clk_refs) + clk_enable(spdif->clk_spdif_out); + spdif->clk_refs++; + tegra_spdif_start_playback(spdif); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + tegra_spdif_stop_playback(spdif); + spdif->clk_refs--; + if (!spdif->clk_refs) + clk_disable(spdif->clk_spdif_out); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int tegra_spdif_probe(struct snd_soc_dai *dai) +{ + struct tegra_spdif *spdif = snd_soc_dai_get_drvdata(dai); + + dai->capture_dma_data = NULL; + dai->playback_dma_data = &spdif->playback_dma_data; + + return 0; +} + +static struct snd_soc_dai_ops tegra_spdif_dai_ops = { + .hw_params = tegra_spdif_hw_params, + .trigger = tegra_spdif_trigger, +}; + +struct snd_soc_dai_driver tegra_spdif_dai = { + .name = DRV_NAME, + .probe = tegra_spdif_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &tegra_spdif_dai_ops, +}; + +static __devinit int tegra_spdif_platform_probe(struct platform_device *pdev) +{ + struct tegra_spdif *spdif; + struct resource *mem, *memregion, *dmareq; + int ret; + + spdif = kzalloc(sizeof(struct tegra_spdif), GFP_KERNEL); + if (!spdif) { + dev_err(&pdev->dev, "Can't allocate tegra_spdif\n"); + ret = -ENOMEM; + goto exit; + } + dev_set_drvdata(&pdev->dev, spdif); + + spdif->clk_spdif_out = clk_get(&pdev->dev, "spdif_out"); + if (IS_ERR(spdif->clk_spdif_out)) { + pr_err("Can't retrieve spdif clock\n"); + ret = PTR_ERR(spdif->clk_spdif_out); + goto err_free; + } + + mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!mem) { + dev_err(&pdev->dev, "No memory resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!dmareq) { + dev_err(&pdev->dev, "No DMA resource\n"); + ret = -ENODEV; + goto err_clk_put; + } + + memregion = request_mem_region(mem->start, resource_size(mem), + DRV_NAME); + if (!memregion) { + dev_err(&pdev->dev, "Memory region already claimed\n"); + ret = -EBUSY; + goto err_clk_put; + } + + spdif->regs = ioremap(mem->start, resource_size(mem)); + if (!spdif->regs) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENOMEM; + goto err_release; + } + + spdif->playback_dma_data.addr = mem->start + TEGRA_SPDIF_DATA_OUT; + spdif->playback_dma_data.wrap = 4; + spdif->playback_dma_data.width = 32; + spdif->playback_dma_data.req_sel = dmareq->start; + + ret = snd_soc_register_dai(&pdev->dev, &tegra_spdif_dai); + if (ret) { + dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); + ret = -ENOMEM; + goto err_unmap; + } + + tegra_spdif_debug_add(spdif); + + return 0; + +err_unmap: + iounmap(spdif->regs); +err_release: + release_mem_region(mem->start, resource_size(mem)); +err_clk_put: + clk_put(spdif->clk_spdif_out); +err_free: + kfree(spdif); +exit: + return ret; +} + +static int __devexit tegra_spdif_platform_remove(struct platform_device *pdev) +{ + struct tegra_spdif *spdif = dev_get_drvdata(&pdev->dev); + struct resource *res; + + snd_soc_unregister_dai(&pdev->dev); + + tegra_spdif_debug_remove(spdif); + + iounmap(spdif->regs); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + release_mem_region(res->start, resource_size(res)); + + clk_put(spdif->clk_spdif_out); + + kfree(spdif); + + return 0; +} + +static struct platform_driver tegra_spdif_driver = { + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, + .probe = tegra_spdif_platform_probe, + .remove = __devexit_p(tegra_spdif_platform_remove), +}; + +static int __init snd_tegra_spdif_init(void) +{ + return platform_driver_register(&tegra_spdif_driver); +} +module_init(snd_tegra_spdif_init); + +static void __exit snd_tegra_spdif_exit(void) +{ + platform_driver_unregister(&tegra_spdif_driver); +} +module_exit(snd_tegra_spdif_exit); + +MODULE_AUTHOR("Stephen Warren <swarren(a)nvidia.com>"); +MODULE_DESCRIPTION("Tegra SPDIF ASoC driver"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/tegra/tegra_spdif.h b/sound/soc/tegra/tegra_spdif.h new file mode 100644 index 0000000..2e03db4 --- /dev/null +++ b/sound/soc/tegra/tegra_spdif.h @@ -0,0 +1,473 @@ +/* + * tegra_spdif.h - Definitions for Tegra SPDIF driver + * + * Author: Stephen Warren <swarren(a)nvidia.com> + * Copyright (C) 2011 - NVIDIA, Inc. + * + * Based on code copyright/by: + * Copyright (c) 2008-2009, NVIDIA Corporation + * + * 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 + * + */ + +#ifndef __TEGRA_SPDIF_H__ +#define __TEGRA_SPDIF_H__ + +#include "tegra_pcm.h" + +/* Offsets from TEGRA_SPDIF_BASE */ + +#define TEGRA_SPDIF_CTRL 0x0 +#define TEGRA_SPDIF_STATUS 0x4 +#define TEGRA_SPDIF_STROBE_CTRL 0x8 +#define TEGRA_SPDIF_DATA_FIFO_CSR 0x0C +#define TEGRA_SPDIF_DATA_OUT 0x40 +#define TEGRA_SPDIF_DATA_IN 0x80 +#define TEGRA_SPDIF_CH_STA_RX_A 0x100 +#define TEGRA_SPDIF_CH_STA_RX_B 0x104 +#define TEGRA_SPDIF_CH_STA_RX_C 0x108 +#define TEGRA_SPDIF_CH_STA_RX_D 0x10C +#define TEGRA_SPDIF_CH_STA_RX_E 0x110 +#define TEGRA_SPDIF_CH_STA_RX_F 0x114 +#define TEGRA_SPDIF_CH_STA_TX_A 0x140 +#define TEGRA_SPDIF_CH_STA_TX_B 0x144 +#define TEGRA_SPDIF_CH_STA_TX_C 0x148 +#define TEGRA_SPDIF_CH_STA_TX_D 0x14C +#define TEGRA_SPDIF_CH_STA_TX_E 0x150 +#define TEGRA_SPDIF_CH_STA_TX_F 0x154 +#define TEGRA_SPDIF_USR_STA_RX_A 0x180 +#define TEGRA_SPDIF_USR_DAT_TX_A 0x1C0 + +/* Fields in TEGRA_SPDIF_CTRL */ + +/* Start capturing from 0=right, 1=left channel */ +#define TEGRA_SPDIF_CTRL_CAP_LC (1 << 30) + +/* SPDIF receiver(RX) enable */ +#define TEGRA_SPDIF_CTRL_RX_EN (1 << 29) + +/* SPDIF Transmitter(TX) enable */ +#define TEGRA_SPDIF_CTRL_TX_EN (1 << 28) + +/* Transmit Channel status */ +#define TEGRA_SPDIF_CTRL_TC_EN (1 << 27) + +/* Transmit user Data */ +#define TEGRA_SPDIF_CTRL_TU_EN (1 << 26) + +/* Interrupt on transmit error */ +#define TEGRA_SPDIF_CTRL_IE_TXE (1 << 25) + +/* Interrupt on receive error */ +#define TEGRA_SPDIF_CTRL_IE_RXE (1 << 24) + +/* Interrupt on invalid preamble */ +#define TEGRA_SPDIF_CTRL_IE_P (1 << 23) + +/* Interrupt on "B" preamble */ +#define TEGRA_SPDIF_CTRL_IE_B (1 << 22) + +/* Interrupt when block of channel status received */ +#define TEGRA_SPDIF_CTRL_IE_C (1 << 21) + +/* Interrupt when a valid information unit (IU) is received */ +#define TEGRA_SPDIF_CTRL_IE_U (1 << 20) + +/* Interrupt when RX user FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_RU (1 << 19) + +/* Interrupt when TX user FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_TU (1 << 18) + +/* Interrupt when RX data FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_RX (1 << 17) + +/* Interrupt when TX data FIFO attention level is reached */ +#define TEGRA_SPDIF_CTRL_QE_TX (1 << 16) + +/* Loopback test mode enable */ +#define TEGRA_SPDIF_CTRL_LBK_EN (1 << 15) + +/* + * Pack data mode: + * 0 = Single data (16 bit needs to be padded to match the + * interface data bit size). + * 1 = Packeted left/right channel data into a single word. + */ +#define TEGRA_SPDIF_CTRL_PACK (1 << 14) + +/* + * 00 = 16bit data + * 01 = 20bit data + * 10 = 24bit data + * 11 = raw data + */ +#define TEGRA_SPDIF_BIT_MODE_16BIT 0 +#define TEGRA_SPDIF_BIT_MODE_20BIT 1 +#define TEGRA_SPDIF_BIT_MODE_24BIT 2 +#define TEGRA_SPDIF_BIT_MODE_RAW 3 + +#define TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT 12 +#define TEGRA_SPDIF_CTRL_BIT_MODE_MASK (3 << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_16BIT (TEGRA_SPDIF_BIT_MODE_16BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_20BIT (TEGRA_SPDIF_BIT_MODE_20BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_24BIT (TEGRA_SPDIF_BIT_MODE_24BIT << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) +#define TEGRA_SPDIF_CTRL_BIT_MODE_RAW (TEGRA_SPDIF_BIT_MODE_RAW << TEGRA_SPDIF_CTRL_BIT_MODE_SHIFT) + +/* Fields in TEGRA_SPDIF_STATUS */ + +/* + * Note: IS_P, IS_B, IS_C, and IS_U are sticky bits. Software must + * write a 1 to the corresponding bit location to clear the status. + */ + +/* + * Receiver(RX) shifter is busy receiving data. + * This bit is asserted when the receiver first locked onto the + * preamble of the data stream after RX_EN is asserted. This bit is + * deasserted when either, + * (a) the end of a frame is reached after RX_EN is deeasserted, or + * (b) the SPDIF data stream becomes inactive. + */ +#define TEGRA_SPDIF_STATUS_RX_BSY (1 << 29) + +/* + * Transmitter(TX) shifter is busy transmitting data. + * This bit is asserted when TX_EN is asserted. + * This bit is deasserted when the end of a frame is reached after + * TX_EN is deasserted. + */ +#define TEGRA_SPDIF_STATUS_TX_BSY (1 << 28) + +/* + * TX is busy shifting out channel status. + * This bit is asserted when both TX_EN and TC_EN are asserted and + * data from CH_STA_TX_A register is loaded into the internal shifter. + * This bit is deasserted when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) CH_STA_TX_F register is loaded into the internal shifter. + */ +#define TEGRA_SPDIF_STATUS_TC_BSY (1 << 27) + +/* + * TX User data FIFO busy. + * This bit is asserted when TX_EN and TXU_EN are asserted and + * there's data in the TX user FIFO. This bit is deassert when either, + * (a) the end of a frame is reached after TX_EN is deasserted, or + * (b) there's no data left in the TX user FIFO. + */ +#define TEGRA_SPDIF_STATUS_TU_BSY (1 << 26) + +/* TX FIFO Underrun error status */ +#define TEGRA_SPDIF_STATUS_TX_ERR (1 << 25) + +/* RX FIFO Overrun error status */ +#define TEGRA_SPDIF_STATUS_RX_ERR (1 << 24) + +/* Preamble status: 0=Preamble OK, 1=bad/missing preamble */ +#define TEGRA_SPDIF_STATUS_IS_P (1 << 23) + +/* B-preamble detection status: 0=not detected, 1=B-preamble detected */ +#define TEGRA_SPDIF_STATUS_IS_B (1 << 22) + +/* + * RX channel block data receive status: + * 0=entire block not recieved yet. + * 1=received entire block of channel status, + */ +#define TEGRA_SPDIF_STATUS_IS_C (1 << 21) + +/* RX User Data Valid flag: 1=valid IU detected, 0 = no IU detected. */ +#define TEGRA_SPDIF_STATUS_IS_U (1 << 20) + +/* + * RX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_RU (1 << 19) + +/* + * TX User FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_TU (1 << 18) + +/* + * RX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_RX (1 << 17) + +/* + * TX Data FIFO Status: + * 1=attention level reached, 0=attention level not reached. + */ +#define TEGRA_SPDIF_STATUS_QS_TX (1 << 16) + +/* Fields in TEGRA_SPDIF_STROBE_CTRL */ + +/* + * Indicates the approximate number of detected SPDIFIN clocks within a + * bi-phase period. + */ +#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT 16 +#define TEGRA_SPDIF_STROBE_CTRL_PERIOD_MASK (0xff << TEGRA_SPDIF_STROBE_CTRL_PERIOD_SHIFT) + +/* Data strobe mode: 0=Auto-locked 1=Manual locked */ +#define TEGRA_SPDIF_STROBE_CTRL_STROBE (1 << 15) + +/* + * Manual data strobe time within the bi-phase clock period (in terms of + * the number of over-sampling clocks). + */ +#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT 8 +#define TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_MASK (0x1f << TEGRA_SPDIF_STROBE_CTRL_DATA_STROBES_SHIFT) + +/* + * Manual SPDIFIN bi-phase clock period (in terms of the number of + * over-sampling clocks). + */ +#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT 0 +#define TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_MASK (0x3f << TEGRA_SPDIF_STROBE_CTRL_CLOCK_PERIOD_SHIFT) + +/* Fields in SPDIF_DATA_FIFO_CSR */ + +/* Clear Receiver User FIFO (RX USR.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_CLR (1 << 31) + +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT 0 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS 1 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS 2 +#define TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT 29 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU2_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU3_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_RU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RU_ATN_LVL_SHIFT) + +/* Number of RX USR.FIFO levels with valid data. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT 24 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RU_FULL_COUNT_SHIFT) + +/* Clear Transmitter User FIFO (TX USR.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_CLR (1 << 23) + +/* TU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT 21 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU2_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_TWO_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU3_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_THREE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_TU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_U_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TU_ATN_LVL_SHIFT) + +/* Number of TX USR.FIFO levels that could be filled. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT 16 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TU_EMPTY_COUNT_SHIFT) + +/* Clear Receiver Data FIFO (RX DATA.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_CLR (1 << 15) + +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT 0 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS 1 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS 2 +#define TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS 3 + +/* RU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT 13 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU8_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_RU12_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_RX_ATN_LVL_SHIFT) + +/* Number of RX DATA.FIFO levels with valid data. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT 8 +#define TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_MASK (0x1f << TEGRA_SPDIF_DATA_FIFO_CSR_RX_FULL_COUNT_SHIFT) + +/* Clear Transmitter Data FIFO (TX DATA.FIFO) */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_CLR (1 << 7) + +/* TU FIFO attention level */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT 5 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_MASK \ + (0x3 << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU1_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_ONE_SLOT << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU4_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_FOUR_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU8_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_EIGHT_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_TU12_WORD_FULL \ + (TEGRA_SPDIF_FIFO_ATN_LVL_D_TWELVE_SLOTS << TEGRA_SPDIF_DATA_FIFO_CSR_TX_ATN_LVL_SHIFT) + +/* Number of TX DATA.FIFO levels that could be filled. */ +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT 0 +#define TEGRA_SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_MASK (0x1f << SPDIF_DATA_FIFO_CSR_TX_EMPTY_COUNT_SHIFT) + +/* Fields in TEGRA_SPDIF_DATA_OUT */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + */ + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_20_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_OUT_DATA_24_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_P (1 << 31) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_C (1 << 30) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_U (1 << 29) +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_V (1 << 28) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT 8 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_OUT_DATA_RAW_DATA_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT 4 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_AUX_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_OUT_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_OUT_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA_SPDIF_DATA_IN */ + +/* + * This register has 5 different formats: + * 16-bit (BIT_MODE=00, PACK=0) + * 20-bit (BIT_MODE=01, PACK=0) + * 24-bit (BIT_MODE=10, PACK=0) + * raw (BIT_MODE=11, PACK=0) + * 16-bit packed (BIT_MODE=00, PACK=1) + * + * Bits 31:24 are common to all modes except 16-bit packed + */ + +#define TEGRA_SPDIF_DATA_IN_DATA_P (1 << 31) +#define TEGRA_SPDIF_DATA_IN_DATA_C (1 << 30) +#define TEGRA_SPDIF_DATA_IN_DATA_U (1 << 29) +#define TEGRA_SPDIF_DATA_IN_DATA_V (1 << 28) + +#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT 24 +#define TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_16_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_20_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_20_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_24_MASK (0xffffff << TEGRA_SPDIF_DATA_IN_DATA_24_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT 8 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_MASK (0xfffff << TEGRA_SPDIF_DATA_IN_DATA_RAW_DATA_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT 4 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_AUX_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_MASK (0xf << TEGRA_SPDIF_DATA_IN_DATA_RAW_PREAMBLE_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT 16 +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_RIGHT_SHIFT) + +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT 0 +#define TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_MASK (0xffff << TEGRA_SPDIF_DATA_IN_DATA_16_PACKED_LEFT_SHIFT) + +/* Fields in TEGRA_SPDIF_CH_STA_RX_A */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_B */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_C */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_D */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_E */ +/* Fields in TEGRA_SPDIF_CH_STA_RX_F */ + +/* + * The 6-word receive channel data page buffer holds a block (192 frames) of + * channel status information. The order of receive is from LSB to MSB + * bit, and from CH_STA_RX_A to CH_STA_RX_F then back to CH_STA_RX_A. + */ + +/* Fields in TEGRA_SPDIF_CH_STA_TX_A */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_B */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_C */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_D */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_E */ +/* Fields in TEGRA_SPDIF_CH_STA_TX_F */ + +/* + * The 6-word transmit channel data page buffer holds a block (192 frames) of + * channel status information. The order of transmission is from LSB to MSB + * bit, and from CH_STA_TX_A to CH_STA_TX_F then back to CH_STA_TX_A. + */ + +/* Fields in TEGRA_SPDIF_USR_STA_RX_A */ + +/* + * This 4-word deep FIFO receives user FIFO field information. The order of + * receive is from LSB to MSB bit. + */ + +/* Fields in TEGRA_SPDIF_USR_DAT_TX_A */ + +/* + * This 4-word deep FIFO transmits user FIFO field information. The order of + * transmission is from LSB to MSB bit. + */ + +struct tegra_spdif { + struct clk *clk_spdif_out; + int clk_refs; + struct tegra_pcm_dma_params capture_dma_data; + struct tegra_pcm_dma_params playback_dma_data; + void __iomem *regs; + struct dentry *debug; + u32 reg_ctrl; +}; + +#endif -- 1.7.0.4

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[alsa-devel] [PATCH] ASoC: core - Add platform IO tracing
by Liam Girdwood 05 Jul '11

05 Jul '11

Trace platform IO just like CODEC IO. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/trace/events/asoc.h | 45 +++++++++++++++++++++++++++++++++++++++++++ sound/soc/soc-core.c | 2 + 2 files changed, 47 insertions(+), 0 deletions(-) diff --git a/include/trace/events/asoc.h b/include/trace/events/asoc.h index ae973d2..603f5a0 100644 --- a/include/trace/events/asoc.h +++ b/include/trace/events/asoc.h @@ -9,6 +9,7 @@ struct snd_soc_jack; struct snd_soc_codec; +struct snd_soc_platform; struct snd_soc_card; struct snd_soc_dapm_widget; @@ -59,6 +60,50 @@ DEFINE_EVENT(snd_soc_reg, snd_soc_reg_read, ); +DECLARE_EVENT_CLASS(snd_soc_preg, + + TP_PROTO(struct snd_soc_platform *platform, unsigned int reg, + unsigned int val), + + TP_ARGS(platform, reg, val), + + TP_STRUCT__entry( + __string( name, platform->name ) + __field( int, id ) + __field( unsigned int, reg ) + __field( unsigned int, val ) + ), + + TP_fast_assign( + __assign_str(name, platform->name); + __entry->id = platform->id; + __entry->reg = reg; + __entry->val = val; + ), + + TP_printk("platform=%s.%d reg=%x val=%x", __get_str(name), + (int)__entry->id, (unsigned int)__entry->reg, + (unsigned int)__entry->val) +); + +DEFINE_EVENT(snd_soc_preg, snd_soc_preg_write, + + TP_PROTO(struct snd_soc_platform *platform, unsigned int reg, + unsigned int val), + + TP_ARGS(platform, reg, val) + +); + +DEFINE_EVENT(snd_soc_preg, snd_soc_preg_read, + + TP_PROTO(struct snd_soc_platform *platform, unsigned int reg, + unsigned int val), + + TP_ARGS(platform, reg, val) + +); + DECLARE_EVENT_CLASS(snd_soc_card, TP_PROTO(struct snd_soc_card *card, int val), diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d08abf4..06c9b14 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1652,6 +1652,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, ret = platform->driver->read(platform, reg); dev_dbg(platform->dev, "read %x => %x\n", reg, ret); + trace_snd_soc_preg_read(platform, reg, ret); return ret; } @@ -1666,6 +1667,7 @@ int snd_soc_platform_write(struct snd_soc_platform *platform, } dev_dbg(platform->dev, "write %x = %x\n", reg, val); + trace_snd_soc_preg_write(platform, reg, val); return platform->driver->write(platform, reg, val); } EXPORT_SYMBOL_GPL(snd_soc_platform_write); -- 1.7.4.1

2 4

[alsa-devel] [PATCH v2 0/5] MFD/input/ASoC: twl6040: irq registration changes
by Peter Ujfalusi 05 Jul '11

05 Jul '11

Hello, Changes since v1: - IRQ number is passed with resource to the slave devices Intro mail froim v1: As Samuel, and Felipe requested for the twl6040 MFD series: http://marc.info/?l=linux-kernel&m=130978512225430&w=2 http://marc.info/?l=linux-omap&m=130978403224218&w=2 This is the update to remove the obfuscation around the threaded irq request in the twl6040 driver and it's slaves. With this change the slaves uses directly the threaded irq request, and the core driver passes the irq number for the slaves to use (configured runtime, so machine drivers do not need to be updated). Regards, Peter PS: for the remaining comments from Samuel I have changed the series, since the requested change was small. I'm not intended to send the series again, unless it is requested (again the change from v6 is minimal - few lines has been removed/replaced). --- Peter Ujfalusi (5): MFD: twl6040: Use resource to provide irq number for slaves input: twl6040-vibra: Do not use wrapper for irq request ASoC: twl6040: Do not use wrapper for irq request MFD: twl6040: Demand valid interrupt configuration MFD: twl6040: Remove wrapper for threaded irq request drivers/input/misc/twl6040-vibra.c | 17 ++++++--- drivers/mfd/twl6040-core.c | 69 ++++++++++++++++++++++++----------- drivers/mfd/twl6040-irq.c | 16 +-------- include/linux/mfd/twl6040.h | 22 ----------- sound/soc/codecs/twl6040.c | 19 +++++++--- 5 files changed, 74 insertions(+), 69 deletions(-) -- 1.7.6

2 6

Re: [alsa-devel] [PATCH v2 2/5] input: twl6040-vibra: Do not use wrapper for irq request
by Péter Ujfalusi 05 Jul '11

05 Jul '11

On Tuesday 05 July 2011 15:57:46 Balbi, Felipe wrote: > > + info->irq = platform_get_irq(pdev, 0); > > + if (!info->irq) { > > this is wrong. platform_get_irq() will return -ENXIO if it can't find > the IRQ you requested: > > 55 /** > 56 * platform_get_irq - get an IRQ for a device > 57 * @dev: platform device > 58 * @num: IRQ number index > 59 */ > 60 int platform_get_irq(struct platform_device *dev, unsigned int num) > 61 { > 62 struct resource *r = platform_get_resource(dev, IORESOURCE_IRQ, > num); 63 > 64 return r ? r->start : -ENXIO; > 65 } > 66 EXPORT_SYMBOL_GPL(platform_get_irq); Yes, I have not checked that.. Will replace the check with (info->irq< 0) checks in both slaves. Thanks for noticing it! I'll wait a bit for other comments, and will resend. -- Péter

1 0

Re: [alsa-devel] [PATCH 1/5] MFD: twl6040: Add irq variable for slave devices
by Péter Ujfalusi 05 Jul '11

05 Jul '11

On Tuesday 05 July 2011 13:22:46 Balbi, Felipe wrote: > Hi, > > On Tue, Jul 05, 2011 at 12:14:22PM +0300, Peter Ujfalusi wrote: > > Add new variable to pass the irq number for the slaves > > devices of twl6040. > > The irq number configuration is done in the twl6040-core > > at probe time, so machine drivers do not need to be > > modified. > > > > Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> > > --- > > > > drivers/mfd/twl6040-core.c | 2 ++ > > include/linux/i2c/twl.h | 2 ++ > > 2 files changed, 4 insertions(+), 0 deletions(-) > > > > diff --git a/drivers/mfd/twl6040-core.c b/drivers/mfd/twl6040-core.c > > index cfaedb5..8607793 100644 > > --- a/drivers/mfd/twl6040-core.c > > +++ b/drivers/mfd/twl6040-core.c > > @@ -501,6 +501,7 @@ static int __devinit twl6040_probe(struct > > platform_device *pdev) > > > > if (pdata->codec) { > > > > cell = &twl6040->cells[children]; > > cell->name = "twl6040-codec"; > > > > + pdata->codec->plug_irq = twl6040->irq_base + TWL6040_IRQ_PLUG; > > are the slave devices platform_devices ?? If so, you could use struct > resource to pass IRQs as that's the "approved" way to do so. Then, on > child driver, you could either rely on a particular order of the > resources or assign them names and use platform_get_irq_byname() to > fetch the correct one (if you have more than one, that is). You mean something like this: static struct resource twl6040_codec_rsrc[] = { { .flags = IORESOURCE_IRQ, }, }; ... @@ -501,6 +501,7 @@ static int __devinit twl6040_probe(struct platform_device *pdev) if (pdata->codec) { cell = &twl6040->cells[children]; cell->name = "twl6040-codec"; + twl6040_codec_rsrc[0].start = twl6040->irq_base + TWL6040_IRQ_PLUG; + twl6040_codec_rsrc[0].end = twl6040->irq_base + TWL6040_IRQ_PLUG; + cell->resource = twl6040_codec_rsrc; + cell->num_resources = ARRAY_SIZE(twl6040_codec_rsrc); cell->platform_data = pdata->codec; cell->pdata_size = sizeof(*pdata->codec); children++; -- Péter

1 0

[alsa-devel] Recording doesn't end gracefully
by Santosh Sivaraj 05 Jul '11

05 Jul '11

Hi all, We were working on 1.0.23 version of the ALSA driver. In the following scenario we are getting some issues with recording. * Play some music in the sound player in the background * Open recorder and start recording * Since playback and recording can't happen simultaneously, we explicitly reset the codec device and configure it for recording. * In this case, after some time the codec driver is powered off by the soc-core and only silence is recorded. All this were done from the Android UI. When the same is repeated from the command line with aplay and arecord we see that aplay is stopped and recording works fine. When we debugged the problem, we found that the following diff seems to get rid of the above problem: diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c984996..247f4ee 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -773,7 +773,8 @@ int snd_soc_pcm_close(struct snd_pcm_substream *substream) /* Muting the DAC suppresses artifacts caused during digital * shutdown, for example from stopping clocks. */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if ((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) snd_soc_dai_digital_mute(codec_dai, 1); if (cpu_dai->driver->ops->shutdown) @@ -869,7 +870,8 @@ int snd_soc_pcm_prepare(struct snd_pcm_substream *substream) } /* cancel any delayed stream shutdown that is pending */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && + if (((substream->stream == SNDRV_PCM_STREAM_PLAYBACK) || + (substream->stream == SNDRV_PCM_STREAM_CAPTURE)) && codec_dai->pop_wait) { codec_dai->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); I am not aware of what consequences/side effects these changes can cause, please advice, whether it is a problem elsewhere -- http://fossix.org, http://dot.slashb.in Things do not change; we change -- Henry David

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 05 Jul '11

05 Jul '11

Hi Samuel, On Monday 04 July 2011 19:39:35 Samuel Ortiz wrote: > That is fine with me, yes. That's why I ACK the MFD patches for Tony to take > them. I have changed the series for Tony in my branch for the following comments: wl6040_is_powered, twl6040_get_rev removal, since these are really small changes, if you want to see it before Tony pulls it, I can send the selected patches (few lines has been changed). > Thanks in advance for the follow up patches. For the twl6040_*_irq changes, I sent the series, since I ended up with a bigger change than I would have expected. Regards, Péter

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[alsa-devel] [PATCH 0/5] MFD/input/ASoC: twl6040: irq registration changes
by Peter Ujfalusi 05 Jul '11

05 Jul '11

Hello, As Samuel, and Felipe requested for the twl6040 MFD series: http://marc.info/?l=linux-kernel&m=130978512225430&w=2 http://marc.info/?l=linux-omap&m=130978403224218&w=2 This is the update to remove the obfuscation around the threaded irq request in the twl6040 driver and it's slaves. With this change the slaves uses directly the threaded irq request, and the core driver passes the irq number for the slaves to use (configured runtime, so machine drivers do not need to be updated). Regards, Peter PS: for the remaining comments from Samuel I have changed the series, since the requested change was small. I'm not intended to send the series again, unless it is requested (again the change from v6 is minimal - few lines has been removed/replaced). --- Peter Ujfalusi (5): MFD: twl6040: Add irq variable for slave devices input: twl6040-vibra: Do not use wrapper for irq request ASoC: twl6040: Do not use wrapper for irq request MFD: twl6040: Demand valid interrupt configuration MFD: twl6040: Remove wrapper for threaded irq request drivers/input/misc/twl6040-vibra.c | 17 +++++++++---- drivers/mfd/twl6040-core.c | 46 ++++++++++++++++++----------------- drivers/mfd/twl6040-irq.c | 16 +----------- include/linux/i2c/twl.h | 2 + include/linux/mfd/twl6040.h | 22 ----------------- sound/soc/codecs/twl6040.c | 17 +++++++++---- 6 files changed, 51 insertions(+), 69 deletions(-) -- 1.7.6

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 05 Jul '11

05 Jul '11

The following changes since commit 56aa53391059e3730a304da4dd96b7b123b9fb75: Merge branch 'for-3.1' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-07-05 07:33:23 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Lars-Peter Clausen (2): ASoC: Add ADAV80x codec driver ASoC: Blackfin: Add machine driver for EVAL-ADAV80X boards Liam Girdwood (2): ASoC: core - Make platform probe more like codec probe. ASoC: core - Add platform read and write. Mark Brown (7): ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values ASoC: Correct left/right swap in wm_hubs DC offset correction ASoC: Change WM9081 speaker output enable to _OUT_DRV Merge branch 'for-3.0' into for-3.1 ASoC: Improve error reporting in Speyside WM8962 driver ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting Merge branch 'for-3.0' into for-3.1 Sangbeom Kim (3): ASoC: SAMSUNG: Add WM8994 PCM Machine driver ASoC: SAMSUNG: Move I2S common register definition ASoC: SAMSUNG: Add idma related register definition Stephen Warren (2): ASoC: Tegra: I2S: Ensure clock is enabled when writing regs ASoC: Tegra: I2S: s/clk_get_sys/clk_get/ MAINTAINERS | 1 + include/sound/soc.h | 8 + sound/soc/blackfin/Kconfig | 13 + sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-i2s-pcm.c | 13 +- sound/soc/blackfin/bfin-eval-adav80x.c | 173 ++++++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 951 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adav80x.h | 35 ++ sound/soc/codecs/wm8994.c | 2 + sound/soc/codecs/wm9081.c | 2 +- sound/soc/codecs/wm_hubs.c | 16 +- sound/soc/samsung/Kconfig | 8 + sound/soc/samsung/Makefile | 2 + sound/soc/samsung/i2s-regs.h | 143 +++++ sound/soc/samsung/i2s.c | 104 +---- sound/soc/samsung/smdk_wm8994pcm.c | 176 ++++++ sound/soc/samsung/speyside_wm8962.c | 12 +- sound/soc/soc-core.c | 81 +++- sound/soc/tegra/tegra_i2s.c | 10 +- 21 files changed, 1622 insertions(+), 136 deletions(-) create mode 100644 sound/soc/blackfin/bfin-eval-adav80x.c create mode 100644 sound/soc/codecs/adav80x.c create mode 100644 sound/soc/codecs/adav80x.h create mode 100644 sound/soc/samsung/i2s-regs.h create mode 100644 sound/soc/samsung/smdk_wm8994pcm.c

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 05 Jul '11

05 Jul '11

The following changes since commit 63bc9750167ac765827470aa5aed1e79aeaaad9e: Merge branch 'for-3.0' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into fix/asoc (2011-07-05 07:33:06 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Mark Brown (2): ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values ASoC: Ensure we delay long enough for WM8994 FLL to lock when starting Stephen Warren (1): ASoC: Tegra: I2S: Ensure clock is enabled when writing regs sound/soc/blackfin/bf5xx-i2s-pcm.c | 13 +++++++++++-- sound/soc/codecs/wm8994.c | 2 ++ sound/soc/tegra/tegra_i2s.c | 6 ++++++ 3 files changed, 19 insertions(+), 2 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for 3.1
by Liam Girdwood 05 Jul '11

05 Jul '11

The following changes since commit d21685ec258f803d3badae5eae821383a34815a9: Merge branch 'for-2.6.40' into for-2.6.41 (2011-05-30 10:54:18 +0800) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-3.1 Jarkko Nikula (1): ASoC: tlv320aic3x: Add correct hw registers to Line1 cross connect muxes Ricardo Neri (4): ASoC: OMAP: Add CPU DAI driver for HDMI ASoC: OMAP4: Add HDMI Audio machine driver for OMAP4 boards ASoC: OMAP: Update Makefile and Kconfig for HDMI audio ASoC: twl6040 - According to TWL6040 specification, gain start at 6dB and not -6dB. sound/soc/codecs/tlv320aic3x.c | 34 +++++--- sound/soc/codecs/twl6040.c | 4 +- sound/soc/omap/Kconfig | 11 +++ sound/soc/omap/Makefile | 4 + sound/soc/omap/omap-hdmi.c | 158 ++++++++++++++++++++++++++++++++++++++ sound/soc/omap/omap-hdmi.h | 36 +++++++++ sound/soc/omap/omap4-hdmi-card.c | 129 +++++++++++++++++++++++++++++++ 7 files changed, 361 insertions(+), 15 deletions(-) create mode 100644 sound/soc/omap/omap-hdmi.c create mode 100644 sound/soc/omap/omap-hdmi.h create mode 100644 sound/soc/omap/omap4-hdmi-card.c

2 1

[alsa-devel] [GIT PULL] ASoC fixes for 3.0
by Liam Girdwood 05 Jul '11

05 Jul '11

The following changes since commit 4a787a3ff3f419c23ab0a5cef677fa441356b818: Merge branch 'for-2.6.40' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into topic/asoc (2011-05-20 11:25:32 +0200) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6.git for-3.0 Jarkko Nikula (2): ASoC: tlv320aic3x: Don't sync first two registers from register cache ASoC: tlv320aic3x: Do soft reset to codec when going to bias off state Michael Williamson (1): audio: tlv320aic26: fix PLL register configuration sound/soc/codecs/tlv320aic26.c | 14 +++++++++++--- sound/soc/codecs/tlv320aic3x.c | 9 ++++++++- 2 files changed, 19 insertions(+), 4 deletions(-)

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[alsa-devel] [PATCH] ASoC: core - Add platform read and write.
by Liam Girdwood 05 Jul '11

05 Jul '11

In preparation for ASoC Dynamic PCM (AKA DSP) support. Allow platform driver to perform IO. Intended for platform DAPM. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 8 ++++++++ sound/soc/soc-core.c | 30 ++++++++++++++++++++++++++++++ 2 files changed, 38 insertions(+), 0 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6424b10..f30f3fe 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -312,6 +312,10 @@ int snd_soc_default_readable_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_default_writable_register(struct snd_soc_codec *codec, unsigned int reg); +int snd_soc_platform_read(struct snd_soc_platform *platform, + unsigned int reg); +int snd_soc_platform_write(struct snd_soc_platform *platform, + unsigned int reg, unsigned int val); /* Utility functions to get clock rates from various things */ int snd_soc_calc_frame_size(int sample_size, int channels, int tdm_slots); @@ -658,6 +662,10 @@ struct snd_soc_platform_driver { /* probe ordering - for components with runtime dependencies */ int probe_order; int remove_order; + + /* platform IO - used for platform DAPM */ + unsigned int (*read)(struct snd_soc_platform *, unsigned int); + int (*write)(struct snd_soc_platform *, unsigned int, unsigned int); }; struct snd_soc_platform { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index d49f0ed..d08abf4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1640,6 +1640,36 @@ int snd_soc_codec_writable_register(struct snd_soc_codec *codec, } EXPORT_SYMBOL_GPL(snd_soc_codec_writable_register); +int snd_soc_platform_read(struct snd_soc_platform *platform, + unsigned int reg) +{ + unsigned int ret; + + if (!platform->driver->read) { + dev_err(platform->dev, "platform has no read back\n"); + return -1; + } + + ret = platform->driver->read(platform, reg); + dev_dbg(platform->dev, "read %x => %x\n", reg, ret); + + return ret; +} +EXPORT_SYMBOL_GPL(snd_soc_platform_read); + +int snd_soc_platform_write(struct snd_soc_platform *platform, + unsigned int reg, unsigned int val) +{ + if (!platform->driver->write) { + dev_err(platform->dev, "platform has no write back\n"); + return -1; + } + + dev_dbg(platform->dev, "write %x = %x\n", reg, val); + return platform->driver->write(platform, reg, val); +} +EXPORT_SYMBOL_GPL(snd_soc_platform_write); + /** * snd_soc_new_ac97_codec - initailise AC97 device * @codec: audio codec -- 1.7.4.1

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Re: [alsa-devel] [PATCH] ASoc: mx27vis-aic32x4: Remove unneeded header
by Mark Brown 04 Jul '11

04 Jul '11

On Mon, Jul 04, 2011 at 04:01:24PM -0300, Fabio Estevam wrote: > There is no machine type check on this code, so remove the mach-types header. > > Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> This isn't really useful - if you're going to change something here you should change the code to check for the machine.

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[PATCH] ASoC: Manage WM8731 ACTIVE bit as a supply widget
by Mark Brown 04 Jul '11

04 Jul '11

Now we have supply widgets there's no need to open code the handling of the ACTIVE bit. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8731.c | 29 +++-------------------------- 1 files changed, 3 insertions(+), 26 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 2dc964b..59d9e59 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -175,6 +175,7 @@ static const struct snd_kcontrol_new wm8731_input_mux_controls = SOC_DAPM_ENUM("Input Select", wm8731_insel_enum); static const struct snd_soc_dapm_widget wm8731_dapm_widgets[] = { +SND_SOC_DAPM_SUPPLY("Active",WM8731_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OSC", WM8731_PWR, 5, 1, NULL, 0), SND_SOC_DAPM_MIXER("Output Mixer", WM8731_PWR, 4, 1, &wm8731_output_mixer_controls[0], @@ -204,6 +205,8 @@ static int wm8731_check_osc(struct snd_soc_dapm_widget *source, static const struct snd_soc_dapm_route wm8731_intercon[] = { {"DAC", NULL, "OSC", wm8731_check_osc}, {"ADC", NULL, "OSC", wm8731_check_osc}, + {"DAC", NULL, "ACTIVE"}, + {"ADC", NULL, "ACTIVE"}, /* output mixer */ {"Output Mixer", "Line Bypass Switch", "Line Input"}, @@ -315,29 +318,6 @@ static int wm8731_hw_params(struct snd_pcm_substream *substream, return 0; } -static int wm8731_pcm_prepare(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* set active */ - snd_soc_write(codec, WM8731_ACTIVE, 0x0001); - - return 0; -} - -static void wm8731_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct snd_soc_codec *codec = dai->codec; - - /* deactivate */ - if (!codec->active) { - udelay(50); - snd_soc_write(codec, WM8731_ACTIVE, 0x0); - } -} - static int wm8731_mute(struct snd_soc_dai *dai, int mute) { struct snd_soc_codec *codec = dai->codec; @@ -480,7 +460,6 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, snd_soc_write(codec, WM8731_PWR, reg | 0x0040); break; case SND_SOC_BIAS_OFF: - snd_soc_write(codec, WM8731_ACTIVE, 0x0); snd_soc_write(codec, WM8731_PWR, 0xffff); regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); @@ -496,9 +475,7 @@ static int wm8731_set_bias_level(struct snd_soc_codec *codec, SNDRV_PCM_FMTBIT_S24_LE) static struct snd_soc_dai_ops wm8731_dai_ops = { - .prepare = wm8731_pcm_prepare, .hw_params = wm8731_hw_params, - .shutdown = wm8731_shutdown, .digital_mute = wm8731_mute, .set_sysclk = wm8731_set_dai_sysclk, .set_fmt = wm8731_set_dai_fmt, -- 1.7.5.4

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[alsa-devel] [PATCH 0/3] ASoC: Tegra: Fixes and SPDIF support
by Stephen Warren 04 Jul '11

04 Jul '11

A couple of small bug-fixes for the existing Tegra SPDIF support, and a new SPDIF driver. I've tested the SPDIF driver on Harmony. However, I'm not submitting the ASoC machine support patch for inclusion in mainline, since it relies on the board files registering some new platform devices, and we can't upstream any board file changes right now. Still, this may be useful for other board vendors to start working with internally. For people interested in testing/using this, I can push the branch I used for testing this on Harmony publicly sometime next week, hopefully. That branch contains the board changes to support this, and the machine driver change. Stephen Warren (3): ASoC: Tegra: I2S: Ensure clock is enabled when writing regs ASoC: Tegra: I2S: s/clk_get_sys/clk_get/ ASoC: Tegra: Implement SPDIF CPU DAI sound/soc/tegra/Kconfig | 9 + sound/soc/tegra/Makefile | 2 + sound/soc/tegra/tegra_i2s.c | 10 +- sound/soc/tegra/tegra_spdif.c | 371 ++++++++++++++++++++++++++++++++ sound/soc/tegra/tegra_spdif.h | 473 +++++++++++++++++++++++++++++++++++++++++ 5 files changed, 862 insertions(+), 3 deletions(-) create mode 100644 sound/soc/tegra/tegra_spdif.c create mode 100644 sound/soc/tegra/tegra_spdif.h

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Samuel, On Monday 04 July 2011 13:48:44 Samuel Ortiz wrote: > I don't see the value of those 3 inline functions. Removing them would make > the code actually more understandable (especially for the 2nd one). Is it OK for you if I fix these as a follow-up series, so I do not need to rewrite this series, and resend it? -- Péter

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Samuel, On Monday 04 July 2011 13:48:44 Samuel Ortiz wrote: > Hi Peter, > > On Tue, Jun 21, 2011 at 04:39:06PM +0300, Peter Ujfalusi wrote: > > +int twl6040_is_powered(struct twl6040 *twl6040) > > +{ > > + return twl6040->power_count; > > +} > > +EXPORT_SYMBOL(twl6040_is_powered); > > Do we really need to export this one ? No, we don't. At the end we do not need this function at all either. I'm going to remove it. > > +static inline int twl6040_get_rev(struct twl6040 *twl6040) > > +{ > > + return twl6040->rev; > > +} No need for this either, I'll remove it. > > + > > +static inline int twl6040_request_irq(struct twl6040 *twl6040, int irq, > > + irq_handler_t handler, > > + unsigned long irqflags, > > + const char *name, > > + void *data) > > +{ > > + if (!twl6040->irq_base) > > + return -EINVAL; > > + > > + return request_threaded_irq(twl6040->irq_base + irq, NULL, handler, > > + irqflags, name, data); > > +} > > + > > +static inline void twl6040_free_irq(struct twl6040 *twl6040, int irq, > > + void *data) > > +{ > > + if (!twl6040->irq_base) > > + return; > > + > > + free_irq(twl6040->irq_base + irq, data); > > +} > > I don't see the value of those 3 inline functions. Removing them would make > the code actually more understandable (especially for the 2nd one). The twl6040_request_irq is also used by the ASoC codec driver, and the vibra driver, but if you think it is better to replace those as well, I will do it right away. -- Péter

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Re: [alsa-devel] [PATCH v6 08/18] mfd: twl6040: Add initial support
by Péter Ujfalusi 04 Jul '11

04 Jul '11

Hi Felipe, Samuel, On Monday 04 July 2011 14:53:30 Balbi, Felipe wrote: > > + ret = twl6040_request_irq(twl6040, TWL6040_IRQ_READY, > > + twl6040_naudint_handler, 0, > > + "twl6040_irq_ready", twl6040); > > why don't you use the normal request_threaded_irq() ?? This is a bit of > obfuscation IMO. I have left this 'macro' in since it is also used within the twl6040-vibra, and twl6040 ASoC codec driver to request the irq for them. But if you think it is better to replace the twl6040_request_irq with direct request_threaded_irq in those drivers as well, I'll do that. -- Péter

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[alsa-devel] [PATCH] ALSA: hda-intel - add position_fix quirk for Gateway NV53A
by Dmitry Torokhov 04 Jul '11

04 Jul '11

Without this option internal microphone does not seem to be working. $ lspci -vvnn | grep -A10 Audio 00:14.2 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] (rev 40) Subsystem: Acer Incorporated [ALI] Device [1025:036e] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin ? routed to IRQ 16 Region 0: Memory at d2300000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Kernel driver in use: HDA Intel -- 01:05.1 Audio device [0403]: ATI Technologies Inc RS880 Audio Device [Radeon HD 4200] [1002:970f] Subsystem: Acer Incorporated [ALI] Device [1025:036e] Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 19 Region 0: Memory at d2210000 (32-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 3 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME- Capabilities: [a0] MSI: Enable- Count=1/1 Maskable- 64bit+ Signed-off-by: Dmitry Torokhov <dtor(a)mail.ru> --- sound/pci/hda/hda_intel.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 486f6de..4fbcea6 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2349,6 +2349,7 @@ static int azx_dev_free(struct snd_device *device) static struct snd_pci_quirk position_fix_list[] __devinitdata = { SND_PCI_QUIRK(0x1025, 0x009f, "Acer Aspire 5110", POS_FIX_LPIB), SND_PCI_QUIRK(0x1025, 0x026f, "Acer Aspire 5538", POS_FIX_LPIB), + SND_PCI_QUIRK(0x1025, 0x036e, "Gateway NV53A", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01cc, "Dell D820", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01de, "Dell Precision 390", POS_FIX_LPIB), SND_PCI_QUIRK(0x1028, 0x01f6, "Dell Latitude 131L", POS_FIX_LPIB),

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Re: [alsa-devel] Noise on snd-hda-intel
by Raymond Yau 04 Jul '11

04 Jul '11

2011/6/24 Sune Mølgaard <sune(a)molgaard.org>: Refer to patch eb541337b7a43822fce7d0c9d967ee149b2d9a96 "ALSA: hda - Make converter setups sticky" The patch create an array of 8 elements for converter caches but there are more than 8 converters in ad1988/ad1989 5 analog Audio Output + 3 analog Audio Input + Digital Audio Output + DIgital Input + snd_array_init(&codec->cvt_setups, sizeof(struct hda_cvt_setup), 8); >> >> Is this a clean install of 11.4 or upgrade from previous version of >> ubuntu which use "default:2" and create "Digital Capture Volume" ? > > It is an upgrade. !!Modprobe options (Sound related) !!-------------------------------- snd-usb-audio: index=-2 snd_hda_intel: model=6stack enable=1 index=0 Seem those snd-usb-audio are loaded before snd-hda-intel and that's why your onboard become card 2 instead of card 0 > >> since pulseaudio use "front:2" for playback and capture and this won't >> create "digital capture volume" control > > How to remedy? Are you upgarded from version which use dmix/dsnoop ? or the "Digital Capture Volume" control is created when remove alsa-pulse-plugin There are three capture subdevices since ad1988/ad1989 have 3 [Audio Input] when application using dsnoop which use subdevice 0 by default However for PA server release the capture device after probe so it is possible that other application (e.g. arecord ) can open subdevice 0 PA server will open the next available subdevice https://qa.mandriva.com/show_bug.cgi?id=52175 card 2: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 3/3 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Simple mixer control 'Capture',0 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 54 Front Left: Capture 48 [89%] [13.50dB] [on] Front Right: Capture 48 [89%] [13.50dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 54 Front Left: Capture 0 [0%] [-58.50dB] [off] Front Right: Capture 0 [0%] [-58.50dB] [off] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch penum Capture channels: Front Left - Front Right Limits: Capture 0 - 54 Front Left: Capture 0 [0%] [-58.50dB] [off] Front Right: Capture 0 [0%] [-58.50dB] [off]

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[alsa-devel] [PATCH] ASoC: core - Make platform probe more like codec probe.
by Liam Girdwood 02 Jul '11

02 Jul '11

In preparation for ASoC dynamic PCM support (AKA ASoC DSP) Platform will also support DAPM so separate out the probe function to simplify the code (just like the codec probe). Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- sound/soc/soc-core.c | 51 +++++++++++++++++++++++++++++++++++-------------- 1 files changed, 36 insertions(+), 15 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 32bc503..d49f0ed 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -986,6 +986,39 @@ err_probe: return ret; } +static int soc_probe_platform(struct snd_soc_card *card, + struct snd_soc_platform *platform) +{ + int ret = 0; + const struct snd_soc_platform_driver *driver = platform->driver; + + platform->card = card; + + if (!try_module_get(platform->dev->driver->owner)) + return -ENODEV; + + if (driver->probe) { + ret = driver->probe(platform); + if (ret < 0) { + dev_err(platform->dev, + "asoc: failed to probe platform %s: %d\n", + platform->name, ret); + goto err_probe; + } + } + + /* mark platform as probed and add to card platform list */ + platform->probed = 1; + list_add(&platform->card_list, &card->platform_dev_list); + + return 0; + +err_probe: + module_put(platform->dev->driver->owner); + + return ret; +} + static void rtd_release(struct device *dev) {} static int soc_post_component_init(struct snd_soc_card *card, @@ -1109,21 +1142,9 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) /* probe the platform */ if (!platform->probed && platform->driver->probe_order == order) { - if (!try_module_get(platform->dev->driver->owner)) - return -ENODEV; - - if (platform->driver->probe) { - ret = platform->driver->probe(platform); - if (ret < 0) { - printk(KERN_ERR "asoc: failed to probe platform %s\n", - platform->name); - module_put(platform->dev->driver->owner); - return ret; - } - } - /* mark platform as probed and add to card platform list */ - platform->probed = 1; - list_add(&platform->card_list, &card->platform_dev_list); + ret = soc_probe_platform(card, platform); + if (ret < 0) + return ret; } /* probe the CODEC DAI */ -- 1.7.4.1

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[alsa-devel] Roland UM-ONE support
by xl600＠teamspode.org 02 Jul '11

02 Jul '11

I am in need of adding UM-ONE support. I believe this interface is simply a repackaged UM-1G, but ALSA 1.0.17 simply won't recognize it. Being new to this level of development, any clues as to how to add this interface support would be welcome. I'd first like to start off by assuming it will work as a UM-1G. If it doesn't, then I'll have to dig deeper. Thanks, Robert

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Re: [alsa-devel] Bug#631963: linux-image-3.0.0-rc4-686-pae: no sound out the rear plugs
by Ben Hutchings 02 Jul '11

02 Jul '11

On Fri, Jul 01, 2011 at 11:06:56PM +0200, Toni Mueller wrote: > > Hi Takashi, > > On Thu, 30.06.2011 at 07:59:19 +0200, Takashi Iwai <tiwai(a)suse.de> wrote: > > I guess this must be the new feature, "Auto-Mute Mode". > > it turns out that you're right - I just had the opportunity to verify > that. Can we consider this closed, then? Or should that feature be disabled by default? Ben. -- Ben Hutchings We get into the habit of living before acquiring the habit of thinking. - Albert Camus

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[alsa-devel] PCM DMA only can fire ISR when it receives 4 samples more than the period, from McBSP port?
by Philip Chu 01 Jul '11

01 Jul '11

HI, I got a problem with my audio capture using McBSP 3 port on OMAP 3530. Linux kernel 2.6.32. For the very first audio data capture, when I expect to receive only one exact period of data through McBSP, PCM DMA ISR never comes. When I check the DMA pointer by using snd_pcm_update_hw_ptr_pos(), I found it was just trash value. If I ask the data sender to send 4 samples more than the period defined, DMA ISR gets fired, the it points to the exact period position, and I assume the extra 4 bytes just get lost? My McBSP is configured as a slave which needs the sender to provide Frame Sync, clocks etc, and those signals are designed to just kick in at the same time. Do the clock signal need to be present earlier than the Frame Sync? Anybody knows what the issue could be? Thanks in advance. Philip Chu

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Re: [alsa-devel] [PATCH 2/5] ASoC: atmel-ssc: add phybase in device structure
by Mark Brown 01 Jul '11

01 Jul '11

On Wed, Jun 29, 2011 at 08:00:16PM +0200, Nicolas Ferre wrote: > struct ssc_device { > struct list_head list; > + resource_size_t phybase; The type here looks rather wrong?

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[alsa-devel] alsa-lib rate plugins - float samples
by Pavel Hofman 01 Jul '11

01 Jul '11

Hi, This is probably a question for Jaroslav Kysela. The better-quality rate plugins offer float resolution, but currently we are deliberately loosing information by down-converting to s16 in plugin_rate API (convert_s16). In rate_samplerate.c, the samples are up-converted to float before calling the libsamplerate API. Speex offers float API too, while we are using the int16 one. There are two ways to handle this: A. Implementing the generic "convert" API function for these plugins, handling the sample - float conversion in each plugin individually. B. Extending the pcm_rate API (pcm_rate.h) with a new optional function convert_float. Since the conversion from general sample format to float is not trivial (using the plugin_ops.h operations), I would be inclined to suggest the extension (well I have it hacked already :-) ). On the other hand, the conversion routine can be located in some common code shared by all the plugins involved and we could keep the existing API, using the generic convert function. Thanks a lot for suggestions. Best regards, Pavel.

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Re: [alsa-devel] [PATCH 1/5] ASoC: wm8731: rework power management
by Mark Brown 01 Jul '11

01 Jul '11

On Wed, Jun 29, 2011 at 08:00:15PM +0200, Nicolas Ferre wrote: Don't mix multiple changes into a single patch! There's no perceptible code overlap between these so I don't understand why you've merged them, it just makes review harder and the changelog less descriptive. > - preserve crystal oscillator across suspend/resume sequence: > enabled by default, it should be kept enabled on resume. This isn't what your code does... > - if codec is in active state: set the active bit at resume time. Your changelog doesn't explain what the actual problem is here... I can see it by studying the diff but the changelog makes me think the change is bad as the subsystem shuts everything down before it enters CODEC suspend... > snd_soc_write(codec, WM8731_ACTIVE, 0x0); > - snd_soc_write(codec, WM8731_PWR, 0xffff); > + /* standby: keep crystal oscillator enabled */ > + snd_soc_write(codec, WM8731_PWR, 0x00df); This doesn't keep the crystal oscillator enabled, this forces it on in suspend (and without looking at the datasheet it also changes way more than the one register bit I'd expect to be changed). If the system isn't using the oscillator then that's not good. I'd expect to see a change to using snd_soc_update_bits() based on your description, or more likely something more involved. > @@ -533,6 +534,8 @@ static int wm8731_suspend(struct snd_soc_codec *codec, pm_message_t state) > static int wm8731_resume(struct snd_soc_codec *codec) > { > wm8731_set_bias_level(codec, SND_SOC_BIAS_STANDBY); > + if (codec->active) > + snd_soc_write(codec, WM8731_ACTIVE, 0x0001); This should be handled as part of the bias level managemnent and cache sync, if the disable happens in bias level management so should the restore.

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[alsa-devel] [PATCH] ASoC: Improve error reporting in Speyside WM8962 driver
by Mark Brown 30 Jun '11

30 Jun '11

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/samsung/speyside_wm8962.c | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/speyside_wm8962.c b/sound/soc/samsung/speyside_wm8962.c index c0ba0bf..55233e8 100644 --- a/sound/soc/samsung/speyside_wm8962.c +++ b/sound/soc/samsung/speyside_wm8962.c @@ -30,14 +30,16 @@ static int speyside_wm8962_set_bias_level(struct snd_soc_card *card, WM8962_FLL_MCLK, 32768, 44100 * 256); if (ret < 0) - pr_err("Failed to start FLL\n"); + pr_err("Failed to start FLL: %d\n"); ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, 44100 * 256, SND_SOC_CLOCK_IN); - if (ret < 0) + if (ret < 0) { + pr_err("Failed to set SYSCLK: %d\n"); return ret; + } } break; @@ -59,13 +61,15 @@ static int speyside_wm8962_set_bias_level_post(struct snd_soc_card *card, case SND_SOC_BIAS_STANDBY: ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, 32768, SND_SOC_CLOCK_IN); - if (ret < 0) + if (ret < 0) { + pr_err("Failed to switch away from FLL: %d\n", ret); return ret; + } ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, 0, 0, 0); if (ret < 0) { - pr_err("Failed to stop FLL\n"); + pr_err("Failed to stop FLL: %d\n", ret); return ret; } break; -- 1.7.5.4

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[alsa-devel] Via Codecs: playback_multi_pcm_prep_0() might do something unneded in redirected mode - or something else could be wrong.
by alex dot baldacchino dot alsasub at gmail dot com 30 Jun '11

30 Jun '11

I've got a doubt about the following piece of code: ... if (mout->hp_nid && mout->hp_nid != nids[HDA_FRONT] && !spec->hp_independent_mode) /* headphone out will just decode front left/right (stereo) */ snd_hda_codec_setup_stream(codec, mout->hp_nid, stream_tag, 0, format); In redirected mode, if mout->hp_nid is not the same as the front dac, headphone jack (spec->autocfg.hp_pins[0]) is connected to it (get inputs from the front dac) one way or the other, so it would seem useless to make a different dac work on the same stream, given its output is being, let me say, 'discarded'. Moreover, via_independent_hp_put() always cleans that stream up for hp_nid in similar conditions: ... if (spec->multiout.hp_nid && spec->multiout.hp_nid != spec->multiout.dac_nids[HDA_FRONT]) snd_hda_codec_setup_stream(codec, spec->multiout.hp_nid, 0, 0, 0); This means that switching Independent HP mode on and off while playing something would prevent the dac corresponding to hp_nid from decoding that stream any further, whereas a simple printk tells me playback_multi_pcm_prep_0 is called just once at the beginning of the streaming. A simple test showed me that piece of code isn't needed in my case (VT2020) and I'd guess that's the same for most codecs, if not all (not yet tested commenting out that block, though): starting with Independent HP mode 'ON', that block isn't executed (!spec->hp_independent_mode is false), but switching it to 'OFF' while playing I get sound out of my headset connected to my case front audio panel; I can do it repeatedly and always works. Though not extremely frequent, I think such a scenario could happen in "real world use", for instance, I can think of someone turning off Independent HP mode while playing, then turning off a speaker system because of someone complaining for loud sound (or asking for silence), or just by mistake, because thinking redirected mode were active while it wasn't. Therefore, if there were any codecs requiring for stream being setup for hp_nid while in redirected mode, to work properly, that could be made clear in playback_multi_pcm_prep_0(), just to make the code a bit clearer, and, most important, via_independent_hp_put should be corrected to take that into account (and perhaps infos on current stream(s) should be recorded somewhere, to do that). Regards.

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[alsa-devel] Via Codecs: functions mute_aa_path() and is_aa_path_mute() (file patch_via.c) seem to support different codecs.
by alex dot baldacchino dot alsasub at gmail dot com 30 Jun '11

30 Jun '11

These functions seem to support different codecs through different switch statements: mute_aa_path: ... /* get nid of MW0 and start & end index */ switch (spec->codec_type) { case VT1708: nid_mixer = 0x17; start_idx = 2; end_idx = 4; break; case VT1709_10CH: case VT1709_6CH: nid_mixer = 0x18; start_idx = 2; end_idx = 4; break; case VT1708B_8CH: case VT1708B_4CH: case VT1708S: case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; break; case VT1718S: nid_mixer = 0x21; start_idx = 1; end_idx = 3; break; default: return; } is_aa_path_mute: ... /* get nid of MW0 and start & end index */ switch (spec->codec_type) { case VT1708B_8CH: case VT1708B_4CH: case VT1708S: case VT1716S: nid_mixer = 0x16; start_idx = 2; end_idx = 4; break; case VT1702: nid_mixer = 0x1a; start_idx = 1; end_idx = 3; break; case VT1718S: nid_mixer = 0x21; start_idx = 1; end_idx = 3; break; case VT2002P: case VT1812: case VT1802: nid_mixer = 0x21; start_idx = 0; end_idx = 2; break; default: return 0; } It would seem that AA-path can be muted but not inquired for mute state for codecs VT1708, VT1709_10CH and VT1709_6CH, vice-versa for codecs VT1702, VT2002P, VT1812, VT1802. If this is the real way such codecs work, I wonder if analog_input_switch_put() works as expected for codec VT1708, in which case, perhaps, it should be modified: static int analog_input_switch_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { int change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); struct hda_codec *codec = snd_kcontrol_chip(kcontrol); set_widgets_power_state(codec); analog_low_current_mode(snd_kcontrol_chip(kcontrol), -1); if (snd_hda_get_bool_hint(codec, "analog_loopback_hp_detect") == 1) { if (is_aa_path_mute(codec)) vt1708_start_hp_work(codec->spec); else vt1708_stop_hp_work(codec->spec); } return change; } Since is_aa_path_mute() always returns 0 for codec VT1708 (default case), and both vt1708_start_hp_work() and vt1708_stop_hp_work() only execute for such a codec (due to an if statement checking for codec type and returning from the function if that's not VT1708), vt1708_stop_hp_work() is the only one that can be executed, so perhaps the code could be clearer without the if statement checking is_aa_path_mute() result. Moreover: static void vt1708_stop_hp_work(struct via_spec *spec) { if (spec->codec_type != VT1708 || spec->autocfg.hp_pins[0] == 0) return; if (snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1 && !is_aa_path_mute(spec->codec)) return; snd_hda_codec_write(spec->codec, 0x1, 0, 0xf81, !spec->vt1708_jack_detectect); cancel_delayed_work_sync(&spec->vt1708_hp_work); } Since !is_aa_path_mute(spec->codec) is always true for VT1708, there is no real dependence on it in the if statement above, thus it should be removed. But this also means that, if snd_hda_get_bool_hint(spec->codec, "analog_loopback_hp_detect") == 1, then analog_input_switch_put() will call vt1708_stop_hp_work(), which in turn will return without performing any operation, so that it would seem useless, unless any reconfiguration might happen just in time between the first call to snd_hda_get_bool_hint in analog_input_switch_put() and the second call within vt1708_stop_hp_work(). Otherwise, if those switch statements should match against each other, they could be replaced with a call to a new inline function, so that a single switch could be moved within it and become easier to maintain. For instance: static inline int get_supported_aa_mute_info( const struct via_spec * const spec, hda_nid_t *mixer, int *first_idx, int *last_idx ) { /* get correct nid of mixer and start & end index */ switch (spec->codec_type) { case VT1708: *mixer = 0x17; *first_idx = 2; *last_idx = 4; return 0; case VT1709_10CH: case VT1709_6CH: *mixer = 0x18; *first_idx = 2; *last_idx = 4; return 0; case VT1708B_8CH: case VT1708B_4CH: case VT1708S: case VT1716S: *mixer = 0x16; *first_idx = 2; *last_idx = 4; return 0; case VT1702: *mixer = 0x1a; *first_idx = 1; *last_idx = 3; return 0; case VT1718S: *mixer = 0x21; *first_idx = 1; *last_idx = 3; return 0; case VT2002P: case VT1812: case VT1802: *mixer = 0x21; *first_idx = 0; *last_idx = 2; return 0; default: return -EINVAL; } } In mute_aa_path: ... if( get_supported_aa_mute_info( spec, &nid_mixer, &start_idx, &end_idx ) < 0 ) return; In is_aa_path_mute: ... if( get_supported_aa_mute_info( spec, &nid_mixer, &start_idx, &end_idx ) < 0 ) return 0; It happens that the list of codec types tested by is_aa_path_mute() is the same tested by function analog_low_current_mode(); however, there should be no dependence on that, because whatever is_aa_path_mute() returned for any other codec would be filtered out by the switch statement within analog_low_current_mode(). Regards.

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[alsa-devel] MOTU Midi Express XT driver
by Torsten Schenk 30 Jun '11

30 Jun '11

Hello all, I'm currently trying to write a driver for the MOTU Midi Express XT. The problem I have is that the device has same idVendor and idProduct as the MOTU FastLane Quad MIDI Interface (for which already a driver exists in snd-usbmidi). If I do an lsusb, I see that the field iProduct identifies the device as a Midi Express XT. But I found no way of matching to iProduct in the usb_device_id structure. Is there a possibility to do so? Otherwise I'm thinking of patching snd-usbmidi in a way that it differentiates between the devices there. Thanks, Torsten Here the lsusb output: Bus 004 Device 003: ID 07fd:0001 Mark of the Unicorn FastLane Quad MIDI Interface Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 1 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x07fd Mark of the Unicorn idProduct 0x0001 FastLane Quad MIDI Interface bcdDevice 1.32 iManufacturer 0 iProduct 119 MIDI Express XT iSerial 0 bNumConfigurations 1 ...

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[alsa-devel] Add combined clock support to Atmel SoC audio
by Michele Da Rold 30 Jun '11

30 Jun '11

I have found this post on the network: http://www.spinics.net/lists/alsa-devel/msg40008.html there is a work on this theme or not? thanks michele

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[alsa-devel] problems on AT91 with SSC not generating frame sync on RF0 pin
by Michele Da Rold 30 Jun '11

30 Jun '11

I have the same problem posted on this mailing list on july 2009 problems on AT91 with SSC not generating frame sync on RF0 pin (http://www.spinics.net/lists/alsa-devel/msg25692.html) someone have a suggestion for the problem? thank you Michele

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Re: [alsa-devel] SB16 build error.
by Ralf Baechle 30 Jun '11

30 Jun '11

On Thu, Jun 30, 2011 at 01:28:03PM +0200, Takashi Iwai wrote: > > > I have no idea how big the soundblaster microcode being loaded actually is, > > > that is if the reduced size of 0x1f00 will be sufficient. > > > > The files found in /lib/firmware/sb16 are all under 2kB, thus likely > > sufficient. > > Too shortly answered. It turned out that some CSP codes (like Qsound) > can be above that size, it's almost 12kB. So the size in the original > code is really the necessary requirement, and the patch breaks for > such a case. > > An ugly workaround would be to fake the ioctl size. But this is > certainly to be avoided, since it has been broken on the specific > platforms for ages, thus breaking for them would be mostly harmless, > too. > > > > Aside of that I > > > don't see a problem - I don't see how the old ioctl can possibly have been > > > used before so there isn't a compatibility problem. > > > > > > Or you could entirely sidestep the problem and use request_firmware() but > > > I guess that's more effort than you want to invest. > > > > Yeah, that's another option I thought of. But it's too intrusive for > > 3.0-rc6, so I'd like waive it for 3.1. > > Actually the request_firmware() was implemented for some auto-loadable > CSP codes. Others need the manual loading, so it is via ioctl. It > can be converted, but I don't think it makes sense for such old > stuff. After all, it still works with x86-ISA as is. In userland an empty definition will be used for _IOC_TYPECHECK so there won't be an error. So userland already is already using the existing value for SNDRV_SB_CSP_IOCTL_LOAD_CODE ... With a crude hack like #define SNDRV_SB_CSP_IOCTL_LOAD_CODE \ _IOC(_IOC_WRITE,'H', 0x11, sizeof(struct snd_sb_csp_microcode)) error checking can be bypassed and all will be fine as long as the resulting value doesn't result in in a a duplicate case value - which it doesn't, at least not in my testing. Should work but isn't nice. Ralf

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Re: [alsa-devel] SB16 build error.
by Ralf Baechle 30 Jun '11

30 Jun '11

On Thu, Jun 30, 2011 at 12:15:56PM +0200, Takashi Iwai wrote: > At Thu, 30 Jun 2011 10:17:54 +0100, > Ralf Baechle wrote: > > > > Found on a MIPS build but certain other architectures will have the same > > issue: > > > > CC sound/isa/sb/sb16_csp.o > > sound/isa/sb/sb16_csp.c: In function ‘snd_sb_csp_ioctl’: > > sound/isa/sb/sb16_csp.c:228: error: case label does not reduce to an integer constant > > make[1]: *** [sound/isa/sb/sb16_csp.o] Error 1 > > make: *** [sound/isa/sb/sb16_csp.o] Error 2 > > > > This error message is caused by the _IOC_TYPECHECK() error check triggering > > due to excessive ioctl size on Alpha, PowerPC, MIPS and SPARC which define > > _IOC_SIZEBITS as 13. On all other architectures define it as 14 so struct > > snd_sb_csp_microcode with it's little over 12kB will just about fit into > > the 16kB limit. > > What about the patch below? I have no idea how big the soundblaster microcode being loaded actually is, that is if the reduced size of 0x1f00 will be sufficient. Aside of that I don't see a problem - I don't see how the old ioctl can possibly have been used before so there isn't a compatibility problem. Or you could entirely sidestep the problem and use request_firmware() but I guess that's more effort than you want to invest. > This is an old ISA driver, so the impact must be very low. True. I notice that sort of stuff in automated mass builds - not necessarily the sort of kernels one would actually use. Still build errors are annoying :) Ralf

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Re: [alsa-devel] [PATCH v6 00/18] MFD/ASoC/Input: TWL4030/TWL60X0 changes
by Péter Ujfalusi 30 Jun '11

30 Jun '11

Hi Samuel, On Thursday 30 June 2011 10:04:43 Samuel Ortiz wrote: > I will do that very soon, no worries. Thank you very much, I'm going to have another series on top of this one coming, and want to make sure that I'm in the right track. Thanks, Péter

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[alsa-devel] SB16 build error.
by Ralf Baechle 30 Jun '11

30 Jun '11

Found on a MIPS build but certain other architectures will have the same issue: CC sound/isa/sb/sb16_csp.o sound/isa/sb/sb16_csp.c: In function ‘snd_sb_csp_ioctl’: sound/isa/sb/sb16_csp.c:228: error: case label does not reduce to an integer constant make[1]: *** [sound/isa/sb/sb16_csp.o] Error 1 make: *** [sound/isa/sb/sb16_csp.o] Error 2 This error message is caused by the _IOC_TYPECHECK() error check triggering due to excessive ioctl size on Alpha, PowerPC, MIPS and SPARC which define _IOC_SIZEBITS as 13. On all other architectures define it as 14 so struct snd_sb_csp_microcode with it's little over 12kB will just about fit into the 16kB limit. Ralf

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[alsa-devel] model needed for Clevo P150HM (Realtek ALC892)
by Julian Sikorski 30 Jun '11

30 Jun '11

Hi, I am coming here from pulseaudio-general mailing list [1]. I recently got myself a Clevo P150HM laptop. It has 2.1 speakers, a built-in microphone and the following connectors: - headphone out - microphone in; can be switched to center/lfe out - spdif out; can be switched to rear speaker out - line in; can be switched to side speaker out Alsa does not seem to reflect that the device is capable of 5.1 and 7.1 sound, and there are two configs showing up in pavucontrol (analog output and analog headphones) which seem to be the same. I have uploaded the output of alsa-info.sh [2]. Please let me know if more information is needed. Regards, Julian [1] http://thread.gmane.org/gmane.comp.audio.pulseaudio.general/10096 [2] http://www.alsa-project.org/db/?f=7c91e874f4272387c2c51e448337567202c0edfe

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[alsa-devel] [PATCH v6 00/18] MFD/ASoC/Input: TWL4030/TWL60X0 changes
by Peter Ujfalusi 30 Jun '11

30 Jun '11

Hello, Changes since v5: - Use alloc_workqueue in the twl6040-vibra driver (comment from Tejun Heo) - Allow user to change the headset power mode, but if the change is not allowed by the use case the change is not written to the HW (comment from Mark Brown) Mark: I have added your acks for the patches, except for patch 15, where you requested a change. Intro message from the v5 series: The series will do five major things, and they are in one series, because they pretty much depending on each other, so it is easier to handle them together. 1. Move the common pmic related configurations out from the board files. New twl-common.c/h file has been created for this purpose. OMAP3 board files received two patches (pdata, and regualtor changes), OMAP4 board files has been changed with one patch. 2. Rename the twl4030-codec MFD driver to twl4030-audio Since the ASoC multicomponent introduction (2.6.37) the twl4030-codec MFD dirver had been using twl4030-audio to register (twl4030-codec is the ASoC codec driver). The documentation refers to this part of twl as audio block, so it is better to rename the driver as well to avoid confusion. 3. Introduction of MFD driver for twl6040. The TWL6040 audio IC has codec and vibra functionality. Convert the existing ASoC codec driver to use the new MFD driver's interface. 4. Vibrator driver for TWL6040 (Input/ForceFeedback driver) The driver itself, and support for the vibrators in SDP4430 board. 5. Updates for twl6040 ASoC codec driver HS ramp configuration, new ABE dais, updated sample rate constraints, SDP4430 configuration. Regards, Peter --- Axel Castaneda Gonzalez (1): ASoC: twl6040: Configure ramp step based on platform Liam Girdwood (3): ASoC: twl6040: add all ABE DAIs ASoC: twl6040: Support other sample rates in constraints. ASoC: twl6040: set default constraints. Misael Lopez Cruz (5): mfd: twl6040: Add initial support ASoC: twl6040: Convert into TWL6040 MFD child input: Add initial support for TWL6040 vibrator ASoC: twl6040: Remove pll and headset mode dependency OMAP4: SDP4430: Add twl6040 codec platform support Peter Ujfalusi (9): OMAP: New twl-common for common TWL configuration OMAP4: Move common twl6030 configuration to twl-common OMAP3: Move common twl configuration to twl-common OMAP3: Move common regulator configuration to twl-common MFD: twl4030-codec: Rename internals from codec to audio MFD: twl4030-codec -> twl4030-audio: Rename the driver MFD: twl4030-audio: Rename platform data MFD: twl6040: Change platform data for soc codec driver OMAP4: SDP4430: Add twl6040 vibrator platform support arch/arm/mach-omap2/Makefile | 2 +- arch/arm/mach-omap2/board-3430sdp.c | 89 +--- arch/arm/mach-omap2/board-4430sdp.c | 187 ++---- arch/arm/mach-omap2/board-cm-t35.c | 48 +-- arch/arm/mach-omap2/board-devkit8000.c | 37 +-- arch/arm/mach-omap2/board-igep0020.c | 44 +-- arch/arm/mach-omap2/board-ldp.c | 15 +- arch/arm/mach-omap2/board-omap3beagle.c | 60 +-- arch/arm/mach-omap2/board-omap3evm.c | 67 +-- arch/arm/mach-omap2/board-omap3pandora.c | 59 +-- arch/arm/mach-omap2/board-omap3stalker.c | 70 +-- arch/arm/mach-omap2/board-omap3touchbook.c | 57 +-- arch/arm/mach-omap2/board-omap4panda.c | 149 +----- arch/arm/mach-omap2/board-overo.c | 59 +-- arch/arm/mach-omap2/board-rm680.c | 8 +- arch/arm/mach-omap2/board-rx51-peripherals.c | 44 +-- arch/arm/mach-omap2/board-zoom-peripherals.c | 88 +--- arch/arm/mach-omap2/common-board-devices.c | 21 - arch/arm/mach-omap2/common-board-devices.h | 26 +- arch/arm/mach-omap2/twl-common.c | 304 ++++++++++ arch/arm/mach-omap2/twl-common.h | 59 ++ arch/arm/plat-omap/include/plat/irqs.h | 12 +- drivers/input/misc/Kconfig | 13 +- drivers/input/misc/Makefile | 1 + drivers/input/misc/twl4030-vibra.c | 12 +- drivers/input/misc/twl6040-vibra.c | 416 +++++++++++++ drivers/mfd/Kconfig | 8 +- drivers/mfd/Makefile | 3 +- drivers/mfd/twl-core.c | 13 +- drivers/mfd/twl4030-audio.c | 277 +++++++++ drivers/mfd/twl4030-codec.c | 277 --------- drivers/mfd/twl6040-core.c | 601 +++++++++++++++++++ drivers/mfd/twl6040-irq.c | 205 +++++++ include/linux/i2c/twl.h | 25 +- .../linux/mfd/{twl4030-codec.h => twl4030-audio.h} | 16 +- include/linux/mfd/twl6040.h | 261 ++++++++ sound/soc/codecs/Kconfig | 3 +- sound/soc/codecs/twl4030.c | 22 +- sound/soc/codecs/twl6040.c | 628 ++++++++------------ sound/soc/codecs/twl6040.h | 118 ---- sound/soc/omap/sdp3430.c | 2 +- sound/soc/omap/sdp4430.c | 2 + sound/soc/omap/zoom2.c | 2 +- 43 files changed, 2635 insertions(+), 1775 deletions(-) create mode 100644 arch/arm/mach-omap2/twl-common.c create mode 100644 arch/arm/mach-omap2/twl-common.h create mode 100644 drivers/input/misc/twl6040-vibra.c create mode 100644 drivers/mfd/twl4030-audio.c delete mode 100644 drivers/mfd/twl4030-codec.c create mode 100644 drivers/mfd/twl6040-core.c create mode 100644 drivers/mfd/twl6040-irq.c rename include/linux/mfd/{twl4030-codec.h => twl4030-audio.h} (96%) create mode 100644 include/linux/mfd/twl6040.h -- 1.7.5.3

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Re: [alsa-devel] Bug#631963: linux-image-3.0.0-rc4-686-pae: no sound out the rear plugs
by Takashi Iwai 30 Jun '11

30 Jun '11

At Wed, 29 Jun 2011 20:45:02 +0200, Toni Mueller wrote: > > Hi, > > On Wed, 29.06.2011 at 07:21:33 +0200, Takashi Iwai <tiwai(a)suse.de> wrote: > > I'd definitely need alsa-info.sh outputs on both working and > > non-working kernels, i.e. on 2.6.38 and 3.0-rc4. > > this first post is for the non-working kernel. The other kernel's > information will follow tomorrow. I guess this must be the new feature, "Auto-Mute Mode". When this is enabled, the line-outs are muted automatically when the headphone jack is plugged. In your case, turn this mixer enum off. Takashi

1 0

[alsa-devel] SBx00 Azalia ALC663 Working but MIC goes away
by Diego Jacobi 30 Jun '11

30 Jun '11

I already sended this mail to this list ... many times .. but it gets bounced or wait for moderation. I find this annoying. I am attaching the files compressed now. I tested the solution mentioned by Takashi. I have installed alsa-driver-kmp-default and restarted, but i found the same behavior so, i didnt make more alsa-info files. However, i dont know what is "the right repository", i choosed "multimedia:audio:KMP/openSUSE_11.4" and also tested "multimedia:audio:KMP/openSUSE_11.4_Update" which provided the same file. The file is installed in linux-i1zx:/home/diego # modinfo snd-hda-intel | grep filename filename: /lib/modules/2.6.37.6-0.5-default/kernel/sound/pci/hda/snd-hda-intel.ko I have also tested model=m51va which also didnt helped. I dont think that the model, or model-autodetection is the solution for this. I mean, the problems gets solved after restarting alsa. So alsa does handles the MIC. I noted that in /etc/modprobe.d i have 2 sound.conf files. One is 50-sound.conf and the other is 50-sound.conf.YaST2save I dont know what is the second one for. If i use yast to edit the configuration, which i did, it edits the 50-sound.conf And also, all other files are numbered 50. What would happen if i change the sound to 51 to make the sound load after everything else? Could it break something ? like Yast ? ---------- Forwarded message ---------- From: Diego Jacobi <jacobidiego(a)gmail.com> Date: 2011/6/26 Subject: SBx00 Azalia ALC663 Working but MIC goes away To: alsa-devel(a)alsa-project.org Cc: Takashi Iwai <tiwai(a)suse.de> Hi people. I am having this problem for a long time ago, and i couldn't find the same on google, or a solution for it. I also don't know the right way to write the question to google, because almost all are about problems that i don't have. OpenSUSE 11.4 installed about 2 Months ago. ASUS laptop n51tp Athlon X2 ATI Mobility™ Radeon® HD 4650 Audio System: Built-in Azalia compliant audio chip, with 3D effect & full duplex Built-in speaker and microphone Support Audio Jack Detect for Vista Basic Logo Support Audio CODEC criteria for Vista Premium Logo AUDIO WORKS but with a problem listed below. ABSTRACT: - MIC never works after booting. It needs an alsa restart and gets broke sometimes.. DETAILED: If i start the laptop with no external headsets or speaker connected in front panel. Audio goes through internal speakers. When i connect an external speaker or headsets (like to talk with skype), they don't work. And MIC doesn't works. This is undesired behavior and the windows driver doesn't work that way. Internal goes mute when i connect an external. Causes trouble when speaking with skype, as everyone can heard. I have to reset alsa with "rcalsasound restart" with headsets plugged in. IF i start the laptop with the headsets plugged in. The internal speakers will not work at all. Always in mute. But the external will work. Also the MIC wont work. I have to reset alsa with "rcalsasound restart" with headsets plugged in. If i do some work with the laptop using different multimedia sources, it may broke. Most of the time i get called by skype and the MIC is surprise surprise, not working. Then i have to close all programs, restart alsa and open skype again to be able to talk. Thats the main reason to post here. TESTS RESULTS: I have runned alsa-info.sh with a different set of tests. So i write the cases for each one, and the filename of the attached. ---------------------------- Boot without anything connected to Front panel. Open Banshee and play. Internal speakers works. Plug headsets. They don't work. ===> alsa-info.txt.noheadsets Open Skype & do test call. MIC doesn't work. "rcalsasound restart" while headsets are plugged Internal AND headsets are working fine. Open Skype & do test call. MIC works. Second call works best. ===> alsa-info.txt.workingright ---------------------------- Boot with headsets plugged. Open Banshee and play. Headsets works. Unplug headsets. Internal Speakers doesn't work. Open Skype & do test call. MIC doesn't works. ===> alsa-info.txt.headsetsnomic "rcalsasound restart" while headsets are plugged Restart Skype & do test call. First call, noise recorded. Second call, MIC works. Unplug headsets. Internal Speakers works. ===> alsa-info.txt.workingright2 ---------------------------- With the "working right" condition. Do normal tasks. Open emesene & Skype. Open Opera. (always leaved opened) Open some pages with Firefox (hotmail & other flashintensiv sites) Went to take a shower & Came back Internal speakers don't work anymore. Headsets works MIC doesn't works. ===> alsa-info.txt.onlyheadsetsnomic rcalsasound restart Internal speakers works. Headsets works Skype requires to be restarted, banshee doesn't. First call, noise recorded. Second call, MIC works. ===> alsa-info.txt.workingright3 In the gnome's Audio-preferences dialog, the input tab, the level-bars : - show as ghosts when it doesn't work. - shows green changing, when it does works. - shows as ghosts if no program is using the mic and changes to green if i start the recorder. Restarting alsa requires closing Opera, Firefox, Skype, Emesene, Banshee, etc. to work or some of them will broke the MIC again. It also requires root privileges. A broken headset also causes this behavior when i plug it, and then MIC stops working until a reset. By a broken headset i mean a very low quality crappy MIC Noganet, which worked fine about 1 week, and then it started to only function by placing the cable on special positions and not moving it. Don't buy Noganet! MY GUESS about the problems: as almost electronic engineer. The MIC may seem an initialization problem, as it never happened on windows, but i don't have it anymore installed, since long now. The driver requires to detect check for high impedance at the front panel and mute the internal ones, and/or provide a way to mute them separately. I wonder why i cant see my headsets as a separate entry of the mixers. A short-circuit problem may be happening in my front panel, and under some circumstances, like moving the cables, it causes some chipset failure over the MIC controller. The driver is not handling this error and it stays crashed until a reset. However, it doesn't explains why seeing a video in flashplayer in opera may cause this too. I will appreciate any help, and thanks to everyone reading this not-short post. Cheers. Diego Below: Data about my system. ============================================================ ============================================================ ============================================================ Only "Master" and "Capture" are shown on alsa-mixer. Both full. But the problem here is not the volume. diego@linux-i1zx:~> skype --version Skype 2.2.0.25 Copyright (c) 2004-2011, Skype Limited linux-i1zx:/home/diego # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.23. linux-i1zx:/home/diego # cat /proc/asound/cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xfacf4000 irq 16 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfadec000 irq 42 linux-i1zx:/home/diego # cat /proc/asound/devices 2: : timer 3: [ 0- 1]: digital audio playback 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: [ 1- 3]: digital audio playback 9: [ 1- 0]: hardware dependent 10: [ 1] : control 11: : sequencer linux-i1zx:/home/diego # cat /proc/asound/oss/devices 0: [0- 0]: mixer 3: [0- 0]: digital audio 4: [0- 0]: digital audio 12: [0- 1]: digital audio 16: [1- 0]: mixer linux-i1zx:/home/diego # cat /proc/asound/timers G0: system timer : 4000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P0-1-0: PCM playback 0-1-0 : SLAVE P1-3-0: PCM playback 1-3-0 : SLAVE linux-i1zx:/home/diego # cat /proc/asound/pcm 00-00: ALC663 Analog : ALC663 Analog : playback 1 : capture 1 00-01: ALC663 Digital : ALC663 Digital : playback 1 01-03: HDMI 0 : HDMI 0 : playback 1 linux-i1zx:/home/diego # cat /etc/modprobe.d/50-sound.conf options snd slots=snd-hda-intel,snd-hda-intel # 5Dex.PIu7BLVZl12:SBx00 Azalia (Intel HDA) alias snd-card-0 snd-hda-intel # NXNs.yqXoVEq89wB:R700 Audio Device [Radeon HD 4000 Series] alias snd-card-1 snd-hda-intel linux-i1zx:/home/diego # lspci 00:00.0 Host bridge: Advanced Micro Devices [AMD] RS780 Host Bridge 00:02.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (ext gfx port 0) 00:05.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 1) 00:06.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 2) 00:07.0 PCI bridge: Advanced Micro Devices [AMD] RS780 PCI to PCI bridge (PCIE port 3) 00:11.0 SATA controller: ATI Technologies Inc SB700/SB800 SATA Controller [AHCI mode] 00:12.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:12.1 USB Controller: ATI Technologies Inc SB700 USB OHCI1 Controller 00:12.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:13.0 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI0 Controller 00:13.1 USB Controller: ATI Technologies Inc SB700 USB OHCI1 Controller 00:13.2 USB Controller: ATI Technologies Inc SB700/SB800 USB EHCI Controller 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 3a) 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) 00:14.3 ISA bridge: ATI Technologies Inc SB700/SB800 LPC host controller 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:14.5 USB Controller: ATI Technologies Inc SB700/SB800 USB OHCI2 Controller 00:18.0 Host bridge: Advanced Micro Devices [AMD] Family 11h Processor HyperTransport Configuration (rev 40) 00:18.1 Host bridge: Advanced Micro Devices [AMD] Family 11h Processor Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] Family 11h Processor DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] Family 11h Processor Miscellaneous Control 00:18.4 Host bridge: Advanced Micro Devices [AMD] Family 11h Processor Link Control 01:00.0 VGA compatible controller: ATI Technologies Inc M96 [Mobility Radeon HD 4650] 01:00.1 Audio device: ATI Technologies Inc RV710/730 02:00.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL8111/8168B PCI Express Gigabit Ethernet controller (rev 02) 03:00.0 Network controller: Atheros Communications Inc. AR928X Wireless Network Adapter (PCI-Express) (rev 01) 06:01.0 FireWire (IEEE 1394): Ricoh Co Ltd R5C832 IEEE 1394 Controller (rev 05) 06:01.1 SD Host controller: Ricoh Co Ltd R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter (rev 22) 06:01.2 System peripheral: Ricoh Co Ltd R5C592 Memory Stick Bus Host Adapter (rev 12) 06:01.3 System peripheral: Ricoh Co Ltd xD-Picture Card Controller (rev 12) linux-i1zx:/home/diego # dmesg | grep sound [ 9.759600] input: HDA ATI SB Mic as /devices/pci0000:00/0000:00:14.2/sound/card0/input11 [23782.924422] input: HDA ATI SB Mic as /devices/pci0000:00/0000:00:14.2/sound/card0/input13 [23897.439740] input: HDA ATI SB Mic as /devices/pci0000:00/0000:00:14.2/sound/card0/input14 [23924.488713] input: HDA ATI SB Mic as /devices/pci0000:00/0000:00:14.2/sound/card0/input15 [23999.613391] input: HDA ATI SB Mic as /devices/pci0000:00/0000:00:14.2/sound/card0/input16 [25937.965371] input: HDA ATI SB Mic as /devices/pci0000:00/0000:00:14.2/sound/card0/input17 linux-i1zx:/home/diego # dmesg | grep hda [ 50.625207] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [23786.414301] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [23900.659802] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [23928.322973] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [24003.419618] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. [25940.535502] hda-intel: IRQ timing workaround is activated for card #1. Suggest a bigger bdl_pos_adj. linux-i1zx:/home/diego # lsmod | grep snd snd_pcm_oss 47887 0 snd_mixer_oss 16849 1 snd_pcm_oss snd_seq 57361 0 snd_seq_device 6646 1 snd_seq snd_hda_codec_hdmi 24875 1 snd_hda_codec_realtek 274207 1 snd_hda_intel 24912 2 snd_hda_codec 95606 3 snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel snd_hwdep 6038 1 snd_hda_codec snd_pcm 87993 4 snd_pcm_oss,snd_hda_codec_hdmi,snd_hda_intel,snd_hda_codec snd_timer 21844 2 snd_seq,snd_pcm snd 66158 15 snd_pcm_oss,snd_mixer_oss,snd_seq,snd_seq_device,snd_hda_codec_hdmi,snd_hda_codec_realtek,snd_hda_intel,snd_hda_codec,snd_hwdep,snd_pcm,snd_timer soundcore 6854 1 snd snd_page_alloc 7897 2 snd_hda_intel,snd_pcm

2 2

[alsa-devel] [PATCH 4/5] ASoC: atmel_ssc_dai/atmel-pcm: adapt to dmaengine usage
by Nicolas Ferre 30 Jun '11

30 Jun '11

Change xfer_size name to apply to PDC and DMA aswell. Specify overrun bit in interrupt mask. Add dmaengine specific routines and replace PDC ones in pcm_ops if appropriate. Uses cyclic DMA API for queuing samples. Signed-off-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> Signed-off-by: Uwe Kleine-König <u.kleine-koenig(a)pengutronix.de> --- sound/soc/atmel/atmel-pcm.c | 382 +++++++++++++++++++++++++++++++++++--- sound/soc/atmel/atmel-pcm.h | 4 +- sound/soc/atmel/atmel_ssc_dai.c | 33 ++-- 3 files changed, 373 insertions(+), 46 deletions(-) diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 51dde4e..fee6696 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -37,6 +37,7 @@ #include <linux/slab.h> #include <linux/dma-mapping.h> #include <linux/atmel_pdc.h> +#include <linux/dmaengine.h> #include <linux/atmel-ssc.h> #include <sound/core.h> @@ -44,6 +45,8 @@ #include <sound/pcm_params.h> #include <sound/soc.h> +#include <mach/at_hdmac.h> + #include "atmel-pcm.h" @@ -53,7 +56,7 @@ /* TODO: These values were taken from the AT91 platform driver, check * them against real values for AT32 */ -static const struct snd_pcm_hardware atmel_pcm_hardware = { +static const struct snd_pcm_hardware atmel_pcm_pdc_hardware = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | @@ -66,6 +69,22 @@ static const struct snd_pcm_hardware atmel_pcm_hardware = { .buffer_bytes_max = 32 * 1024, }; +static const struct snd_pcm_hardware atmel_pcm_dma_hardware = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_PAUSE, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .period_bytes_min = 256, /* not too low to absorb DMA programming overhead */ + .period_bytes_max = 2 * 0xffff, /* if 2 bytes format */ + .periods_min = 8, + .periods_max = 1024, /* no limit */ + .buffer_bytes_max = 64 * 1024, /* 64KiB */ +}; + +static const struct snd_pcm_hardware *atmel_pcm_hardware; + /*--------------------------------------------------------------------------*\ * Data types @@ -77,12 +96,19 @@ struct atmel_runtime_data { size_t period_size; dma_addr_t period_ptr; /* physical address of next period */ + int periods; /* period index of period_ptr */ /* PDC register save */ u32 pdc_xpr_save; u32 pdc_xcr_save; u32 pdc_xnpr_save; u32 pdc_xncr_save; + + /* dmaengine data */ + struct at_dma_slave atslave; + struct dma_async_tx_descriptor *desc; + dma_cookie_t cookie; + struct dma_chan *dma_chan; }; @@ -94,7 +120,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, { struct snd_pcm_substream *substream = pcm->streams[stream].substream; struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = atmel_pcm_hardware.buffer_bytes_max; + size_t size = atmel_pcm_hardware->buffer_bytes_max; buf->dev.type = SNDRV_DMA_TYPE_DEV; buf->dev.dev = pcm->card->dev; @@ -116,7 +142,7 @@ static int atmel_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, /*--------------------------------------------------------------------------*\ * ISR \*--------------------------------------------------------------------------*/ -static void atmel_pcm_dma_irq(u32 ssc_sr, +static void atmel_pcm_pdc_irq(u32 ssc_sr, struct snd_pcm_substream *substream) { struct atmel_runtime_data *prtd = substream->runtime->private_data; @@ -142,7 +168,7 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->period_ptr); ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); + prtd->period_size / params->data_xfer_size); ssc_writex(params->ssc->regs, ATMEL_PDC_PTCR, params->mask->pdc_enable); } @@ -156,12 +182,163 @@ static void atmel_pcm_dma_irq(u32 ssc_sr, ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->period_ptr); ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); + prtd->period_size / params->data_xfer_size); } snd_pcm_period_elapsed(substream); } +/** + * atmel_pcm_dma_irq: SSC interrupt handler for DMAENGINE enabled SSC + * + * We use DMAENGINE to send/receive data to/from SSC so this ISR is only to + * check if any overrun occured. + */ +static void atmel_pcm_dma_irq(u32 ssc_sr, + struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + if (ssc_sr & params->mask->ssc_error) { + if (snd_pcm_running(substream)) + pr_warning("atmel-pcm: buffer %s on %s" + " (SSC_SR=%#x)\n", + substream->stream == SNDRV_PCM_STREAM_PLAYBACK + ? "underrun" : "overrun", + params->name, ssc_sr); + + /* stop RX and capture stream: will be enabled again at restart */ + ssc_writex(params->ssc->regs, SSC_CR, params->mask->ssc_disable); + snd_pcm_stop(substream, SNDRV_PCM_STATE_XRUN); + + /* now drain RHR and read status to remove xrun condition */ + ssc_readx(params->ssc->regs, SSC_RHR); + ssc_readx(params->ssc->regs, SSC_SR); + } +} + +/*--------------------------------------------------------------------------*\ + * DMAENGINE operations +\*--------------------------------------------------------------------------*/ +static bool filter(struct dma_chan *chan, void *slave) +{ + struct at_dma_slave *sl = slave; + + if (sl->dma_dev == chan->device->dev) { + chan->private = sl; + return true; + } else { + return false; + } +} + +static int atmel_pcm_dma_alloc(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct ssc_device *ssc = prtd->params->ssc; + struct at_dma_slave *sdata = NULL; + + if (ssc->pdev) + sdata = ssc->pdev->dev.platform_data; + + if (sdata && sdata->dma_dev) { + dma_cap_mask_t mask; + + /* setup DMA addresses */ + sdata->rx_reg = (dma_addr_t)ssc->phybase + SSC_RHR; + sdata->tx_reg = (dma_addr_t)ssc->phybase + SSC_THR; + + /* Try to grab a DMA channel */ + dma_cap_zero(mask); + dma_cap_set(DMA_CYCLIC, mask); + prtd->dma_chan = dma_request_channel(mask, filter, sdata); + } + if (!prtd->dma_chan) { + pr_err("atmel-pcm: " + "DMA channel not available, unable to use SSC-audio\n"); + return -EBUSY; + } + + return 0; +} + +static void audio_dma_irq(void *data) +{ + struct snd_pcm_substream *substream = (struct snd_pcm_substream *)data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + + prtd->period_ptr += prtd->period_size; + if (prtd->period_ptr >= prtd->dma_buffer_end) + prtd->period_ptr = prtd->dma_buffer; + + snd_pcm_period_elapsed(substream); +} + +static void atmel_pcm_dma_slave_config(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct dma_chan *chan = prtd->dma_chan; + struct dma_slave_config slave_config; + enum dma_slave_buswidth buswidth; + + switch (prtd->params->data_xfer_size) { + case 1: + buswidth = DMA_SLAVE_BUSWIDTH_1_BYTE; + break; + case 2: + buswidth = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 4: + buswidth = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + return; + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + slave_config.direction = DMA_TO_DEVICE; + slave_config.dst_addr_width = buswidth; + } else { + slave_config.direction = DMA_FROM_DEVICE; + slave_config.src_addr_width = buswidth; + } + + dmaengine_slave_config(chan, &slave_config); +} + +static int atmel_pcm_dma_prep(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + struct dma_chan *chan = prtd->dma_chan; + + if (prtd->desc) + /* already been there: redo the prep job */ + dmaengine_terminate_all(chan); + + /* setup dma configuration */ + atmel_pcm_dma_slave_config(substream); + + prtd->desc = chan->device->device_prep_dma_cyclic(chan, prtd->dma_buffer, + prtd->period_size * prtd->periods, + prtd->period_size, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_TO_DEVICE : DMA_FROM_DEVICE); + if (!prtd->desc) { + dev_err(&chan->dev->device, "cannot prepare slave dma\n"); + return -EINVAL; + } + + prtd->desc->callback = audio_dma_irq; + prtd->desc->callback_param = substream; + + return 0; +} /*--------------------------------------------------------------------------*\ * PCM operations @@ -172,6 +349,7 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct atmel_runtime_data *prtd = runtime->private_data; struct snd_soc_pcm_runtime *rtd = substream->private_data; + int ret; /* this may get called several times by oss emulation * with different params */ @@ -180,15 +358,35 @@ static int atmel_pcm_hw_params(struct snd_pcm_substream *substream, runtime->dma_bytes = params_buffer_bytes(params); prtd->params = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - prtd->params->dma_intr_handler = atmel_pcm_dma_irq; prtd->dma_buffer = runtime->dma_addr; prtd->dma_buffer_end = runtime->dma_addr + runtime->dma_bytes; prtd->period_size = params_period_bytes(params); + prtd->periods = params_periods(params); + + if (ssc_use_dmaengine()) { + if (prtd->dma_chan == NULL) { + ret = atmel_pcm_dma_alloc(substream, params); + if (ret) + return ret; + } + ret = atmel_pcm_dma_prep(substream); + if (ret) { + dma_release_channel(prtd->dma_chan); + prtd->dma_chan = NULL; + return ret; + } + + prtd->params->dma_intr_handler = atmel_pcm_dma_irq; + } else { + prtd->params->dma_intr_handler = atmel_pcm_pdc_irq; + } pr_debug("atmel-pcm: " - "hw_params: DMA for %s initialized " + "hw_params: %s%s for %s initialized " "(dma_bytes=%u, period_size=%u)\n", + prtd->dma_chan ? "DMA " : "PDC", + prtd->dma_chan ? dma_chan_name(prtd->dma_chan): "", prtd->params->name, runtime->dma_bytes, prtd->period_size); @@ -201,15 +399,30 @@ static int atmel_pcm_hw_free(struct snd_pcm_substream *substream) struct atmel_pcm_dma_params *params = prtd->params; if (params != NULL) { - ssc_writex(params->ssc->regs, SSC_PDC_PTCR, - params->mask->pdc_disable); + if (ssc_use_dmaengine()) { + struct dma_chan *chan = prtd->dma_chan; + + if (chan) { + dmaengine_terminate_all(chan); + prtd->cookie = 0; + prtd->desc = NULL; + dma_release_channel(chan); + prtd->dma_chan = NULL; + } + } else { + ssc_writex(params->ssc->regs, SSC_PDC_PTCR, + params->mask->pdc_disable); + } prtd->params->dma_intr_handler = NULL; } return 0; } -static int atmel_pcm_prepare(struct snd_pcm_substream *substream) +/*--------------------------------------------------------------------------*\ + * PCM callbacks using PDC +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_pdc_prepare(struct snd_pcm_substream *substream) { struct atmel_runtime_data *prtd = substream->runtime->private_data; struct atmel_pcm_dma_params *params = prtd->params; @@ -221,7 +434,7 @@ static int atmel_pcm_prepare(struct snd_pcm_substream *substream) return 0; } -static int atmel_pcm_trigger(struct snd_pcm_substream *substream, +static int atmel_pcm_pdc_trigger(struct snd_pcm_substream *substream, int cmd) { struct snd_pcm_runtime *rtd = substream->runtime; @@ -240,13 +453,13 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream, ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->period_ptr); ssc_writex(params->ssc->regs, params->pdc->xcr, - prtd->period_size / params->pdc_xfer_size); + prtd->period_size / params->data_xfer_size); prtd->period_ptr += prtd->period_size; ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->period_ptr); ssc_writex(params->ssc->regs, params->pdc->xncr, - prtd->period_size / params->pdc_xfer_size); + prtd->period_size / params->data_xfer_size); pr_debug("atmel-pcm: trigger: " "period_ptr=%lx, xpr=%u, " @@ -264,7 +477,7 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream, pr_debug("sr=%u imr=%u\n", ssc_readx(params->ssc->regs, SSC_SR), - ssc_readx(params->ssc->regs, SSC_IER)); + ssc_readx(params->ssc->regs, SSC_IMR)); break; /* SNDRV_PCM_TRIGGER_START */ case SNDRV_PCM_TRIGGER_STOP: @@ -287,7 +500,7 @@ static int atmel_pcm_trigger(struct snd_pcm_substream *substream, return ret; } -static snd_pcm_uframes_t atmel_pcm_pointer( +static snd_pcm_uframes_t atmel_pcm_pdc_pointer( struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -305,13 +518,107 @@ static snd_pcm_uframes_t atmel_pcm_pointer( return x; } +/*--------------------------------------------------------------------------*\ + * PCM callbacks using DMAENGINE +\*--------------------------------------------------------------------------*/ +static int atmel_pcm_dma_prepare(struct snd_pcm_substream *substream) +{ + struct atmel_runtime_data *prtd = substream->runtime->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + + ssc_writex(params->ssc->regs, SSC_IDR, params->mask->ssc_error); + return 0; +} + +static int atmel_pcm_dma_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_pcm_runtime *rtd = substream->runtime; + struct atmel_runtime_data *prtd = rtd->private_data; + struct atmel_pcm_dma_params *params = prtd->params; + dma_cookie_t cookie; + int ret = 0; + + pr_debug("atmel-pcm: trigger %d: buffer_size = %ld," + " dma_area = %p, dma_bytes = %u\n", + cmd, rtd->buffer_size, rtd->dma_area, rtd->dma_bytes); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + + if (prtd->cookie < DMA_MIN_COOKIE) { + cookie = dmaengine_submit(prtd->desc); + if (dma_submit_error(cookie)) { + ret = -EINVAL; + break; + } + prtd->cookie = cookie; + prtd->period_ptr = prtd->dma_buffer; + } + + + pr_debug("atmel-pcm: trigger: start sr=0x%08x imr=0x%08u\n", + ssc_readx(params->ssc->regs, SSC_SR), + ssc_readx(params->ssc->regs, SSC_IMR)); + + /* fallback */ + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + dmaengine_resume(prtd->dma_chan); + + /* It not already done or comming from xrun state */ + ssc_readx(params->ssc->regs, SSC_SR); + ssc_writex(params->ssc->regs, SSC_IER, params->mask->ssc_error); + ssc_writex(params->ssc->regs, SSC_CR, params->mask->ssc_enable); + + break; + case SNDRV_PCM_TRIGGER_STOP: + pr_debug("atmel-pcm: trigger: stop cmd = %d\n", cmd); + + ssc_writex(params->ssc->regs, SSC_IDR, params->mask->ssc_error); + + /* fallback */ + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + dmaengine_pause(prtd->dma_chan); + break; + default: + ret = -EINVAL; + } + + return ret; +} + +static snd_pcm_uframes_t atmel_pcm_dma_pointer( + struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct atmel_runtime_data *prtd = runtime->private_data; + snd_pcm_uframes_t x; + + dev_vdbg(substream->pcm->card->dev, "%s: 0x%08x %ld\n", + __func__, prtd->period_ptr, + bytes_to_frames(runtime, prtd->period_ptr - prtd->dma_buffer)); + + x = bytes_to_frames(runtime, prtd->period_ptr - prtd->dma_buffer); + + if (x >= runtime->buffer_size) + x = 0; + + return x; +} + + +/*--------------------------------------------------------------------------*\ + * PCM open/close/mmap +\*--------------------------------------------------------------------------*/ static int atmel_pcm_open(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct atmel_runtime_data *prtd; int ret = 0; - snd_soc_set_runtime_hwparams(substream, &atmel_pcm_hardware); + snd_soc_set_runtime_hwparams(substream, atmel_pcm_hardware); /* ensure that buffer size is a multiple of period size */ ret = snd_pcm_hw_constraint_integer(runtime, @@ -352,9 +659,9 @@ static struct snd_pcm_ops atmel_pcm_ops = { .ioctl = snd_pcm_lib_ioctl, .hw_params = atmel_pcm_hw_params, .hw_free = atmel_pcm_hw_free, - .prepare = atmel_pcm_prepare, - .trigger = atmel_pcm_trigger, - .pointer = atmel_pcm_pointer, + .prepare = atmel_pcm_pdc_prepare, + .trigger = atmel_pcm_pdc_trigger, + .pointer = atmel_pcm_pdc_pointer, .mmap = atmel_pcm_mmap, }; @@ -426,14 +733,16 @@ static int atmel_pcm_suspend(struct snd_soc_dai *dai) prtd = runtime->private_data; params = prtd->params; - /* disable the PDC and save the PDC registers */ + if (!ssc_use_dmaengine()) { + /* disable the PDC and save the PDC registers */ - ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_disable); - prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); - prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); - prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); - prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + prtd->pdc_xpr_save = ssc_readx(params->ssc->regs, params->pdc->xpr); + prtd->pdc_xcr_save = ssc_readx(params->ssc->regs, params->pdc->xcr); + prtd->pdc_xnpr_save = ssc_readx(params->ssc->regs, params->pdc->xnpr); + prtd->pdc_xncr_save = ssc_readx(params->ssc->regs, params->pdc->xncr); + } return 0; } @@ -450,13 +759,15 @@ static int atmel_pcm_resume(struct snd_soc_dai *dai) prtd = runtime->private_data; params = prtd->params; - /* restore the PDC registers and enable the PDC */ - ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); - ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); - ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); - ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); + if (!ssc_use_dmaengine()) { + /* restore the PDC registers and enable the PDC */ + ssc_writex(params->ssc->regs, params->pdc->xpr, prtd->pdc_xpr_save); + ssc_writex(params->ssc->regs, params->pdc->xcr, prtd->pdc_xcr_save); + ssc_writex(params->ssc->regs, params->pdc->xnpr, prtd->pdc_xnpr_save); + ssc_writex(params->ssc->regs, params->pdc->xncr, prtd->pdc_xncr_save); - ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + ssc_writel(params->ssc->regs, PDC_PTCR, params->mask->pdc_enable); + } return 0; } #else @@ -495,6 +806,15 @@ static struct platform_driver atmel_pcm_driver = { static int __init snd_atmel_pcm_init(void) { + if (ssc_use_dmaengine()) { + atmel_pcm_ops.prepare = atmel_pcm_dma_prepare; + atmel_pcm_ops.trigger = atmel_pcm_dma_trigger; + atmel_pcm_ops.pointer = atmel_pcm_dma_pointer; + + atmel_pcm_hardware = &atmel_pcm_dma_hardware; + } else { + atmel_pcm_hardware = &atmel_pcm_pdc_hardware; + } return platform_driver_register(&atmel_pcm_driver); } module_init(snd_atmel_pcm_init); diff --git a/sound/soc/atmel/atmel-pcm.h b/sound/soc/atmel/atmel-pcm.h index 5e0a95e..b007376 100644 --- a/sound/soc/atmel/atmel-pcm.h +++ b/sound/soc/atmel/atmel-pcm.h @@ -50,6 +50,7 @@ struct atmel_pdc_regs { struct atmel_ssc_mask { u32 ssc_enable; /* SSC recv/trans enable */ u32 ssc_disable; /* SSC recv/trans disable */ + u32 ssc_error; /* SSC error conditions */ u32 ssc_endx; /* SSC ENDTX or ENDRX */ u32 ssc_endbuf; /* SSC TXBUFE or RXBUFF */ u32 pdc_enable; /* PDC recv/trans enable */ @@ -66,7 +67,8 @@ struct atmel_ssc_mask { */ struct atmel_pcm_dma_params { char *name; /* stream identifier */ - int pdc_xfer_size; /* PDC counter increment in bytes */ + int data_xfer_size; /* PDC counter increment in bytes, + DMA data transfer size in bytes */ struct ssc_device *ssc; /* SSC device for stream */ struct atmel_pdc_regs *pdc; /* PDC receive or transmit registers */ struct atmel_ssc_mask *mask; /* SSC & PDC status bits */ diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index a7a7bbc..9412fc3 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -48,7 +48,8 @@ #include "atmel_ssc_dai.h" -#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) +#if defined(CONFIG_ARCH_AT91SAM9260) || defined(CONFIG_ARCH_AT91SAM9G20) \ + || defined(CONFIG_ARCH_AT91SAM9X5) #define NUM_SSC_DEVICES 1 #else #define NUM_SSC_DEVICES 3 @@ -86,6 +87,7 @@ static struct atmel_ssc_mask ssc_tx_mask = { static struct atmel_ssc_mask ssc_rx_mask = { .ssc_enable = SSC_BIT(CR_RXEN), .ssc_disable = SSC_BIT(CR_RXDIS), + .ssc_error = SSC_BIT(SR_OVRUN), .ssc_endx = SSC_BIT(SR_ENDRX), .ssc_endbuf = SSC_BIT(SR_RXBUFF), .pdc_enable = ATMEL_PDC_RXTEN, @@ -183,7 +185,8 @@ static irqreturn_t atmel_ssc_interrupt(int irq, void *dev_id) if ((dma_params != NULL) && (dma_params->dma_intr_handler != NULL)) { ssc_substream_mask = (dma_params->mask->ssc_endx | - dma_params->mask->ssc_endbuf); + dma_params->mask->ssc_endbuf | + dma_params->mask->ssc_error); if (ssc_sr & ssc_substream_mask) { dma_params->dma_intr_handler(ssc_sr, dma_params-> @@ -376,19 +379,19 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: bits = 8; - dma_params->pdc_xfer_size = 1; + dma_params->data_xfer_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: bits = 16; - dma_params->pdc_xfer_size = 2; + dma_params->data_xfer_size = 2; break; case SNDRV_PCM_FORMAT_S24_LE: bits = 24; - dma_params->pdc_xfer_size = 4; + dma_params->data_xfer_size = 4; break; case SNDRV_PCM_FORMAT_S32_LE: bits = 32; - dma_params->pdc_xfer_size = 4; + dma_params->data_xfer_size = 4; break; default: printk(KERN_WARNING "atmel_ssc_dai: unsupported PCM format"); @@ -561,15 +564,17 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, /* Reset the SSC and its PDC registers */ ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); - ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); + if (!ssc_use_dmaengine()) { + ssc_writel(ssc_p->ssc->regs, PDC_RPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_RNCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); - ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TCR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNPR, 0); + ssc_writel(ssc_p->ssc->regs, PDC_TNCR, 0); + } ret = request_irq(ssc_p->ssc->irq, atmel_ssc_interrupt, 0, ssc_p->name, ssc_p); -- 1.7.3 _______________________________________________ linux-arm-kernel mailing list linux-arm-kernel(a)lists.infradead.org http://lists.infradead.org/mailman/listinfo/linux-arm-kernel

3 2

[alsa-devel] asoc: how to enable micbias and codec power when headset inserted
by Babu, Ramesh 29 Jun '11

29 Jun '11

Hi I have question on handling mic bias and codec bias when headset is inserted. When headset is inserted, the mic bias and codec bias should be ON, in order to get the button press interrupts from the headset. Do I need to have dapm path to achieve this? Reference to existing code would be nice ! Regards Ramesh

2 1

[alsa-devel] [PATCH] ASoC: Fix Blackfin I2S _pointer() implementation return in bounds values
by Mark Brown 29 Jun '11

29 Jun '11

The Blackfin DMA controller can report one frame beyond the end of the buffer in the wraparound case but ALSA requires that the pointer always be in the buffer. Do the wraparound to handle this. A similar bug is likely to apply to the other Blackfin PCM drivers but the code is less obvious to inspection and I don't have a user to test. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/blackfin/bf5xx-i2s-pcm.c | 14 ++++++++++++-- 1 files changed, 12 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index 4a805a8..819d3bd 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -138,11 +138,21 @@ static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) pr_debug("%s enter\n", __func__); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { diff = sport_curr_offset_tx(sport); - frames = bytes_to_frames(substream->runtime, diff); } else { diff = sport_curr_offset_rx(sport); - frames = bytes_to_frames(substream->runtime, diff); } + + /* + * TX at least can report one frame beyond the end of the + * buffer if we hit the wraparound case - clamp to within the + * buffer as the ALSA APIs require. + */ + if (diff == snd_pcm_lib_buffer_bytes(substream)) + diff = 0; + } + + frames = bytes_to_frames(substream->runtime, diff); + return frames; } -- 1.7.5.4

3 3

[alsa-devel] [PATCH] ASoC: Correct left/right swap in wm_hubs DC offset correction
by Mark Brown 29 Jun '11

29 Jun '11

It was consistently wrong for everything except WM8993 so should be no functional change. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_hubs.c | 16 ++++++++-------- 1 files changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 4ebc131..2d6c88b 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -133,9 +133,9 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) break; case 1: reg = snd_soc_read(codec, WM8993_DC_SERVO_3); - reg_l = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) + reg_r = (reg & WM8993_DCS_DAC_WR_VAL_1_MASK) >> WM8993_DCS_DAC_WR_VAL_1_SHIFT; - reg_r = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; + reg_l = reg & WM8993_DCS_DAC_WR_VAL_0_MASK; break; default: WARN(1, "Unknown DCS readback method\n"); @@ -149,13 +149,13 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Applying %d code DC servo correction\n", hubs->dcs_codes); - /* HPOUT1L */ - offset = reg_l; + /* HPOUT1R */ + offset = reg_r; offset += hubs->dcs_codes; dcs_cfg = (u8)offset << WM8993_DCS_DAC_WR_VAL_1_SHIFT; - /* HPOUT1R */ - offset = reg_r; + /* HPOUT1L */ + offset = reg_l; offset += hubs->dcs_codes; dcs_cfg |= (u8)offset; @@ -167,8 +167,8 @@ static void calibrate_dc_servo(struct snd_soc_codec *codec) WM8993_DCS_TRIG_DAC_WR_0 | WM8993_DCS_TRIG_DAC_WR_1); } else { - dcs_cfg = reg_l << WM8993_DCS_DAC_WR_VAL_1_SHIFT; - dcs_cfg |= reg_r; + dcs_cfg = reg_r << WM8993_DCS_DAC_WR_VAL_1_SHIFT; + dcs_cfg |= reg_l; } /* Save the callibrated offset if we're in class W mode and -- 1.7.5.4

2 1

[alsa-devel] [PATCH] ASoC: Change WM9081 speaker output enable to _OUT_DRV
by Mark Brown 29 Jun '11

29 Jun '11

More for neatness than any actual performance improvement. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9081.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 91c6b39..a469132 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -727,7 +727,7 @@ SND_SOC_DAPM_MIXER_NAMED_CTL("Mixer", SND_SOC_NOPM, 0, 0, SND_SOC_DAPM_PGA("LINEOUT PGA", WM9081_POWER_MANAGEMENT, 4, 0, NULL, 0), SND_SOC_DAPM_PGA("Speaker PGA", WM9081_POWER_MANAGEMENT, 2, 0, NULL, 0), -SND_SOC_DAPM_PGA("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0), +SND_SOC_DAPM_OUT_DRV("Speaker", WM9081_POWER_MANAGEMENT, 1, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LINEOUT"), SND_SOC_DAPM_OUTPUT("SPKN"), -- 1.7.5.4

2 1

[alsa-devel] ASoC: imx dma tx burst size set to 6 and to 4
by Lambrecht Jürgen 29 Jun '11

29 Jun '11

Hello, There is an inconsistency in this code: /sound/soc/imx/imx-ssi.c: ssi->dma_params_tx.burstsize = 4; /sound/soc/imx/imx-pcm-dma-mx2.c: ssi->dma_params_tx.burstsize = 6; Datasheet: This sets the field TFWM0 in register SFCSR (SSI_SFCSR_TFWM0): the number of data words needed to set the empty flag (TFE0). Can be set from 1 to 8; FIFO size is 8 words (of 24b). And this is also the DMA burst size. Which is best depends on the application (the higher the more efficient, but the more risk for a FIFO underrun). I would take 4. But I guess it should only be set at 1 place, and what is then the best place? Regards, Jürgen details: /sound/soc/imx/imx-ssi.c: ssi->dma_params_tx.burstsize = 4; Set in commit commit 0a93421b6adf8ba127b3eafc4c16e3a14017e2ae Author: Javier Martin <javier.martin(a)vista-silicon.com> Date: Tue Mar 1 15:02:06 2011 +0100 ASoC: Fix burstsize and DSP_B format problems in imx-ssi. When choosing IMX_DMA flag, burtsizes are set to its default value (0) which leads to driver malfunction. Change them to 4. DSP_B interface needs additional flag to match DSP_B formats as described in several codecs as wm8741 and aic3205. Signed-off-by: Javier Martin <javier.martin(a)vista-silicon.com> Acked-by: Liam Girdwood <lrg(a)slimlogic.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Index in diff: diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 30894ea..bc92ec6 100644 /sound/soc/imx/imx-pcm-dma-mx2.c: ssi->dma_params_tx.burstsize = 6; Set in commit commit 2c4cf17a52f04fbe929977252d5b8ab81d2c6e9b Author: Wolfram Sang <w.sang(a)pengutronix.de> Date: Fri Mar 25 16:51:44 2011 +0100 ASoC: imx: set watermarks for mx2-dma They got accidently removed by f0fba2a (ASoC: multi-component - ASoC Multi-Component Support). Reintroduce them and get rid of the superfluous defines because the fiq-driver has its own hardcoded values. Signed-off-by: Wolfram Sang <w.sang(a)pengutronix.de> Acked-by: Liam Girdwood <lrg(a)ti.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: stable(a)kernel.org Index in diff: diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c index 671ef8d..b2ed764 100644 -- Jürgen Lambrecht R&D Associate Tel: +32 (0)51 303045 Fax: +32 (0)51 310670 http://www.televic-rail.com Televic Rail NV - Leo Bekaertlaan 1 - 8870 Izegem - Belgium Company number 0825.539.581 - RPR Kortrijk

1 0

[alsa-devel] [PATCH] pass ELD to HDMI/DP audio driver
by Wu Fengguang 29 Jun '11

29 Jun '11

Add ELD support for Eaglelake, IbexPeak/Ironlake and SandyBridge/CougarPoint. ELD (EDID-Like Data) describes to the HDMI/DP audio driver the audio capabilities of the plugged monitor. It's built and passed to audio driver in 2 steps: (1) at get_modes time, parse EDID and save ELD to drm_connector.eld[] (2) at mode_set time, write drm_connector.eld[] to the Transcoder's hw ELD buffer and set the ELD_valid bit to inform HDMI/DP audio driver ELD selection policy: it's possible for one encoder to be associated with multiple connectors (ie. monitors), in which case the first found ELD will be used. This policy may not be suitable for all users, but let's start it simple first. The impact of ELD selection policy: assume there are two monitors, one supports stereo playback and the other has 8-channel output; cloned display mode is used, so that the two monitors are associated with the same internal encoder. If only the stereo playback capability is reported, the user won't be able to start 8-channel playback; if the 8-channel ELD is reported, then user space applications may send 8-channel samples down, however the user may actually be listening to the 2-channel monitor and not connecting speakers to the 8-channel monitor. Overall, it's more safe to report maximum profiles to the user space, so that the user can at least be able to do 8-channel playback if he want to. This patch is tested OK on G45/HDMI and IbexPeak/HDMI. DisplayPort is tested on several IbexPeak and Sandybridge boxes, however not working, possibly due to hardware/monitor problems. One known problem is that the GEN6_AUD_CNTL_ST/DIP_Port_Select field is always 0 (reserved). Without knowing the port number, I worked it around by setting the ELD_valid bit for ALL the three ports. CC: Zhao Yakui <yakui.zhao(a)intel.com> CC: Wang Zhenyu <zhenyu.z.wang(a)intel.com> CC: Jeremy Bush <contractfrombelow(a)gmail.com> CC: Christopher White <c.white(a)pulseforce.com> CC: "Bossart, Pierre-louis" <pierre-louis.bossart(a)intel.com> Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- drivers/gpu/drm/drm_edid.c | 171 +++++++++++++++++++++++++ drivers/gpu/drm/i915/i915_drv.h | 2 drivers/gpu/drm/i915/i915_reg.h | 20 ++ drivers/gpu/drm/i915/intel_display.c | 119 +++++++++++++++++ drivers/gpu/drm/i915/intel_dp.c | 2 drivers/gpu/drm/i915/intel_drv.h | 3 drivers/gpu/drm/i915/intel_hdmi.c | 2 drivers/gpu/drm/i915/intel_modes.c | 2 include/drm/drm_crtc.h | 10 + include/drm/drm_edid.h | 9 + 10 files changed, 340 insertions(+) --- sound-2.6.orig/drivers/gpu/drm/drm_edid.c 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/drivers/gpu/drm/drm_edid.c 2011-06-29 14:13:20.000000000 +0800 @@ -1292,6 +1292,7 @@ add_detailed_modes(struct drm_connector #define HDMI_IDENTIFIER 0x000C03 #define AUDIO_BLOCK 0x01 #define VENDOR_BLOCK 0x03 +#define SPEAKER_BLOCK 0x04 #define EDID_BASIC_AUDIO (1 << 6) /** @@ -1320,6 +1321,176 @@ u8 *drm_find_cea_extension(struct edid * } EXPORT_SYMBOL(drm_find_cea_extension); +static void +parse_hdmi_vsdb(struct drm_connector *connector, uint8_t *db) +{ + connector->eld[5] |= (db[6] >> 7) << 1; /* Supports_AI */ + + connector->dvi_dual = db[6] & 1; + connector->max_tmds_clock = db[7] * 5; + + connector->latency_present[0] = db[8] >> 7; + connector->latency_present[1] = (db[8] >> 6) & 1; + connector->video_latency[0] = db[9]; + connector->audio_latency[0] = db[10]; + connector->video_latency[1] = db[11]; + connector->audio_latency[1] = db[12]; + + DRM_LOG_KMS("HDMI: DVI dual %d, " + "max TMDS clock %d, " + "latency present %d %d, " + "video latency %d %d, " + "audio latency %d %d\n", + connector->dvi_dual, + connector->max_tmds_clock, + (int) connector->latency_present[0], + (int) connector->latency_present[1], + connector->video_latency[0], + connector->video_latency[1], + connector->audio_latency[0], + connector->audio_latency[1]); +} + +static void +monitor_name(struct detailed_timing *t, void *data) +{ + if (t->data.other_data.type == EDID_DETAIL_MONITOR_NAME) + *(u8 **)data = t->data.other_data.data.str.str; +} + +/** + * drm_edid_to_eld - build ELD from EDID + * @connector: connector corresponding to the HDMI/DP sink + * @edid: EDID to parse + * + * Fill the ELD (EDID-Like Data) buffer for passing to the audio driver. + * Some ELD fields are left to the graphics driver caller: + * - Conn_Type + * - HDCP + * - Port_ID + */ +void drm_edid_to_eld(struct drm_connector *connector, struct edid *edid) +{ + uint8_t *eld = connector->eld; + u8 *cea; + u8 *name; + u8 *db; + int sad_count = 0; + int mnl; + int dbl; + + memset(eld, 0, sizeof(connector->eld)); + + cea = drm_find_cea_extension(edid); + if (!cea) { + DRM_DEBUG_KMS("ELD: no CEA Extension found\n"); + return; + } + + name = NULL; + drm_for_each_detailed_block((u8 *)edid, monitor_name, &name); + for (mnl = 0; name && mnl < 13; mnl++) { + if (name[mnl] == 0x0a) + break; + eld[20 + mnl] = name[mnl]; + } + eld[4] = (cea[1] << 5) | mnl; + DRM_DEBUG_KMS("ELD monitor %s\n", eld + 20); + + eld[0] = 2 << 3; /* ELD version: 2 */ + + eld[16] = edid->mfg_id[0]; + eld[17] = edid->mfg_id[1]; + eld[18] = edid->prod_code[0]; + eld[19] = edid->prod_code[1]; + + for (db = cea + 4; db < cea + cea[2]; db += dbl + 1) { + dbl = db[0] & 0x1f; + + switch ((db[0] & 0xe0) >> 5) { + case AUDIO_BLOCK: /* Audio Data Block, contains SADs */ + sad_count = dbl / 3; + memcpy(eld + 20 + mnl, &db[1], dbl); + break; + case SPEAKER_BLOCK: /* Speaker Allocation Data Block */ + eld[7] = db[1]; + break; + case VENDOR_BLOCK: + /* HDMI Vendor-Specific Data Block */ + if (db[1] == 0x03 && db[2] == 0x0c && db[3] == 0) + parse_hdmi_vsdb(connector, db); + break; + default: + break; + } + } + eld[5] |= sad_count << 4; + eld[2] = (20 + mnl + sad_count * 3 + 3) / 4; + + DRM_DEBUG_KMS("ELD size %d, SAD count %d\n", (int)eld[2], sad_count); +} +EXPORT_SYMBOL(drm_edid_to_eld); + +/** + * drm_av_sync_delay - HDMI/DP sink audio-video sync delay in milli-seconds + * @connector: connector associated with the HDMI/DP sink + * @mode: the display mode + */ +int drm_av_sync_delay(struct drm_connector *connector, + struct drm_display_mode *mode) +{ + int i = !!(mode->flags & DRM_MODE_FLAG_INTERLACE); + int a, v; + + if (!connector->latency_present[0]) + return 0; + if (!connector->latency_present[1]) + i = 0; + + a = connector->audio_latency[i]; + v = connector->video_latency[i]; + + /* + * HDMI/DP sink doesn't support audio or video? + */ + if (a == 255 || v == 255) + return 0; + + /* + * Convert raw edid values to milli-seconds. + * Treat unknown latency as 0ms. + */ + if (a) + a = 2 * (a - 1); + if (v) + v = 2 * (v - 1); + + return max(v - a, 0); +} +EXPORT_SYMBOL(drm_av_sync_delay); + +/** + * drm_select_eld - select one ELD from multiple HDMI/DP sinks + * @encoder: the encoder just changed display mode + * @mode: the adjusted display mode + * + * It's possible for one encoder to be associated with multiple HDMI/DP sinks. + * The policy is now hard coded to simply use the first HDMI/DP sink's ELD. + */ +struct drm_connector *drm_select_eld(struct drm_encoder *encoder, + struct drm_display_mode *mode) +{ + struct drm_connector *connector; + struct drm_device *dev = encoder->dev; + + list_for_each_entry(connector, &dev->mode_config.connector_list, head) + if (connector->encoder == encoder) + return connector; + + return NULL; +} +EXPORT_SYMBOL(drm_select_eld); + /** * drm_detect_hdmi_monitor - detect whether monitor is hdmi. * @edid: monitor EDID information --- sound-2.6.orig/drivers/gpu/drm/i915/intel_hdmi.c 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/drivers/gpu/drm/i915/intel_hdmi.c 2011-06-29 13:19:35.000000000 +0800 @@ -151,6 +151,8 @@ static void intel_hdmi_mode_set(struct d POSTING_READ(intel_hdmi->sdvox_reg); intel_hdmi_set_avi_infoframe(encoder); + + intel_write_eld(encoder, crtc, adjusted_mode); } static void intel_hdmi_dpms(struct drm_encoder *encoder, int mode) --- sound-2.6.orig/include/drm/drm_edid.h 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/include/drm/drm_edid.h 2011-06-29 13:19:35.000000000 +0800 @@ -230,4 +230,13 @@ struct edid { #define EDID_PRODUCT_ID(e) ((e)->prod_code[0] | ((e)->prod_code[1] << 8)) +struct drm_encoder; +struct drm_connector; +struct drm_display_mode; +void drm_edid_to_eld(struct drm_connector *connector, struct edid *edid); +int drm_av_sync_delay(struct drm_connector *connector, + struct drm_display_mode *mode); +struct drm_connector *drm_select_eld(struct drm_encoder *encoder, + struct drm_display_mode *mode); + #endif /* __DRM_EDID_H__ */ --- sound-2.6.orig/drivers/gpu/drm/i915/intel_display.c 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/drivers/gpu/drm/i915/intel_display.c 2011-06-29 14:05:05.000000000 +0800 @@ -30,6 +30,7 @@ #include <linux/kernel.h> #include <linux/slab.h> #include <linux/vgaarb.h> +#include <drm/drm_edid.h> #include "drmP.h" #include "intel_drv.h" #include "i915_drm.h" @@ -5297,6 +5298,121 @@ static int intel_crtc_mode_set(struct dr return ret; } +static void g4x_write_eld(struct drm_connector *connector, + struct drm_crtc *crtc) +{ + struct drm_i915_private *dev_priv = connector->dev->dev_private; + uint8_t *eld = connector->eld; + uint32_t eldv; + uint32_t len; + uint32_t i; + + i = I915_READ(G4X_AUD_VID_DID); + + if (i == INTEL_AUDIO_DEVBLC || i == INTEL_AUDIO_DEVCL) + eldv = G4X_ELDV_DEVCL_DEVBLC; + else + eldv = G4X_ELDV_DEVCTG; + + i = I915_READ(G4X_AUD_CNTL_ST); + i &= ~(eldv | G4X_ELD_ADDR); + len = (i >> 9) & 0x1f; /* ELD buffer size */ + I915_WRITE(G4X_AUD_CNTL_ST, i); + + if (!eld[0]) + return; + + len = min_t(uint8_t, eld[2], len); + for (i = 0; i < len; i++) + I915_WRITE(G4X_HDMIW_HDMIEDID, *((uint32_t *)eld + i)); + + i = I915_READ(G4X_AUD_CNTL_ST); + i |= eldv; + I915_WRITE(G4X_AUD_CNTL_ST, i); +} + +static void ironlake_write_eld(struct drm_connector *connector, + struct drm_crtc *crtc) +{ + struct drm_i915_private *dev_priv = connector->dev->dev_private; + uint8_t *eld = connector->eld; + uint32_t eldv; + uint32_t i; + int len; + int hdmiw_hdmiedid; + int aud_cntl_st; + + i = to_intel_crtc(crtc)->pipe; + hdmiw_hdmiedid = GEN6_HDMIW_HDMIEDID_A + i * 0x100; + aud_cntl_st = GEN6_AUD_CNTL_ST_A + i * 0x100; + + DRM_DEBUG_DRIVER("ELD on pipe %c\n", pipe_name(i)); + + i = I915_READ(aud_cntl_st); + i = (i >> 29) & 0x3; /* DIP_Port_Select, 0x1 = PortB */ + if (!i) { + DRM_DEBUG_DRIVER("Audio directed to unknown port\n"); + /* operate blindly on all ports */ + eldv = GEN6_ELD_VALIDB; + eldv |= GEN6_ELD_VALIDB << 4; + eldv |= GEN6_ELD_VALIDB << 8; + } else { + DRM_DEBUG_DRIVER("ELD on port %c\n", 'A' + i); + eldv = GEN6_ELD_VALIDB << ((i - 1) * 4); + } + + i = I915_READ(GEN6_AUD_CNTL_ST2); + i &= ~eldv; + I915_WRITE(GEN6_AUD_CNTL_ST2, i); + + if (!eld[0]) + return; + + if (intel_pipe_has_type(crtc, INTEL_OUTPUT_DISPLAYPORT)) { + DRM_DEBUG_DRIVER("ELD: DisplayPort detected\n"); + eld[5] |= (1 << 2); /* Conn_Type, 0x1 = DisplayPort */ + } + + i = I915_READ(aud_cntl_st); + i &= ~GEN6_ELD_ADDRESS; + I915_WRITE(aud_cntl_st, i); + + i = I915_READ(aud_cntl_st); + len = (i & GEN6_ELD_BUFFER_SIZE) >> 10; + len = min_t(uint8_t, eld[2], len); + DRM_DEBUG_DRIVER("ELD size %d\n", len); + for (i = 0; i < len; i++) + I915_WRITE(hdmiw_hdmiedid, *((uint32_t *)eld + i)); + + i = I915_READ(GEN6_AUD_CNTL_ST2); + i |= eldv; + I915_WRITE(GEN6_AUD_CNTL_ST2, i); +} + +void intel_write_eld(struct drm_encoder *encoder, + struct drm_crtc *crtc, + struct drm_display_mode *mode) +{ + struct drm_connector *connector; + struct drm_device *dev = encoder->dev; + struct drm_i915_private *dev_priv = dev->dev_private; + + connector = drm_select_eld(encoder, mode); + if (!connector) + return; + + DRM_DEBUG_DRIVER("ELD on [CONNECTOR:%d:%s], [ENCODER:%d:%s]\n", + connector->base.id, + drm_get_connector_name(connector), + connector->encoder->base.id, + drm_get_encoder_name(connector->encoder)); + + connector->eld[6] = drm_av_sync_delay(connector, mode) / 2; + + if (dev_priv->display.write_eld) + dev_priv->display.write_eld(connector, crtc); +} + /** Loads the palette/gamma unit for the CRTC with the prepared values */ void intel_crtc_load_lut(struct drm_crtc *crtc) { @@ -7636,6 +7752,7 @@ static void intel_init_display(struct dr } dev_priv->display.fdi_link_train = ironlake_fdi_link_train; dev_priv->display.init_clock_gating = ironlake_init_clock_gating; + dev_priv->display.write_eld = ironlake_write_eld; } else if (IS_GEN6(dev)) { if (SNB_READ_WM0_LATENCY()) { dev_priv->display.update_wm = sandybridge_update_wm; @@ -7646,6 +7763,7 @@ static void intel_init_display(struct dr } dev_priv->display.fdi_link_train = gen6_fdi_link_train; dev_priv->display.init_clock_gating = gen6_init_clock_gating; + dev_priv->display.write_eld = ironlake_write_eld; } else if (IS_IVYBRIDGE(dev)) { /* FIXME: detect B0+ stepping and use auto training */ dev_priv->display.fdi_link_train = ivb_manual_fdi_link_train; @@ -7676,6 +7794,7 @@ static void intel_init_display(struct dr } else dev_priv->display.update_wm = pineview_update_wm; } else if (IS_G4X(dev)) { + dev_priv->display.write_eld = g4x_write_eld; dev_priv->display.update_wm = g4x_update_wm; dev_priv->display.init_clock_gating = g4x_init_clock_gating; } else if (IS_GEN4(dev)) { --- sound-2.6.orig/drivers/gpu/drm/i915/i915_reg.h 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/drivers/gpu/drm/i915/i915_reg.h 2011-06-29 13:19:35.000000000 +0800 @@ -3436,4 +3436,24 @@ #define GEN6_PCODE_WRITE_MIN_FREQ_TABLE 0x9 #define GEN6_PCODE_DATA 0x138128 +#define G4X_AUD_VID_DID 0x62020 +#define INTEL_AUDIO_DEVCL 0x808629FB +#define INTEL_AUDIO_DEVBLC 0x80862801 +#define INTEL_AUDIO_DEVCTG 0x80862802 + +#define G4X_AUD_CNTL_ST 0x620B4 +#define G4X_ELDV_DEVCL_DEVBLC (1 << 13) +#define G4X_ELDV_DEVCTG (1 << 14) +#define G4X_ELD_ADDR (0xf << 5) +#define G4X_ELD_ACK (1 << 4) +#define G4X_HDMIW_HDMIEDID 0x6210C +#define GEN6_HDMIW_HDMIEDID_A 0xE2050 +#define GEN6_AUD_CNTL_ST_A 0xE20B4 +#define GEN6_ELD_BUFFER_SIZE (0x1f << 10) +#define GEN6_ELD_ADDRESS (0x1f << 5) +#define GEN6_ELD_ACK (1 << 4) +#define GEN6_AUD_CNTL_ST2 0xE20C0 +#define GEN6_ELD_VALIDB (1 << 0) +#define GEN6_CP_READYB (1 << 1) + #endif /* _I915_REG_H_ */ --- sound-2.6.orig/drivers/gpu/drm/i915/intel_drv.h 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/drivers/gpu/drm/i915/intel_drv.h 2011-06-29 13:19:35.000000000 +0800 @@ -346,4 +346,7 @@ extern void intel_fb_output_poll_changed extern void intel_fb_restore_mode(struct drm_device *dev); extern void intel_init_clock_gating(struct drm_device *dev); +extern void intel_write_eld(struct drm_encoder *encoder, + struct drm_crtc *crtc, + struct drm_display_mode *mode); #endif /* __INTEL_DRV_H__ */ --- sound-2.6.orig/drivers/gpu/drm/i915/intel_modes.c 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/drivers/gpu/drm/i915/intel_modes.c 2011-06-29 13:19:35.000000000 +0800 @@ -26,6 +26,7 @@ #include <linux/slab.h> #include <linux/i2c.h> #include <linux/fb.h> +#include <drm/drm_edid.h> #include "drmP.h" #include "intel_drv.h" #include "i915_drv.h" @@ -74,6 +75,7 @@ int intel_ddc_get_modes(struct drm_conne if (edid) { drm_mode_connector_update_edid_property(connector, edid); ret = drm_add_edid_modes(connector, edid); + drm_edid_to_eld(connector, edid); connector->display_info.raw_edid = NULL; kfree(edid); } --- sound-2.6.orig/include/drm/drm_crtc.h 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/include/drm/drm_crtc.h 2011-06-29 13:19:35.000000000 +0800 @@ -464,6 +464,8 @@ enum drm_connector_force { /* DACs should rarely do this without a lot of testing */ #define DRM_CONNECTOR_POLL_DISCONNECT (1 << 2) +#define MAX_ELD_BYTES 128 + /** * drm_connector - central DRM connector control structure * @crtc: CRTC this connector is currently connected to, NULL if none @@ -520,6 +522,14 @@ struct drm_connector { uint32_t encoder_ids[DRM_CONNECTOR_MAX_ENCODER]; uint32_t force_encoder_id; struct drm_encoder *encoder; /* currently active encoder */ + + /* EDID bits */ + uint8_t eld[MAX_ELD_BYTES]; + bool dvi_dual; + int max_tmds_clock; /* in MHz */ + bool latency_present[2]; + int video_latency[2]; /* [0]: progressive, [1]: interlaced */ + int audio_latency[2]; }; /** --- sound-2.6.orig/drivers/gpu/drm/i915/intel_dp.c 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/drivers/gpu/drm/i915/intel_dp.c 2011-06-29 13:19:35.000000000 +0800 @@ -793,6 +793,8 @@ intel_dp_mode_set(struct drm_encoder *en else intel_dp->DP |= DP_PLL_FREQ_270MHZ; } + + intel_write_eld(encoder, crtc, adjusted_mode); } static void ironlake_edp_panel_vdd_on(struct intel_dp *intel_dp) --- sound-2.6.orig/drivers/gpu/drm/i915/i915_drv.h 2011-06-29 13:18:31.000000000 +0800 +++ sound-2.6/drivers/gpu/drm/i915/i915_drv.h 2011-06-29 13:19:35.000000000 +0800 @@ -208,6 +208,8 @@ struct drm_i915_display_funcs { struct drm_display_mode *adjusted_mode, int x, int y, struct drm_framebuffer *old_fb); + void (*write_eld)(struct drm_connector *connector, + struct drm_crtc *crtc); void (*fdi_link_train)(struct drm_crtc *crtc); void (*init_clock_gating)(struct drm_device *dev); void (*init_pch_clock_gating)(struct drm_device *dev);

2 3

[alsa-devel] opening a device multiple times?
by Ákos Maróy 29 Jun '11

29 Jun '11

Hi, I'm writing a piece of code that uses ALSA audio devices, where I open the same output device multiple times, as a simple way to have parallel audio channels, where I can send audio data to each 'channel' (PCM handle) simultaneously, and the ALSA layer will mix the audio together. My experience is that this works with some audio devices (for example, with a pulse audio - ALSA gateway), but not on some other devices (for example, on a USB sound device, driven by ALSA). In the latter case, snd_pcm_open() will refuse to open the audio device multiple times. I wonder what the generic concept here is - is it generally 'OK' to open an audio device multiple times? If not, why not? :) Akos

2 1

Re: [alsa-devel] [PATCH] ASoC: Rename WM8915 to WM8996
by Kukjin Kim 29 Jun '11

29 Jun '11

Mark Brown wrote: > > For marketing reasons the part will be called WM8996. In order to avoid > user confusion rename the driver to reflect this. > > Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> > --- > > The cross tree issues make this difficult to merge as there's > dependencies in both ASoC and Samsung trees - Kukjin, would you be OK > with merging this via ALSA late on in the merge window after both > Samsung and ASoC trees have merged? > Yes, it's ok to me :) As a note, basically my tree will be sent to upstream via rmk during merge window so I'm not sure when it is merged. > I will have a go at constructing some branches which do the right thing > and can be merged now (so a base rename patch then a merge of that into > ASoC which deals with all the work that has been done in the driver this > release cycle) but I wanted to get this first option out there now. > > arch/arm/mach-s3c64xx/mach-crag6410.c | 18 +- If required, Acked-by: Kukjin Kim <kgene.kim(a)samsung.com> Thanks. Best regards, Kgene. -- Kukjin Kim <kgene.kim(a)samsung.com>, Senior Engineer, SW Solution Development Team, Samsung Electronics Co., Ltd.

2 1

[alsa-devel] [PATCH] aplay: increase channel limit to 256
by Tim Blechmann 29 Jun '11

29 Jun '11

aplay is limited to 32 channels, which makes it impossible to use it for testing devices with more channels. here we increase the limit to 256, which should be sufficient for a virtual device made of 4 64-channel soundcards. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- aplay/aplay.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index 50e01ef..07e6bc0 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -526,7 +526,7 @@ int main(int argc, char *argv[]) break; case 'c': rhwparams.channels = strtol(optarg, NULL, 0); - if (rhwparams.channels < 1 || rhwparams.channels > 32) { + if (rhwparams.channels < 1 || rhwparams.channels > 256) { error(_("value %i for channels is invalid"), rhwparams.channels); return 1; } @@ -1015,7 +1015,7 @@ static int test_au(int fd, void *buffer) if (hwparams.rate < 2000 || hwparams.rate > 256000) return -1; hwparams.channels = BE_INT(ap->channels); - if (hwparams.channels < 1 || hwparams.channels > 128) + if (hwparams.channels < 1 || hwparams.channels > 256) return -1; if ((size_t)safe_read(fd, buffer + sizeof(AuHeader), BE_INT(ap->hdr_size) - sizeof(AuHeader)) != BE_INT(ap->hdr_size) - sizeof(AuHeader)) { error(_("read error")); -- 1.7.4.1

2 1

[alsa-devel] [PATCH] HDMI - fix ELD monitor name length
by Wu Fengguang 29 Jun '11

29 Jun '11

I noticed that the last character of the ELD monitor name is lost, this fixes the issue. Signed-off-by: Wu Fengguang <fengguang.wu(a)intel.com> --- sound/pci/hda/hda_eld.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) --- sound-2.6.orig/sound/pci/hda/hda_eld.c 2011-06-29 13:08:23.000000000 +0800 +++ sound-2.6/sound/pci/hda/hda_eld.c 2011-06-29 14:13:25.000000000 +0800 @@ -294,7 +294,7 @@ static int hdmi_update_eld(struct hdmi_e snd_printd(KERN_INFO "HDMI: out of range MNL %d\n", mnl); goto out_fail; } else - strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl); + strlcpy(e->monitor_name, buf + ELD_FIXED_BYTES, mnl + 1); for (i = 0; i < e->sad_count; i++) { if (ELD_FIXED_BYTES + mnl + 3 * (i + 1) > size) {

2 2

Re: [alsa-devel] Bug#631963: linux-image-3.0.0-rc4-686-pae: no sound out the rear plugs
by Takashi Iwai 29 Jun '11

29 Jun '11

At Wed, 29 Jun 2011 04:58:53 +0100, Ben Hutchings wrote: > > On Tue, 2011-06-28 at 19:24 +0200, Toni Mueller wrote: > > Package: linux-2.6 > > Version: 3.0.0~rc4-1~experimental.1 > > Severity: normal > > Tags: upstream > > > > > > Hi, > > > > when using this kernel, I hear no sound using the speakers, only using > > the headphones, plugged in at the front. > > > > With 2.6.38 (dmesg attached), the speakers get fed, too. > [...] > > 00:14.2 Audio device [0403]: ATI Technologies Inc SBx00 Azalia (Intel HDA) [1002:4383] > > Subsystem: ASUSTeK Computer Inc. Device [1043:841b] > > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- > > Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > > Latency: 64, Cache Line Size: 64 bytes > > Interrupt: pin A routed to IRQ 16 > > Region 0: Memory at fe8f0000 (64-bit, non-prefetchable) [size=16K] > > Capabilities: <access denied> > > Kernel driver in use: HDA Intel > [...] > > Takashi, this looks like a regression in control parsing in the realtek > codec. The kernel logs are at <http://bugs.debian.org/631963> but I > don't think they'll tell you much. I'd definitely need alsa-info.sh outputs on both working and non-working kernels, i.e. on 2.6.38 and 3.0-rc4. > Toni, can you provide the output of 'amixer -c 0 controls' under 2.6.38 > and 3.0-rc4? alsa-info.sh outputs contain already these. Toni, could you give alsa-info.sh outputs of the above kernels? thanks, Takashi

1 0

[alsa-devel] compile problems
by David Henderson 28 Jun '11

28 Jun '11

Greetings all! I've tried this question in the "alsa user" group without success, but was pointed in the direction of this mailing list. Currently I'm trying to compile the alsa packages to get it compiled for a custom Linux distro. I'm not having any problems compiling the alsa-lib package, but when I try to compile alsa-utils, I keep getting the following error: checking for libasound headers version>= 1.0.16... not present. configure: error: Sufficiently new version of libasound not found. To provide more details about the situation, I'm using an existing distro (Kubuntu 9.10) to build the alsa packages and are using the /opt/staging/alsa directory to compile and "install" to (using the DESTDIR parameter). Seeing as how this is a staging directory and not the actual place the package is being installed to, I'm having the afore mentioned problem because the alsa-utils package is looking under /... for the header files instead of the /opt/staging/alsa/... directory. Is there a compile-time parameter that I can use so that alsa-utils looks in the staging directory for the header files - just during the compile phase? Thanks, Dave

4 17

[alsa-devel] Mixer control for signed gain value
by Patrick Lai 28 Jun '11

28 Jun '11

Hi, The CODEC I am working with takes byte-long gain value in two's complement form. In soc-core.c, the closest functions I see are snd_soc_info_volsw_s8 snd_soc_get_volsw_s8 snd_soc_set_volsw_s8 These functions are coded to work with two integer values. For my purpose, I suppose that I can always set 2nd integer input as zero. However, would I still be able to set volume like amixer sset "Codec Gain" 100 as percentage. If yes, how will amixer know to put upper 8 bits of 16 bit gain in 2nd integer value element? Thanks Patrick -- Sent by an employee of the Qualcomm Innovation Center, Inc. The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum.

2 3

[alsa-devel] [PATCH v3 1/3] ASoC: Add ADAV80x codec driver
by Lars-Peter Clausen 28 Jun '11

28 Jun '11

This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v2: * Allow to enable/disable SYSCLK outputs by using snd_soc_dai_set_sysclk Changes since v1: * Sync register on resume * Don't use arrays for the mux enums and controls * Rework PLL handling. Use DAPM for powering the PLL and OSC down and use set_sysclk to select the system clock instead of doing this implicitly in set_pll. --- MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 951 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adav80x.h | 35 ++ 5 files changed, 993 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/adav80x.c create mode 100644 sound/soc/codecs/adav80x.h diff --git a/MAINTAINERS b/MAINTAINERS index b3b3505..943f81e 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -534,6 +534,7 @@ L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ S: Supported F: sound/soc/codecs/adau* +F: sound/soc/codecs/adav* F: sound/soc/codecs/ad1* F: sound/soc/codecs/ssm* diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2998e65..ff43405 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -137,6 +138,9 @@ config SND_SOC_ADAU1701 select SIGMA tristate +config SND_SOC_ADAV80X + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 51cd3d4..4957431 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o +snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -99,6 +100,7 @@ obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c new file mode 100644 index 0000000..e30fba6 --- /dev/null +++ b/sound/soc/codecs/adav80x.c @@ -0,0 +1,951 @@ +/* + * ADAV80X Audio Codec driver supporting ADAV801, ADAV803 + * + * Copyright 2011 Analog Devices Inc. + * Author: Yi Li <yi.li(a)analog.com> + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> + +#include "adav80x.h" + +#define ADAV80X_PLAYBACK_CTRL 0x04 +#define ADAV80X_AUX_IN_CTRL 0x05 +#define ADAV80X_REC_CTRL 0x06 +#define ADAV80X_AUX_OUT_CTRL 0x07 +#define ADAV80X_DPATH_CTRL1 0x62 +#define ADAV80X_DPATH_CTRL2 0x63 +#define ADAV80X_DAC_CTRL1 0x64 +#define ADAV80X_DAC_CTRL2 0x65 +#define ADAV80X_DAC_CTRL3 0x66 +#define ADAV80X_DAC_L_VOL 0x68 +#define ADAV80X_DAC_R_VOL 0x69 +#define ADAV80X_PGA_L_VOL 0x6c +#define ADAV80X_PGA_R_VOL 0x6d +#define ADAV80X_ADC_CTRL1 0x6e +#define ADAV80X_ADC_CTRL2 0x6f +#define ADAV80X_ADC_L_VOL 0x70 +#define ADAV80X_ADC_R_VOL 0x71 +#define ADAV80X_PLL_CTRL1 0x74 +#define ADAV80X_PLL_CTRL2 0x75 +#define ADAV80X_ICLK_CTRL1 0x76 +#define ADAV80X_ICLK_CTRL2 0x77 +#define ADAV80X_PLL_CLK_SRC 0x78 +#define ADAV80X_PLL_OUTE 0x7a + +#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll) 0x00 +#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll) (0x40 << (pll)) +#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll) (0x40 << (pll)) + +#define ADAV80X_ICLK_CTRL1_DAC_SRC(src) ((src) << 5) +#define ADAV80X_ICLK_CTRL1_ADC_SRC(src) ((src) << 2) +#define ADAV80X_ICLK_CTRL1_ICLK2_SRC(src) (src) +#define ADAV80X_ICLK_CTRL2_ICLK1_SRC(src) ((src) << 3) + +#define ADAV80X_PLL_CTRL1_PLLDIV 0x10 +#define ADAV80X_PLL_CTRL1_PLLPD(pll) (0x04 << (pll)) +#define ADAV80X_PLL_CTRL1_XTLPD 0x02 + +#define ADAV80X_PLL_CTRL2_FIELD(pll, x) ((x) << ((pll) * 4)) + +#define ADAV80X_PLL_CTRL2_FS_48(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x00) +#define ADAV80X_PLL_CTRL2_FS_32(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x08) +#define ADAV80X_PLL_CTRL2_FS_44(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c) + +#define ADAV80X_PLL_CTRL2_SEL(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x02) +#define ADAV80X_PLL_CTRL2_DOUB(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x01) +#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f) + +#define ADAV80X_ADC_CTRL1_MODULATOR_MASK 0x80 +#define ADAV80X_ADC_CTRL1_MODULATOR_128FS 0x00 +#define ADAV80X_ADC_CTRL1_MODULATOR_64FS 0x80 + +#define ADAV80X_DAC_CTRL1_PD 0x80 + +#define ADAV80X_DAC_CTRL2_DIV1 0x00 +#define ADAV80X_DAC_CTRL2_DIV1_5 0x10 +#define ADAV80X_DAC_CTRL2_DIV2 0x20 +#define ADAV80X_DAC_CTRL2_DIV3 0x30 +#define ADAV80X_DAC_CTRL2_DIV_MASK 0x30 + +#define ADAV80X_DAC_CTRL2_INTERPOL_256FS 0x00 +#define ADAV80X_DAC_CTRL2_INTERPOL_128FS 0x40 +#define ADAV80X_DAC_CTRL2_INTERPOL_64FS 0x80 +#define ADAV80X_DAC_CTRL2_INTERPOL_MASK 0xc0 + +#define ADAV80X_DAC_CTRL2_DEEMPH_NONE 0x00 +#define ADAV80X_DAC_CTRL2_DEEMPH_44 0x01 +#define ADAV80X_DAC_CTRL2_DEEMPH_32 0x02 +#define ADAV80X_DAC_CTRL2_DEEMPH_48 0x03 +#define ADAV80X_DAC_CTRL2_DEEMPH_MASK 0x01 + +#define ADAV80X_CAPTURE_MODE_MASTER 0x20 +#define ADAV80X_CAPTURE_WORD_LEN24 0x00 +#define ADAV80X_CAPTURE_WORD_LEN20 0x04 +#define ADAV80X_CAPTRUE_WORD_LEN18 0x08 +#define ADAV80X_CAPTURE_WORD_LEN16 0x0c +#define ADAV80X_CAPTURE_WORD_LEN_MASK 0x0c + +#define ADAV80X_CAPTURE_MODE_LEFT_J 0x00 +#define ADAV80X_CAPTURE_MODE_I2S 0x01 +#define ADAV80X_CAPTURE_MODE_RIGHT_J 0x03 +#define ADAV80X_CAPTURE_MODE_MASK 0x03 + +#define ADAV80X_PLAYBACK_MODE_MASTER 0x10 +#define ADAV80X_PLAYBACK_MODE_LEFT_J 0x00 +#define ADAV80X_PLAYBACK_MODE_I2S 0x01 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24 0x04 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20 0x05 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18 0x06 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16 0x07 +#define ADAV80X_PLAYBACK_MODE_MASK 0x07 + +#define ADAV80X_PLL_OUTE_SYSCLKPD(x) BIT(2 - (x)) + +static u8 adav80x_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, + 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, + 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, + 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, + 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, + 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, + 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +}; + +struct adav80x { + enum snd_soc_control_type control_type; + + enum adav80x_clk_src clk_src; + unsigned int sysclk; + enum adav80x_pll_src pll_src; + + unsigned int dai_fmt[2]; + unsigned int rate; + bool deemph; + bool sysclk_pd[3]; +}; + +static const char *adav80x_mux_text[] = { + "ADC", + "Playback", + "Aux Playback", +}; + +static const unsigned int adav80x_mux_values[] = { + 0, 2, 3, +}; + +#define ADAV80X_MUX_ENUM_DECL(name, reg, shift) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, reg, shift, 7, \ + ARRAY_SIZE(adav80x_mux_text), adav80x_mux_text, \ + adav80x_mux_values) + +static ADAV80X_MUX_ENUM_DECL(adav80x_aux_capture_enum, ADAV80X_DPATH_CTRL1, 0); +static ADAV80X_MUX_ENUM_DECL(adav80x_capture_enum, ADAV80X_DPATH_CTRL1, 3); +static ADAV80X_MUX_ENUM_DECL(adav80x_dac_enum, ADAV80X_DPATH_CTRL2, 3); + +static const struct snd_kcontrol_new adav80x_aux_capture_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_aux_capture_enum); +static const struct snd_kcontrol_new adav80x_capture_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_capture_enum); +static const struct snd_kcontrol_new adav80x_dac_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_dac_enum); + +#define ADAV80X_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1), + SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1), + + SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0), + + ADAV80X_MUX("Aux Capture Select", &adav80x_aux_capture_mux_ctrl), + ADAV80X_MUX("Capture Select", &adav80x_capture_mux_ctrl), + ADAV80X_MUX("DAC Select", &adav80x_dac_mux_ctrl), + + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + + SND_SOC_DAPM_SUPPLY("SYSCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", ADAV80X_PLL_CTRL1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2", ADAV80X_PLL_CTRL1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("OSC", ADAV80X_PLL_CTRL1, 1, 1, NULL, 0), +}; + +static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + const char *clk; + + switch (adav80x->clk_src) { + case ADAV80X_CLK_PLL1: + clk = "PLL1"; + break; + case ADAV80X_CLK_PLL2: + clk = "PLL2"; + break; + case ADAV80X_CLK_XTAL: + clk = "OSC"; + break; + default: + return 0; + } + + return strcmp(source->name, clk) == 0; +} + +static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL; +} + + +static const struct snd_soc_dapm_route adav80x_dapm_routes[] = { + { "DAC Select", "ADC", "ADC" }, + { "DAC Select", "Playback", "AIFIN" }, + { "DAC Select", "Aux Playback", "AIFAUXIN" }, + { "DAC", NULL, "DAC Select" }, + + { "Capture Select", "ADC", "ADC" }, + { "Capture Select", "Playback", "AIFIN" }, + { "Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFOUT", NULL, "Capture Select" }, + + { "Aux Capture Select", "ADC", "ADC" }, + { "Aux Capture Select", "Playback", "AIFIN" }, + { "Aux Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFAUXOUT", NULL, "Aux Capture Select" }, + + { "VOUTR", NULL, "DAC" }, + { "VOUTL", NULL, "DAC" }, + + { "Left PGA", NULL, "VINL" }, + { "Right PGA", NULL, "VINR" }, + { "ADC", NULL, "Left PGA" }, + { "ADC", NULL, "Right PGA" }, + + { "SYSCLK", NULL, "PLL1", adav80x_dapm_sysclk_check }, + { "SYSCLK", NULL, "PLL2", adav80x_dapm_sysclk_check }, + { "SYSCLK", NULL, "OSC", adav80x_dapm_sysclk_check }, + { "PLL1", NULL, "OSC", adav80x_dapm_pll_check }, + { "PLL2", NULL, "OSC", adav80x_dapm_pll_check }, + + { "ADC", NULL, "SYSCLK" }, + { "DAC", NULL, "SYSCLK" }, + { "AIFOUT", NULL, "SYSCLK" }, + { "AIFAUXOUT", NULL, "SYSCLK" }, + { "AIFIN", NULL, "SYSCLK" }, + { "AIFAUXIN", NULL, "SYSCLK" }, +}; + +static int adav80x_set_deemph(struct snd_soc_codec *codec) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->deemph) { + switch (adav80x->rate) { + case 32000: + val = ADAV80X_DAC_CTRL2_DEEMPH_32; + break; + case 44100: + val = ADAV80X_DAC_CTRL2_DEEMPH_44; + break; + case 48000: + case 64000: + case 88200: + case 96000: + val = ADAV80X_DAC_CTRL2_DEEMPH_48; + break; + default: + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + break; + } + } else { + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + } + + return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); +} + +static int adav80x_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + adav80x->deemph = deemph; + + return adav80x_set_deemph(codec); +} + +static int adav80x_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = adav80x->deemph; + return 0; +}; + +static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adav80x_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL, + ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL, + ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL, + ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv), + + SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0), + SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0), + + SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0, + adav80x_get_deemph, adav80x_put_deemph), +}; + +static unsigned int adav80x_port_ctrl_regs[2][2] = { + { ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, }, + { ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL }, +}; + +static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int capture = 0x00; + unsigned int playback = 0x00; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + capture |= ADAV80X_CAPTURE_MODE_MASTER; + playback |= ADAV80X_PLAYBACK_MODE_MASTER; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + capture |= ADAV80X_CAPTURE_MODE_I2S; + playback |= ADAV80X_PLAYBACK_MODE_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + capture |= ADAV80X_CAPTURE_MODE_LEFT_J; + playback |= ADAV80X_PLAYBACK_MODE_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + capture |= ADAV80X_CAPTURE_MODE_RIGHT_J; + playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, + capture); + snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + + adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adav80x_set_adc_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_ADC_CTRL1_MODULATOR_128FS; + else + val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; + + snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); + + return 0; +} + +static int adav80x_set_dac_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS; + else + val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; + + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, + val); + + return 0; +} + +static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_CAPTURE_WORD_LEN16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_CAPTRUE_WORD_LEN18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_CAPTURE_WORD_LEN20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_CAPTURE_WORD_LEN24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_WORD_LEN_MASK, val); + + return 0; +} + +static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + ADAV80X_PLAYBACK_MODE_MASK, val); + + return 0; +} + +static int adav80x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + + if (rate * 256 != adav80x->sysclk) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + adav80x_set_playback_pcm_format(codec, dai, + params_format(params)); + adav80x_set_dac_clock(codec, rate); + } else { + adav80x_set_capture_pcm_format(codec, dai, + params_format(params)); + adav80x_set_adc_clock(codec, rate); + } + adav80x->rate = rate; + adav80x_set_deemph(codec); + + return 0; +} + +static int adav80x_set_sysclk(struct snd_soc_codec *codec, + int clk_id, unsigned int freq, int dir) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (dir == SND_SOC_CLOCK_IN) { + switch (clk_id) { + case ADAV80X_CLK_XIN: + case ADAV80X_CLK_XTAL: + case ADAV80X_CLK_MCLKI: + case ADAV80X_CLK_PLL1: + case ADAV80X_CLK_PLL2: + break; + default: + return -EINVAL; + } + + adav80x->sysclk = freq; + + if (adav80x->clk_src != clk_id) { + unsigned int iclk_ctrl1, iclk_ctrl2; + + adav80x->clk_src = clk_id; + if (clk_id == ADAV80X_CLK_XTAL) + clk_id = ADAV80X_CLK_XIN; + + iclk_ctrl1 = ADAV80X_ICLK_CTRL1_DAC_SRC(clk_id) | + ADAV80X_ICLK_CTRL1_ADC_SRC(clk_id) | + ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); + iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); + + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + + snd_soc_dapm_sync(&codec->dapm); + } + } else { + unsigned int mask; + + switch (clk_id) { + case ADAV80X_CLK_SYSCLK1: + case ADAV80X_CLK_SYSCLK2: + case ADAV80X_CLK_SYSCLK3: + break; + default: + return -EINVAL; + } + + clk_id -= ADAV80X_CLK_SYSCLK1; + mask = ADAV80X_PLL_OUTE_SYSCLKPD(clk_id); + + if (freq == 0) { + snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, mask); + adav80x->sysclk_pd[clk_id] = true; + } else { + snd_soc_update_bits(codec, ADAV80X_PLL_OUTE, mask, 0); + adav80x->sysclk_pd[clk_id] = false; + } + + if (adav80x->sysclk_pd[0]) + snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); + else + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + + if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2]) + snd_soc_dapm_disable_pin(&codec->dapm, "PLL2"); + else + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); + + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int pll_ctrl1 = 0; + unsigned int pll_ctrl2 = 0; + unsigned int pll_src; + + switch (source) { + case ADAV80X_PLL_SRC_XTAL: + case ADAV80X_PLL_SRC_XIN: + case ADAV80X_PLL_SRC_MCLKI: + break; + default: + return -EINVAL; + } + + if (!freq_out) + return 0; + + switch (freq_in) { + case 27000000: + break; + case 54000000: + if (source == ADAV80X_PLL_SRC_XIN) { + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; + break; + } + default: + return -EINVAL; + } + + if (freq_out > 12288000) { + pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll_id); + freq_out /= 2; + } + + /* freq_out = sample_rate * 256 */ + switch (freq_out) { + case 8192000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll_id); + break; + case 11289600: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll_id); + break; + case 12288000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll_id); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, + pll_ctrl1); + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); + + if (source != adav80x->pll_src) { + if (source == ADAV80X_PLL_SRC_MCLKI) + pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll_id); + else + pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); + + snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); + + adav80x->pll_src = source; + + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int adav80x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAV80X_DAC_CTRL1_PD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +/* Enforce the same sample rate on all audio interfaces */ +static int adav80x_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active || !adav80x->rate) + return 0; + + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate); +} + +static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active) + adav80x->rate = 0; +} + +static const struct snd_soc_dai_ops adav80x_dai_ops = { + .set_fmt = adav80x_set_dai_fmt, + .hw_params = adav80x_hw_params, + .startup = adav80x_dai_startup, + .shutdown = adav80x_dai_shutdown, +}; + +#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver adav80x_dais[] = { + { + .name = "adav80x-hifi", + .id = 0, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, + { + .name = "adav80x-aux", + .id = 1, + .playback = { + .stream_name = "Aux Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "Aux Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, +}; + +static int adav80x_probe(struct snd_soc_codec *codec) +{ + int ret; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Force PLLs on for SYSCLK output */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); + + /* Power down S/PDIF receiver, since it is currently not supported */ + snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x20); + /* Disable DAC zero flag */ + snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +} + +static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adav80x_resume(struct snd_soc_codec *codec) +{ + adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->cache_sync = 1; + snd_soc_cache_sync(codec); + + return 0; +} + +static int adav80x_remove(struct snd_soc_codec *codec) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static struct snd_soc_codec_driver adav80x_codec_driver = { + .probe = adav80x_probe, + .remove = adav80x_remove, + .suspend = adav80x_suspend, + .resume = adav80x_resume, + .set_bias_level = adav80x_set_bias_level, + + .set_pll = adav80x_set_pll, + .set_sysclk = adav80x_set_sysclk, + + .reg_word_size = sizeof(u8), + .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), + .reg_cache_default = adav80x_default_regs, + + .controls = adav80x_controls, + .num_controls = ARRAY_SIZE(adav80x_controls), + .dapm_widgets = adav80x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets), + .dapm_routes = adav80x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), +}; + +static int __devinit adav80x_bus_probe(struct device *dev, + enum snd_soc_control_type control_type) +{ + struct adav80x *adav80x; + int ret; + + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + dev_set_drvdata(dev, adav80x); + adav80x->control_type = control_type; + + ret = snd_soc_register_codec(dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); + if (ret) + kfree(adav80x); + + return ret; +} + +static int __devexit adav80x_bus_remove(struct device *dev) +{ + snd_soc_unregister_codec(dev); + kfree(dev_get_drvdata(dev)); + return 0; +} + +#if defined(CONFIG_SPI_MASTER) +static int __devinit adav80x_spi_probe(struct spi_device *spi) +{ + return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); +} + +static int __devexit adav80x_spi_remove(struct spi_device *spi) +{ + return adav80x_bus_remove(&spi->dev); +} + +static struct spi_driver adav80x_spi_driver = { + .driver = { + .name = "adav801", + .owner = THIS_MODULE, + }, + .probe = adav80x_spi_probe, + .remove = __devexit_p(adav80x_spi_remove), +}; +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id adav80x_id[] = { + { "adav803", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adav80x_id); + +static int __devinit adav80x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return adav80x_bus_probe(&client->dev, SND_SOC_I2C); +} + +static int __devexit adav80x_i2c_remove(struct i2c_client *client) +{ + return adav80x_bus_remove(&client->dev); +} + +static struct i2c_driver adav80x_i2c_driver = { + .driver = { + .name = "adav803", + .owner = THIS_MODULE, + }, + .probe = adav80x_i2c_probe, + .remove = __devexit_p(adav80x_i2c_remove), + .id_table = adav80x_id, +}; +#endif + +static int __init adav80x_init(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&adav80x_i2c_driver); + if (ret) + return ret; +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&adav80x_spi_driver); +#endif + + return ret; +} +module_init(adav80x_init); + +static void __exit adav80x_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&adav80x_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&adav80x_spi_driver); +#endif +} +module_exit(adav80x_exit); + +MODULE_DESCRIPTION("ASoC ADAV80x driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_AUTHOR("Yi Li <yi.li(a)analog.com>>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h new file mode 100644 index 0000000..adb0fc7 --- /dev/null +++ b/sound/soc/codecs/adav80x.h @@ -0,0 +1,35 @@ +/* + * header file for ADAV80X parts + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAV80X_H +#define _ADAV80X_H + +enum adav80x_pll_src { + ADAV80X_PLL_SRC_XIN, + ADAV80X_PLL_SRC_XTAL, + ADAV80X_PLL_SRC_MCLKI, +}; + +enum adav80x_pll { + ADAV80X_PLL1 = 0, + ADAV80X_PLL2 = 1, +}; + +enum adav80x_clk_src { + ADAV80X_CLK_XIN = 0, + ADAV80X_CLK_MCLKI = 1, + ADAV80X_CLK_PLL1 = 2, + ADAV80X_CLK_PLL2 = 3, + ADAV80X_CLK_XTAL = 6, + + ADAV80X_CLK_SYSCLK1 = 6, + ADAV80X_CLK_SYSCLK2 = 7, + ADAV80X_CLK_SYSCLK3 = 8, +}; + +#endif -- 1.7.2.5

4 8

[alsa-devel] [PATCH] alsa: update author email for ABDAC, AC97C and AT73C213
by Hans-Christian Egtvedt 28 Jun '11

28 Jun '11

This patch updates the email address of the sound drivers supported by me to an email account I will use on a more regular basis in the future. Signed-off-by: Hans-Christian Egtvedt <hans-christian.egtvedt(a)atmel.com> --- sound/atmel/abdac.c | 2 +- sound/atmel/ac97c.c | 2 +- sound/spi/at73c213.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 6e24091..bfee60c 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -599,4 +599,4 @@ module_exit(atmel_abdac_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Driver for Atmel Audio Bitstream DAC (ABDAC)"); -MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt(a)atmel.com>"); +MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt(a)samfundet.no>"); diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index b310702..ac35222 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -1199,4 +1199,4 @@ module_exit(atmel_ac97c_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Driver for Atmel AC97 controller"); -MODULE_AUTHOR("Hans-Christian Egtvedt <hans-christian.egtvedt(a)atmel.com>"); +MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt(a)samfundet.no>"); diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c index 337a002..4dd051b 100644 --- a/sound/spi/at73c213.c +++ b/sound/spi/at73c213.c @@ -1124,6 +1124,6 @@ static void __exit at73c213_exit(void) } module_exit(at73c213_exit); -MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt(a)atmel.com>"); +MODULE_AUTHOR("Hans-Christian Egtvedt <egtvedt(a)samfundet.no>"); MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); MODULE_LICENSE("GPL"); -- 1.7.4.1

2 1

[PATCH - resample 1/1] Using float instead of s16 for libsamplerate
by Pavel Hofman 28 Jun '11

28 Jun '11

Natively libsamplerate operates in floats. Using the new convert_float API instead of dropping resoluton in convert_s16. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/rate/rate_samplerate.c b/rate/rate_samplerate.c index 53a0627..c612983 100644 --- a/rate/rate_samplerate.c +++ b/rate/rate_samplerate.c @@ -23,12 +23,12 @@ #include <alsa/asoundlib.h> #include <alsa/pcm_rate.h> +typedef float float_t; + struct rate_src { double ratio; int converter; unsigned int channels; - float *src_buf; - float *dst_buf; SRC_STATE *state; SRC_DATA data; }; @@ -53,9 +53,6 @@ static void pcm_src_free(void *obj) { struct rate_src *rate = obj; - free(rate->src_buf); - free(rate->dst_buf); - rate->src_buf = rate->dst_buf = NULL; if (rate->state) { src_delete(rate->state); @@ -79,17 +76,6 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info) rate->ratio = (double)info->out.rate / (double)info->in.rate; - free(rate->src_buf); - rate->src_buf = malloc(sizeof(float) * rate->channels * info->in.period_size); - free(rate->dst_buf); - rate->dst_buf = malloc(sizeof(float) * rate->channels * info->out.period_size); - if (! rate->src_buf || ! rate->dst_buf) { - pcm_src_free(rate); - return -ENOMEM; - } - - rate->data.data_in = rate->src_buf; - rate->data.data_out = rate->dst_buf; rate->data.src_ratio = rate->ratio; rate->data.end_of_input = 0; @@ -112,8 +98,8 @@ static void pcm_src_reset(void *obj) src_reset(rate->state); } -static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames, - const int16_t *src, unsigned int src_frames) +static void pcm_src_convert_float(void *obj, float_t *dst, unsigned int dst_frames, + const float_t *src, unsigned int src_frames) { struct rate_src *rate = obj; unsigned int ofs; @@ -121,15 +107,17 @@ static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int dst_frames rate->data.input_frames = src_frames; rate->data.output_frames = dst_frames; rate->data.end_of_input = 0; + rate->data.data_in = src; + rate->data.data_out = dst; + - src_short_to_float_array(src, rate->src_buf, src_frames * rate->channels); src_process(rate->state, &rate->data); if (rate->data.output_frames_gen < dst_frames) ofs = dst_frames - rate->data.output_frames_gen; else ofs = 0; - src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels, - rate->data.output_frames_gen * rate->channels); + /*src_float_to_short_array(rate->dst_buf, dst + ofs * rate->channels, + rate->data.output_frames_gen * rate->channels); */ } static void pcm_src_close(void *obj) @@ -157,7 +145,7 @@ static snd_pcm_rate_ops_t pcm_src_ops = { .free = pcm_src_free, .reset = pcm_src_reset, .adjust_pitch = pcm_src_adjust_pitch, - .convert_s16 = pcm_src_convert_s16, + .convert_float = pcm_src_convert_float, .input_frames = input_frames, .output_frames = output_frames, #if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002 -- 1.7.0.4 --------------000009060503020103050906--

1 0

[PATCH - rate plugin 1/1] Support for float samples in rate converters
by Pavel Hofman 28 Jun '11

28 Jun '11

Some rate converters have native float resolution, no need to loose information by converting to s16. Signed-off-by: Pavel Hofman <pavel.hofman(a)ivitera.com> diff --git a/include/pcm_rate.h b/include/pcm_rate.h index 4d70df2..e92900b 100644 --- a/include/pcm_rate.h +++ b/include/pcm_rate.h @@ -91,6 +91,11 @@ typedef struct snd_pcm_rate_ops { void (*convert_s16)(void *obj, int16_t *dst, unsigned int dst_frames, const int16_t *src, unsigned int src_frames); /** + * convert a float interleaved-data array; exclusive with convert + */ + void (*convert_float)(void *obj, float *dst, unsigned int dst_frames, + const float *src, unsigned int src_frames); + /** * compute the frame size for input */ snd_pcm_uframes_t (*input_frames)(void *obj, snd_pcm_uframes_t frames); diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 70e30e5..9f4ec23 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -35,7 +35,7 @@ #include "iatomic.h" #include "plugin_ops.h" - +typedef float float_t; #if 0 #define DEBUG_REFINE #endif @@ -66,8 +66,8 @@ struct _snd_pcm_rate { snd_pcm_rate_ops_t ops; unsigned int get_idx; unsigned int put_idx; - int16_t *src_buf; - int16_t *dst_buf; + void *src_buf; + void *dst_buf; int start_pending; /* start is triggered but not commited to slave */ snd_htimestamp_t trigger_tstamp; unsigned int plugin_version; @@ -310,13 +310,23 @@ static int snd_pcm_rate_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) rate->sareas[chn].step = swidth; } - if (rate->ops.convert_s16) { + if (rate->ops.convert_float) { + rate->get_idx = snd_pcm_linear_get_index(rate->info.in.format, SND_PCM_FORMAT_S32); + rate->put_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S32, rate->info.out.format); + free(rate->src_buf); + rate->src_buf = (float_t *) malloc(channels * rate->info.in.period_size * sizeof(float_t)); + free(rate->dst_buf); + rate->dst_buf = (float_t *) malloc(channels * rate->info.out.period_size * sizeof(float_t)); + if (! rate->src_buf || ! rate->dst_buf) + goto error; + } + else if (rate->ops.convert_s16) { rate->get_idx = snd_pcm_linear_get_index(rate->info.in.format, SND_PCM_FORMAT_S16); rate->put_idx = snd_pcm_linear_put_index(SND_PCM_FORMAT_S16, rate->info.out.format); free(rate->src_buf); - rate->src_buf = malloc(channels * rate->info.in.period_size * 2); + rate->src_buf = (int16_t *) malloc(channels * rate->info.in.period_size * 2); free(rate->dst_buf); - rate->dst_buf = malloc(channels * rate->info.out.period_size * 2); + rate->dst_buf = (int16_t *) malloc(channels * rate->info.out.period_size * 2); if (! rate->src_buf || ! rate->dst_buf) goto error; } @@ -503,6 +513,85 @@ static void convert_from_s16(snd_pcm_rate_t *rate, const int16_t *buf, } } } +static void convert_to_float(snd_pcm_rate_t *rate, float_t *buf, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, unsigned int frames, + unsigned int channels) +{ +#ifndef DOC_HIDDEN +#define GET32_LABELS +#include "plugin_ops.h" +#undef GET32_LABELS +#endif /* DOC_HIDDEN */ + void *get32 = get32_labels[rate->get_idx]; + const char *src; + int32_t sample; + const char *srcs[channels]; + int src_step[channels]; + unsigned int c; + + for (c = 0; c < channels; c++) { + srcs[c] = snd_pcm_channel_area_addr(areas + c, offset); + src_step[c] = snd_pcm_channel_area_step(areas + c); + } + + while (frames--) { + for (c = 0; c < channels; c++) { + src = srcs[c]; + // converting the source format to int32 first + goto *get32; +#ifndef DOC_HIDDEN +#define GET32_END after_get +#include "plugin_ops.h" +#undef GET32_END +#endif /* DOC_HIDDEN */ + after_get: + // converting to float for the plugin + *buf++ = (float_t) sample; + srcs[c] += src_step[c]; + } + } +} + +static void convert_from_float(snd_pcm_rate_t *rate, const float_t *buf, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, unsigned int frames, + unsigned int channels) +{ +#ifndef DOC_HIDDEN +#define PUT32_LABELS +#include "plugin_ops.h" +#undef PUT32_LABELS +#endif /* DOC_HIDDEN */ + void *put32 = put32_labels[rate->put_idx]; + char *dst; + int32_t sample; + char *dsts[channels]; + int dst_step[channels]; + unsigned int c; + + for (c = 0; c < channels; c++) { + dsts[c] = snd_pcm_channel_area_addr(areas + c, offset); + dst_step[c] = snd_pcm_channel_area_step(areas + c); + } + + while (frames--) { + for (c = 0; c < channels; c++) { + dst = dsts[c]; + // plugin returned float, converting to int32 first + sample = (int32_t) *buf++; + // converting int32 to the destination format + goto *put32; +#ifndef DOC_HIDDEN +#define PUT32_END after_put +#include "plugin_ops.h" +#undef PUT32_END +#endif /* DOC_HIDDEN */ + after_put: + dsts[c] += dst_step[c]; + } + } +} static void do_convert(const snd_pcm_channel_area_t *dst_areas, snd_pcm_uframes_t dst_offset, unsigned int dst_frames, @@ -511,18 +600,36 @@ static void do_convert(const snd_pcm_channel_area_t *dst_areas, unsigned int channels, snd_pcm_rate_t *rate) { - if (rate->ops.convert_s16) { + if (rate->ops.convert_float) { + const float_t *src; + float_t *dst; + if (! rate->src_buf) + src = src_areas->addr + src_offset * sizeof(float_t) * channels; + else { + convert_to_float(rate, rate->src_buf, src_areas, src_offset, + src_frames, channels); + src = rate->src_buf; + } + if (! rate->dst_buf) + dst = dst_areas->addr + dst_offset * sizeof(float_t) * channels; + else + dst = rate->dst_buf; + rate->ops.convert_float(rate->obj, dst, dst_frames, src, src_frames); + if (dst == rate->dst_buf) + convert_from_float(rate, rate->dst_buf, dst_areas, dst_offset, + dst_frames, channels); + } else if (rate->ops.convert_s16) { const int16_t *src; int16_t *dst; if (! rate->src_buf) - src = src_areas->addr + src_offset * 2 * channels; + src = src_areas->addr + src_offset * sizeof(int16_t) * channels; else { convert_to_s16(rate, rate->src_buf, src_areas, src_offset, src_frames, channels); src = rate->src_buf; } if (! rate->dst_buf) - dst = dst_areas->addr + dst_offset * 2 * channels; + dst = dst_areas->addr + dst_offset * sizeof(int16_t) * channels; else dst = rate->dst_buf; rate->ops.convert_s16(rate->obj, dst, dst_frames, src, src_frames); @@ -1409,7 +1516,7 @@ int snd_pcm_rate_open(snd_pcm_t **pcmp, const char *name, } #endif - if (! rate->ops.init || ! (rate->ops.convert || rate->ops.convert_s16) || + if (! rate->ops.init || ! (rate->ops.convert || rate->ops.convert_s16 || rate->ops.convert_float) || ! rate->ops.input_frames || ! rate->ops.output_frames) { SNDERR("Inproper rate plugin %s initialization", type); snd_pcm_close(pcm); -- 1.7.0.4 --------------000009060503020103050906 Content-Type: text/x-diff; name="0001-Using-float-instead-of-s16-for-libsamplerate.patch" Content-Transfer-Encoding: 7bit Content-Disposition: attachment; filename*0="0001-Using-float-instead-of-s16-for-libsamplerate.patch"

1 0

Re: [alsa-devel] SBx00 Azalia ALC663 Working but MIC goes away
by Takashi Iwai 28 Jun '11

28 Jun '11

At Sun, 26 Jun 2011 20:38:43 -0300, Diego Jacobi wrote: > > Hi people. > > I am having this problem for a long time ago, and i couldn't find the > same on google, or a solution for it. > I also don't know the right way to write the question to google, > because almost all are about problems that i don't have. > > OpenSUSE 11.4 installed about 2 Months ago. Did you read my post yesterday? Already tried alsa-driver-kmp? Takashi

1 0

[alsa-devel] Realtek ALC662 rev1 no sound over HDMI
by crow 28 Jun '11

28 Jun '11

I am trying to get sound working on kernel 2.6.39 (under archlinux x86_64 bit distribution) over HDMI to my digital receiver but without luck. Before this post i asked in #alsa for help, and everything we tried over there didnt get the sound over HDMI. I connected headphones to the green jack and ran speaker-test and there is sound over jack, analog sound plays fine. But runing alsamixer un-muting spdif/iec958/spdif (1) and trying to re run speaker-test doesn't output sound over HDMI to my digital receiver. I tried to use /etc/asound.conf for global settings (tooked from 2.6.35 archlinux i686 installation) but does not work with 2.6.39 installation, even I am not able to play analog sound: speaker-test -c2 -twav -l1 This above works without asound.conf, and it plays analog sound on headphones. This is how i tried to play sound over HDMI: speaker-test -c2 -D plughw:0,3 -twav -l1 The above command does output sound on 2.6.35 (but just over Fron Right) I run alsa-info.sh script on both installation and it can be found on (2) kernel 2.6.39 and (3) kernel 2.6.35. I was hoping to get sound working with 2.6.39 (current official archlinux kernel) over HDMI by just un-muting spdif with alsamixer, but it seems not to be that easy. (1) http://www.abload.de/img/alsa01ji6j.png (2) http://www.alsa-project.org/db/?f=7bf61ad34a4bb51b615773aab69ae904693a07d7 (3) http://www.alsa-project.org/db/?f=3694c9052c2b8f5d78f8a9e2a465a7c94885449b

3 6

[alsa-devel] [PATCH V3 1/4] ASoC: SAMSUNG: Move I2S common register definition
by Sangbeom Kim 26 Jun '11

26 Jun '11

I2S registers can be used for control idma. Previously, register is defined in i2s.c. For sharing the registers, It is moved to i2s-regs.h Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/i2s-regs.h | 123 ++++++++++++++++++++++++++++++++++++++++++ sound/soc/samsung/i2s.c | 104 +----------------------------------- 2 files changed, 124 insertions(+), 103 deletions(-) create mode 100644 sound/soc/samsung/i2s-regs.h diff --git a/sound/soc/samsung/i2s-regs.h b/sound/soc/samsung/i2s-regs.h new file mode 100644 index 0000000..1fa0cd1 --- /dev/null +++ b/sound/soc/samsung/i2s-regs.h @@ -0,0 +1,123 @@ +/* + * linux/sound/soc/samsung/i2s-regs.h + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd. + * http://www.samsung.com + * + * Samsung I2S driver's register header + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ + +#ifndef __SND_SOC_SAMSUNG_I2S_REGS_H +#define __SND_SOC_SAMSUNG_I2S_REGS_H + +#define I2SCON 0x0 +#define I2SMOD 0x4 +#define I2SFIC 0x8 +#define I2SPSR 0xc +#define I2STXD 0x10 +#define I2SRXD 0x14 +#define I2SFICS 0x18 +#define I2STXDS 0x1c + +#define CON_RSTCLR (1 << 31) +#define CON_FRXOFSTATUS (1 << 26) +#define CON_FRXORINTEN (1 << 25) +#define CON_FTXSURSTAT (1 << 24) +#define CON_FTXSURINTEN (1 << 23) +#define CON_TXSDMA_PAUSE (1 << 20) +#define CON_TXSDMA_ACTIVE (1 << 18) + +#define CON_FTXURSTATUS (1 << 17) +#define CON_FTXURINTEN (1 << 16) +#define CON_TXFIFO2_EMPTY (1 << 15) +#define CON_TXFIFO1_EMPTY (1 << 14) +#define CON_TXFIFO2_FULL (1 << 13) +#define CON_TXFIFO1_FULL (1 << 12) + +#define CON_LRINDEX (1 << 11) +#define CON_TXFIFO_EMPTY (1 << 10) +#define CON_RXFIFO_EMPTY (1 << 9) +#define CON_TXFIFO_FULL (1 << 8) +#define CON_RXFIFO_FULL (1 << 7) +#define CON_TXDMA_PAUSE (1 << 6) +#define CON_RXDMA_PAUSE (1 << 5) +#define CON_TXCH_PAUSE (1 << 4) +#define CON_RXCH_PAUSE (1 << 3) +#define CON_TXDMA_ACTIVE (1 << 2) +#define CON_RXDMA_ACTIVE (1 << 1) +#define CON_ACTIVE (1 << 0) + +#define MOD_OPCLK_CDCLK_OUT (0 << 30) +#define MOD_OPCLK_CDCLK_IN (1 << 30) +#define MOD_OPCLK_BCLK_OUT (2 << 30) +#define MOD_OPCLK_PCLK (3 << 30) +#define MOD_OPCLK_MASK (3 << 30) +#define MOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ + +#define MOD_BLCS_SHIFT 26 +#define MOD_BLCS_16BIT (0 << MOD_BLCS_SHIFT) +#define MOD_BLCS_8BIT (1 << MOD_BLCS_SHIFT) +#define MOD_BLCS_24BIT (2 << MOD_BLCS_SHIFT) +#define MOD_BLCS_MASK (3 << MOD_BLCS_SHIFT) +#define MOD_BLCP_SHIFT 24 +#define MOD_BLCP_16BIT (0 << MOD_BLCP_SHIFT) +#define MOD_BLCP_8BIT (1 << MOD_BLCP_SHIFT) +#define MOD_BLCP_24BIT (2 << MOD_BLCP_SHIFT) +#define MOD_BLCP_MASK (3 << MOD_BLCP_SHIFT) + +#define MOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ +#define MOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ +#define MOD_C1DD_HHALF (1 << 19) +#define MOD_C1DD_LHALF (1 << 18) +#define MOD_DC2_EN (1 << 17) +#define MOD_DC1_EN (1 << 16) +#define MOD_BLC_16BIT (0 << 13) +#define MOD_BLC_8BIT (1 << 13) +#define MOD_BLC_24BIT (2 << 13) +#define MOD_BLC_MASK (3 << 13) + +#define MOD_IMS_SYSMUX (1 << 10) +#define MOD_SLAVE (1 << 11) +#define MOD_TXONLY (0 << 8) +#define MOD_RXONLY (1 << 8) +#define MOD_TXRX (2 << 8) +#define MOD_MASK (3 << 8) +#define MOD_LR_LLOW (0 << 7) +#define MOD_LR_RLOW (1 << 7) +#define MOD_SDF_IIS (0 << 5) +#define MOD_SDF_MSB (1 << 5) +#define MOD_SDF_LSB (2 << 5) +#define MOD_SDF_MASK (3 << 5) +#define MOD_RCLK_256FS (0 << 3) +#define MOD_RCLK_512FS (1 << 3) +#define MOD_RCLK_384FS (2 << 3) +#define MOD_RCLK_768FS (3 << 3) +#define MOD_RCLK_MASK (3 << 3) +#define MOD_BCLK_32FS (0 << 1) +#define MOD_BCLK_48FS (1 << 1) +#define MOD_BCLK_16FS (2 << 1) +#define MOD_BCLK_24FS (3 << 1) +#define MOD_BCLK_MASK (3 << 1) +#define MOD_8BIT (1 << 0) + +#define MOD_CDCLKCON (1 << 12) + +#define PSR_PSREN (1 << 15) + +#define FIC_TX2COUNT(x) (((x) >> 24) & 0xf) +#define FIC_TX1COUNT(x) (((x) >> 16) & 0xf) + +#define FIC_TXFLUSH (1 << 15) +#define FIC_RXFLUSH (1 << 7) +#define FIC_TXCOUNT(x) (((x) >> 8) & 0xf) +#define FIC_RXCOUNT(x) (((x) >> 0) & 0xf) +#define FICS_TXCOUNT(x) (((x) >> 8) & 0x7f) + +#endif /* __SND_SOC_SAMSUNG_I2S_REGS_H */ + + diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 992a732..1568eea 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -22,109 +22,7 @@ #include "dma.h" #include "i2s.h" - -#define I2SCON 0x0 -#define I2SMOD 0x4 -#define I2SFIC 0x8 -#define I2SPSR 0xc -#define I2STXD 0x10 -#define I2SRXD 0x14 -#define I2SFICS 0x18 -#define I2STXDS 0x1c - -#define CON_RSTCLR (1 << 31) -#define CON_FRXOFSTATUS (1 << 26) -#define CON_FRXORINTEN (1 << 25) -#define CON_FTXSURSTAT (1 << 24) -#define CON_FTXSURINTEN (1 << 23) -#define CON_TXSDMA_PAUSE (1 << 20) -#define CON_TXSDMA_ACTIVE (1 << 18) - -#define CON_FTXURSTATUS (1 << 17) -#define CON_FTXURINTEN (1 << 16) -#define CON_TXFIFO2_EMPTY (1 << 15) -#define CON_TXFIFO1_EMPTY (1 << 14) -#define CON_TXFIFO2_FULL (1 << 13) -#define CON_TXFIFO1_FULL (1 << 12) - -#define CON_LRINDEX (1 << 11) -#define CON_TXFIFO_EMPTY (1 << 10) -#define CON_RXFIFO_EMPTY (1 << 9) -#define CON_TXFIFO_FULL (1 << 8) -#define CON_RXFIFO_FULL (1 << 7) -#define CON_TXDMA_PAUSE (1 << 6) -#define CON_RXDMA_PAUSE (1 << 5) -#define CON_TXCH_PAUSE (1 << 4) -#define CON_RXCH_PAUSE (1 << 3) -#define CON_TXDMA_ACTIVE (1 << 2) -#define CON_RXDMA_ACTIVE (1 << 1) -#define CON_ACTIVE (1 << 0) - -#define MOD_OPCLK_CDCLK_OUT (0 << 30) -#define MOD_OPCLK_CDCLK_IN (1 << 30) -#define MOD_OPCLK_BCLK_OUT (2 << 30) -#define MOD_OPCLK_PCLK (3 << 30) -#define MOD_OPCLK_MASK (3 << 30) -#define MOD_TXS_IDMA (1 << 28) /* Sec_TXFIFO use I-DMA */ - -#define MOD_BLCS_SHIFT 26 -#define MOD_BLCS_16BIT (0 << MOD_BLCS_SHIFT) -#define MOD_BLCS_8BIT (1 << MOD_BLCS_SHIFT) -#define MOD_BLCS_24BIT (2 << MOD_BLCS_SHIFT) -#define MOD_BLCS_MASK (3 << MOD_BLCS_SHIFT) -#define MOD_BLCP_SHIFT 24 -#define MOD_BLCP_16BIT (0 << MOD_BLCP_SHIFT) -#define MOD_BLCP_8BIT (1 << MOD_BLCP_SHIFT) -#define MOD_BLCP_24BIT (2 << MOD_BLCP_SHIFT) -#define MOD_BLCP_MASK (3 << MOD_BLCP_SHIFT) - -#define MOD_C2DD_HHALF (1 << 21) /* Discard Higher-half */ -#define MOD_C2DD_LHALF (1 << 20) /* Discard Lower-half */ -#define MOD_C1DD_HHALF (1 << 19) -#define MOD_C1DD_LHALF (1 << 18) -#define MOD_DC2_EN (1 << 17) -#define MOD_DC1_EN (1 << 16) -#define MOD_BLC_16BIT (0 << 13) -#define MOD_BLC_8BIT (1 << 13) -#define MOD_BLC_24BIT (2 << 13) -#define MOD_BLC_MASK (3 << 13) - -#define MOD_IMS_SYSMUX (1 << 10) -#define MOD_SLAVE (1 << 11) -#define MOD_TXONLY (0 << 8) -#define MOD_RXONLY (1 << 8) -#define MOD_TXRX (2 << 8) -#define MOD_MASK (3 << 8) -#define MOD_LR_LLOW (0 << 7) -#define MOD_LR_RLOW (1 << 7) -#define MOD_SDF_IIS (0 << 5) -#define MOD_SDF_MSB (1 << 5) -#define MOD_SDF_LSB (2 << 5) -#define MOD_SDF_MASK (3 << 5) -#define MOD_RCLK_256FS (0 << 3) -#define MOD_RCLK_512FS (1 << 3) -#define MOD_RCLK_384FS (2 << 3) -#define MOD_RCLK_768FS (3 << 3) -#define MOD_RCLK_MASK (3 << 3) -#define MOD_BCLK_32FS (0 << 1) -#define MOD_BCLK_48FS (1 << 1) -#define MOD_BCLK_16FS (2 << 1) -#define MOD_BCLK_24FS (3 << 1) -#define MOD_BCLK_MASK (3 << 1) -#define MOD_8BIT (1 << 0) - -#define MOD_CDCLKCON (1 << 12) - -#define PSR_PSREN (1 << 15) - -#define FIC_TX2COUNT(x) (((x) >> 24) & 0xf) -#define FIC_TX1COUNT(x) (((x) >> 16) & 0xf) - -#define FIC_TXFLUSH (1 << 15) -#define FIC_RXFLUSH (1 << 7) -#define FIC_TXCOUNT(x) (((x) >> 8) & 0xf) -#define FIC_RXCOUNT(x) (((x) >> 0) & 0xf) -#define FICS_TXCOUNT(x) (((x) >> 8) & 0x7f) +#include "i2s-regs.h" #define msecs_to_loops(t) (loops_per_jiffy / 1000 * HZ * t) -- 1.7.1

3 9

[alsa-devel] [PATCH] [PATCH V2] ASoC: SAMSUNG: Add WM8994 PCM Machine driver
by Sangbeom Kim 26 Jun '11

26 Jun '11

This patch add WM8994 PCM machine driver to support PCM audio on SMDKV310, SMDKC210 boards. Playback and Capture supports 8kHz sampling rates. and It is tested on SMDKV310, SMDKC210. Signed-off-by: Sangbeom Kim <sbkim73(a)samsung.com> --- sound/soc/samsung/Kconfig | 8 ++ sound/soc/samsung/Makefile | 2 + sound/soc/samsung/smdk_wm8994pcm.c | 176 ++++++++++++++++++++++++++++++++++++ 3 files changed, 186 insertions(+), 0 deletions(-) create mode 100644 sound/soc/samsung/smdk_wm8994pcm.c diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index d155cbb..556113e 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -171,6 +171,14 @@ config SND_SOC_SMDK_WM8580_PCM help Say Y if you want to add support for SoC audio on the SMDK. +config SND_SOC_SMDK_WM8994_PCM + tristate "SoC PCM Audio support for WM8994 on SMDK" + depends on SND_SOC_SAMSUNG && (MACH_SMDKC210 || MACH_SMDKV310) + select SND_SOC_WM8994 + select SND_SAMSUNG_PCM + help + Say Y if you want to add support for SoC audio on the SMDK + config SND_SOC_SPEYSIDE tristate "Audio support for Wolfson Speyside" depends on SND_SOC_SAMSUNG && MACH_WLF_CRAGG_6410 diff --git a/sound/soc/samsung/Makefile b/sound/soc/samsung/Makefile index 2b9e251..41becdd 100644 --- a/sound/soc/samsung/Makefile +++ b/sound/soc/samsung/Makefile @@ -37,6 +37,7 @@ snd-soc-s3c64xx-smartq-wm8987-objs := smartq_wm8987.o snd-soc-goni-wm8994-objs := goni_wm8994.o snd-soc-smdk-spdif-objs := smdk_spdif.o snd-soc-smdk-wm8580pcm-objs := smdk_wm8580pcm.o +snd-soc-smdk-wm8994pcm-objs := smdk_wm8994pcm.o snd-soc-speyside-objs := speyside.o obj-$(CONFIG_SND_SOC_SAMSUNG_JIVE_WM8750) += snd-soc-jive-wm8750.o @@ -56,4 +57,5 @@ obj-$(CONFIG_SND_SOC_SMARTQ) += snd-soc-s3c64xx-smartq-wm8987.o obj-$(CONFIG_SND_SOC_SAMSUNG_SMDK_SPDIF) += snd-soc-smdk-spdif.o obj-$(CONFIG_SND_SOC_GONI_AQUILA_WM8994) += snd-soc-goni-wm8994.o obj-$(CONFIG_SND_SOC_SMDK_WM8580_PCM) += snd-soc-smdk-wm8580pcm.o +obj-$(CONFIG_SND_SOC_SMDK_WM8994_PCM) += snd-soc-smdk-wm8994pcm.o obj-$(CONFIG_SND_SOC_SPEYSIDE) += snd-soc-speyside.o diff --git a/sound/soc/samsung/smdk_wm8994pcm.c b/sound/soc/samsung/smdk_wm8994pcm.c new file mode 100644 index 0000000..5f21116 --- /dev/null +++ b/sound/soc/samsung/smdk_wm8994pcm.c @@ -0,0 +1,176 @@ +/* + * sound/soc/samsung/smdk_wm8994pcm.c + * + * Copyright (c) 2011 Samsung Electronics Co., Ltd + * http://www.samsung.com + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + */ +#include <sound/soc.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> + +#include "../codecs/wm8994.h" +#include "dma.h" +#include "pcm.h" + +/* + * Board Settings: + * o '1' means 'ON' + * o '0' means 'OFF' + * o 'X' means 'Don't care' + * + * SMDKC210, SMDKV310: CFG3- 1001, CFG5-1000, CFG7-111111 + */ + +/* + * Configure audio route as :- + * $ amixer sset 'DAC1' on,on + * $ amixer sset 'Right Headphone Mux' 'DAC' + * $ amixer sset 'Left Headphone Mux' 'DAC' + * $ amixer sset 'DAC1R Mixer AIF1.1' on + * $ amixer sset 'DAC1L Mixer AIF1.1' on + * $ amixer sset 'IN2L' on + * $ amixer sset 'IN2L PGA IN2LN' on + * $ amixer sset 'MIXINL IN2L' on + * $ amixer sset 'AIF1ADC1L Mixer ADC/DMIC' on + * $ amixer sset 'IN2R' on + * $ amixer sset 'IN2R PGA IN2RN' on + * $ amixer sset 'MIXINR IN2R' on + * $ amixer sset 'AIF1ADC1R Mixer ADC/DMIC' on + */ + +/* SMDK has a 16.9344MHZ crystal attached to WM8994 */ +#define SMDK_WM8994_FREQ 16934400 + +static int smdk_wm8994_pcm_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->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; + unsigned long mclk_freq; + int rfs, ret; + + switch(params_rate(params)) { + case 8000: + rfs = 512; + break; + default: + dev_err(cpu_dai->dev, "%s:%d Sampling Rate %u not supported!\n", + __func__, __LINE__, params_rate(params)); + return -EINVAL; + } + + mclk_freq = params_rate(params) * rfs; + + /* Set the codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + /* Set the cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_B + | SND_SOC_DAIFMT_IB_NF + | SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8994_SYSCLK_FLL1, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(codec_dai, WM8994_FLL1, WM8994_FLL_SRC_MCLK1, + SMDK_WM8994_FREQ, mclk_freq); + if (ret < 0) + return ret; + + /* Set PCM source clock on CPU */ + ret = snd_soc_dai_set_sysclk(cpu_dai, S3C_PCM_CLKSRC_MUX, + mclk_freq, SND_SOC_CLOCK_IN); + if (ret < 0) + return ret; + + /* Set SCLK_DIV for making bclk */ + ret = snd_soc_dai_set_clkdiv(cpu_dai, S3C_PCM_SCLK_PER_FS, rfs); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_ops smdk_wm8994_pcm_ops = { + .hw_params = smdk_wm8994_pcm_hw_params, +}; + +static struct snd_soc_dai_link smdk_dai[] = { + { + .name = "WM8994 PAIF PCM", + .stream_name = "Primary PCM", + .cpu_dai_name = "samsung-pcm.0", + .codec_dai_name = "wm8994-aif1", + .platform_name = "samsung-audio", + .codec_name = "wm8994-codec", + .ops = &smdk_wm8994_pcm_ops, + }, +}; + +static struct snd_soc_card smdk_pcm = { + .name = "SMDK-PCM", + .dai_link = smdk_dai, + .num_links = 1, +}; + +static int __devinit snd_smdk_probe(struct platform_device *pdev) +{ + int ret = 0; + + smdk_pcm.dev = &pdev->dev; + ret = snd_soc_register_card(&smdk_pcm); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed %d\n", ret); + return ret; + } + + return 0; +} + +static int __devexit snd_smdk_remove(struct platform_device *pdev) +{ + snd_soc_unregister_card(&smdk_pcm); + platform_set_drvdata(pdev, NULL); + return 0; +} + +static struct platform_driver snd_smdk_driver = { + .driver = { + .owner = THIS_MODULE, + .name = "samsung-smdk-pcm", + }, + .probe = snd_smdk_probe, + .remove = __devexit_p(snd_smdk_remove), +}; + +static int __init smdk_audio_init(void) +{ + return platform_driver_register(&snd_smdk_driver); +} + +module_init(smdk_audio_init); + +static void __exit smdk_audio_exit(void) +{ + platform_driver_unregister(&snd_smdk_driver); +} + +module_exit(smdk_audio_exit); + +MODULE_AUTHOR("Sangbeom Kim, <sbkim73(a)samsung.com>"); +MODULE_DESCRIPTION("ALSA SoC SMDK WM8994 for PCM"); +MODULE_LICENSE("GPL"); -- 1.7.1

4 3

[alsa-devel] [PATCH] alsa: lx6464es - include mac address in device name
by Tim Blechmann 25 Jun '11

25 Jun '11

each device has a unique mac address, which can be used to distinguish multiple devices in the same machine. we therefore include the full mac address in the device shortname and the last 6 bytes in the device id. Signed-off-by: Tim Blechmann <tim(a)klingt.org> --- sound/pci/lx6464es/lx6464es.c | 21 +++++++++++++-------- sound/pci/lx6464es/lx6464es.h | 2 ++ sound/pci/lx6464es/lx_core.c | 14 +++++++------- sound/pci/lx6464es/lx_core.h | 2 +- 4 files changed, 23 insertions(+), 16 deletions(-) diff --git a/sound/pci/lx6464es/lx6464es.c b/sound/pci/lx6464es/lx6464es.c index 1bd7a54..2736cf1 100644 --- a/sound/pci/lx6464es/lx6464es.c +++ b/sound/pci/lx6464es/lx6464es.c @@ -762,7 +762,6 @@ static int lx_set_granularity(struct lx6464es *chip, u32 gran) static int __devinit lx_init_dsp(struct lx6464es *chip) { int err; - u8 mac_address[6]; int i; snd_printdd("->lx_init_dsp\n"); @@ -787,11 +786,11 @@ static int __devinit lx_init_dsp(struct lx6464es *chip) /** \todo the mac address should be ready by not, but it isn't, * so we wait for it */ for (i = 0; i != 1000; ++i) { - err = lx_dsp_get_mac(chip, mac_address); + err = lx_dsp_get_mac(chip); if (err) return err; - if (mac_address[0] || mac_address[1] || mac_address[2] || - mac_address[3] || mac_address[4] || mac_address[5]) + if (chip->mac_address[0] || chip->mac_address[1] || chip->mac_address[2] || + chip->mac_address[3] || chip->mac_address[4] || chip->mac_address[5]) goto mac_ready; msleep(1); } @@ -800,8 +799,8 @@ static int __devinit lx_init_dsp(struct lx6464es *chip) mac_ready: snd_printd(LXP "mac address ready read after: %dms\n", i); snd_printk(LXP "mac address: %02X.%02X.%02X.%02X.%02X.%02X\n", - mac_address[0], mac_address[1], mac_address[2], - mac_address[3], mac_address[4], mac_address[5]); + chip->mac_address[0], chip->mac_address[1], chip->mac_address[2], + chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]); err = lx_init_get_version_features(chip); if (err) @@ -1108,8 +1107,14 @@ static int __devinit snd_lx6464es_probe(struct pci_dev *pci, goto out_free; } - strcpy(card->driver, "lx6464es"); - strcpy(card->shortname, "Digigram LX6464ES"); + strcpy(card->driver, "LX6464ES"); + sprintf(card->id, "LX6464ES_%02X%02X%02X", + chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]); + + sprintf(card->shortname, "LX6464ES %02X.%02X.%02X.%02X.%02X.%02X", + chip->mac_address[0], chip->mac_address[1], chip->mac_address[2], + chip->mac_address[3], chip->mac_address[4], chip->mac_address[5]); + sprintf(card->longname, "%s at 0x%lx, 0x%p, irq %i", card->shortname, chip->port_plx, chip->port_dsp_bar, chip->irq); diff --git a/sound/pci/lx6464es/lx6464es.h b/sound/pci/lx6464es/lx6464es.h index aea621e..e2a124a 100644 --- a/sound/pci/lx6464es/lx6464es.h +++ b/sound/pci/lx6464es/lx6464es.h @@ -69,6 +69,8 @@ struct lx6464es { struct pci_dev *pci; int irq; + u8 mac_address[6]; + spinlock_t lock; /* interrupt spinlock */ struct mutex setup_mutex; /* mutex used in hw_params, open * and close */ diff --git a/sound/pci/lx6464es/lx_core.c b/sound/pci/lx6464es/lx_core.c index 617f98b..5c8717e 100644 --- a/sound/pci/lx6464es/lx_core.c +++ b/sound/pci/lx6464es/lx_core.c @@ -424,7 +424,7 @@ int lx_dsp_get_clock_frequency(struct lx6464es *chip, u32 *rfreq) return ret; } -int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address) +int lx_dsp_get_mac(struct lx6464es *chip) { u32 macmsb, maclsb; @@ -432,12 +432,12 @@ int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address) maclsb = lx_dsp_reg_read(chip, eReg_ADMACESLSB) & 0x00FFFFFF; /* todo: endianess handling */ - mac_address[5] = ((u8 *)(&maclsb))[0]; - mac_address[4] = ((u8 *)(&maclsb))[1]; - mac_address[3] = ((u8 *)(&maclsb))[2]; - mac_address[2] = ((u8 *)(&macmsb))[0]; - mac_address[1] = ((u8 *)(&macmsb))[1]; - mac_address[0] = ((u8 *)(&macmsb))[2]; + chip->mac_address[5] = ((u8 *)(&maclsb))[0]; + chip->mac_address[4] = ((u8 *)(&maclsb))[1]; + chip->mac_address[3] = ((u8 *)(&maclsb))[2]; + chip->mac_address[2] = ((u8 *)(&macmsb))[0]; + chip->mac_address[1] = ((u8 *)(&macmsb))[1]; + chip->mac_address[0] = ((u8 *)(&macmsb))[2]; return 0; } diff --git a/sound/pci/lx6464es/lx_core.h b/sound/pci/lx6464es/lx_core.h index 6bd9cbb..1dd5629 100644 --- a/sound/pci/lx6464es/lx_core.h +++ b/sound/pci/lx6464es/lx_core.h @@ -116,7 +116,7 @@ int __devinit lx_dsp_get_version(struct lx6464es *chip, u32 *rdsp_version); int lx_dsp_get_clock_frequency(struct lx6464es *chip, u32 *rfreq); int lx_dsp_set_granularity(struct lx6464es *chip, u32 gran); int lx_dsp_read_async_events(struct lx6464es *chip, u32 *data); -int lx_dsp_get_mac(struct lx6464es *chip, u8 *mac_address); +int lx_dsp_get_mac(struct lx6464es *chip); /* low-level pipe handling */ -- 1.7.4.1

2 1

[alsa-devel] [PATCH] SOUND: Fix non-ISA_DMA_API build failure
by Ralf Baechle 24 Jun '11

24 Jun '11

A build with ISA && ISA_DMA && !ISA_DMA_API results in: CC sound/isa/es18xx.o sound/isa/es18xx.c: In function ‘snd_es18xx_playback1_prepare’: sound/isa/es18xx.c:501:9: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/es18xx.c: In function ‘snd_es18xx_playback_pointer’: sound/isa/es18xx.c:818:3: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[2]: *** [sound/isa/es18xx.o] Error 1 CC sound/isa/sscape.o sound/isa/sscape.c: In function ‘upload_dma_data’: sound/isa/sscape.c:481:3: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[2]: *** [sound/isa/sscape.o] Error 1 CC sound/isa/ad1816a/ad1816a_lib.o sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_playback_prepare’: sound/isa/ad1816a/ad1816a_lib.c:244:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_playback_pointer’: sound/isa/ad1816a/ad1816a_lib.c:302:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_free’: sound/isa/ad1816a/ad1816a_lib.c:544:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/ad1816a/ad1816a_lib.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/ad1816a] Error 2 CC sound/isa/es1688/es1688_lib.o sound/isa/es1688/es1688_lib.c: In function ‘snd_es1688_playback_prepare’: sound/isa/es1688/es1688_lib.c:417:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/es1688/es1688_lib.c: In function ‘snd_es1688_playback_pointer’: sound/isa/es1688/es1688_lib.c:509:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/es1688/es1688_lib.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/es1688] Error 2 CC sound/isa/gus/gus_dma.o sound/isa/gus/gus_dma.c: In function ‘snd_gf1_dma_program’: sound/isa/gus/gus_dma.c:79:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/gus/gus_dma.c: In function ‘snd_gf1_dma_done’: sound/isa/gus/gus_dma.c:177:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/gus/gus_dma.o] Error 1 CC sound/isa/gus/gus_pcm.o sound/isa/gus/gus_pcm.c: In function ‘snd_gf1_pcm_capture_prepare’: sound/isa/gus/gus_pcm.c:591:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/gus/gus_pcm.c: In function ‘snd_gf1_pcm_capture_pointer’: sound/isa/gus/gus_pcm.c:619:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/gus/gus_pcm.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/gus] Error 2 CC sound/isa/sb/sb16_csp.o sound/isa/sb/sb16_csp.c: In function ‘snd_sb_csp_ioctl’: sound/isa/sb/sb16_csp.c:228:227: error: case label does not reduce to an integer constant make[3]: *** [sound/isa/sb/sb16_csp.o] Error 1 CC sound/isa/sb/sb16_main.o sound/isa/sb/sb16_main.c: In function ‘snd_sb16_playback_prepare’: sound/isa/sb/sb16_main.c:276:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/sb/sb16_main.c: In function ‘snd_sb16_playback_pointer’: sound/isa/sb/sb16_main.c:456:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/sb/sb16_main.o] Error 1 CC sound/isa/sb/sb8_main.o sound/isa/sb/sb8_main.c: In function ‘snd_sb8_playback_prepare’: sound/isa/sb/sb8_main.c:172:3: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/sb/sb8_main.c: In function ‘snd_sb8_playback_pointer’: sound/isa/sb/sb8_main.c:425:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/sb/sb8_main.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/sb] Error 2 CC sound/isa/wss/wss_lib.o sound/isa/wss/wss_lib.c: In function ‘snd_wss_playback_prepare’: sound/isa/wss/wss_lib.c:1025:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/wss/wss_lib.c: In function ‘snd_wss_playback_pointer’: sound/isa/wss/wss_lib.c:1160:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] sound/isa/wss/wss_lib.c: In function ‘snd_wss_free’: sound/isa/wss/wss_lib.c:1695:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/wss/wss_lib.o] Error 1 Disabling these in Kconfig by adding a dependency on ISA_DMA_API then results in: warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) warning: (RADIO_MIROPCM20) selects SND_MIRO which has unmet direct dependencies (SOUND && !M68K && SND && SND_ISA && ISA_DMA_API) # [...] LD .tmp_vmlinux1 sound/built-in.o: In function `snd_miro_probe': /home/ralf/src/linux/linux-mips/sound/isa/opti9xx/miro.c:1321: undefined reference to `snd_wss_create' /home/ralf/src/linux/linux-mips/sound/isa/opti9xx/miro.c:1327: undefined reference to `snd_wss_pcm' /home/ralf/src/linux/linux-mips/sound/isa/opti9xx/miro.c:1331: undefined reference to `snd_wss_mixer' /home/ralf/src/linux/linux-mips/sound/isa/opti9xx/miro.c:1335: undefined reference to `snd_wss_timer' /home/ralf/src/linux/linux-mips/sound/isa/opti9xx/miro.c:1391: undefined reference to `snd_opl4_create' make: *** [.tmp_vmlinux1] Error 1 Which makes also touching drivers/media/radio/Kconfig necessary. Fixed by adding an explicit dependency on ISA_DMA_API for all of the config statment that either result in the direction inclusion of code that calls the ISA DMA API or selects something which in turn would use the ISA DMA API. The sole ISA sound driver that does not use the ISA DMA API is the Adlib driver so replaced the dependency of SND_ISA on ISA_DMA_API and add it to each of the drivers individually. Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> --- With this (and a separate MIPS-specific patch) I can get all audio drivers in allyesconfig to build. The generated i386/x86_64 allyesconfigs remain unchanged by this patch. drivers/media/radio/Kconfig | 2 +- sound/isa/Kconfig | 40 ++++++++++++++++++++++++++++++---------- 2 files changed, 31 insertions(+), 11 deletions(-) diff --git a/drivers/media/radio/Kconfig b/drivers/media/radio/Kconfig index e4c97fd..0aeed28 100644 --- a/drivers/media/radio/Kconfig +++ b/drivers/media/radio/Kconfig @@ -168,7 +168,7 @@ config RADIO_MAXIRADIO config RADIO_MIROPCM20 tristate "miroSOUND PCM20 radio" - depends on ISA && VIDEO_V4L2 && SND + depends on ISA && ISA_DMA_API && VIDEO_V4L2 && SND select SND_ISA select SND_MIRO ---help--- diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index 52064cf..25230c8 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -19,7 +19,7 @@ config SND_SB16_DSP menuconfig SND_ISA bool "ISA sound devices" - depends on ISA && ISA_DMA_API + depends on ISA default y help Support for sound devices connected via the ISA bus. @@ -37,7 +37,7 @@ config SND_ADLIB config SND_AD1816A tristate "Analog Devices SoundPort AD1816A" - depends on PNP + depends on ISA_DMA_API && PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -52,6 +52,7 @@ config SND_AD1816A config SND_AD1848 tristate "Generic AD1848/CS4248 driver" select SND_WSS_LIB + depends on ISA_DMA_API help Say Y here to include support for AD1848 (Analog Devices) or CS4248 (Cirrus Logic - Crystal Semiconductors) chips. @@ -64,7 +65,7 @@ config SND_AD1848 config SND_ALS100 tristate "Diamond Tech. DT-019x and Avance Logic ALSxxx" - depends on PNP + depends on ISA_DMA_API && PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -79,7 +80,7 @@ config SND_ALS100 config SND_AZT1605 tristate "Aztech AZT1605 Driver" - depends on SND + depends on ISA_DMA_API && SND select SND_WSS_LIB select SND_MPU401_UART select SND_OPL3_LIB @@ -92,7 +93,7 @@ config SND_AZT1605 config SND_AZT2316 tristate "Aztech AZT2316 Driver" - depends on SND + depends on ISA_DMA_API && SND select SND_WSS_LIB select SND_MPU401_UART select SND_OPL3_LIB @@ -105,7 +106,7 @@ config SND_AZT2316 config SND_AZT2320 tristate "Aztech Systems AZT2320" - depends on PNP + depends on ISA_DMA_API && PNP select ISAPNP select SND_OPL3_LIB select SND_MPU401_UART @@ -119,6 +120,7 @@ config SND_AZT2320 config SND_CMI8330 tristate "C-Media CMI8330" + depends on ISA_DMA_API select SND_WSS_LIB select SND_SB16_DSP select SND_OPL3_LIB @@ -132,6 +134,7 @@ config SND_CMI8330 config SND_CS4231 tristate "Generic Cirrus Logic CS4231 driver" + depends on ISA_DMA_API select SND_MPU401_UART select SND_WSS_LIB help @@ -143,6 +146,7 @@ config SND_CS4231 config SND_CS4236 tristate "Generic Cirrus Logic CS4232/CS4236+ driver" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB @@ -156,6 +160,7 @@ config SND_CS4236 config SND_ES1688 tristate "Generic ESS ES688/ES1688 and ES968 PnP driver" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -168,6 +173,7 @@ config SND_ES1688 config SND_ES18XX tristate "Generic ESS ES18xx driver" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -179,7 +185,7 @@ config SND_ES18XX config SND_SC6000 tristate "Gallant SC-6000/6600/7000 and Audio Excel DSP 16" - depends on HAS_IOPORT + depends on ISA_DMA_API && HAS_IOPORT select SND_WSS_LIB select SND_OPL3_LIB select SND_MPU401_UART @@ -195,6 +201,7 @@ config SND_SC6000 config SND_GUSCLASSIC tristate "Gravis UltraSound Classic" + depends on ISA_DMA_API select SND_RAWMIDI select SND_PCM help @@ -206,6 +213,7 @@ config SND_GUSCLASSIC config SND_GUSEXTREME tristate "Gravis UltraSound Extreme" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_PCM @@ -218,6 +226,7 @@ config SND_GUSEXTREME config SND_GUSMAX tristate "Gravis UltraSound MAX" + depends on ISA_DMA_API select SND_RAWMIDI select SND_WSS_LIB help @@ -229,7 +238,7 @@ config SND_GUSMAX config SND_INTERWAVE tristate "AMD InterWave, Gravis UltraSound PnP" - depends on PNP + depends on ISA_DMA_API && PNP select SND_RAWMIDI select SND_WSS_LIB help @@ -242,7 +251,7 @@ config SND_INTERWAVE config SND_INTERWAVE_STB tristate "AMD InterWave + TEA6330T (UltraSound 32-Pro)" - depends on PNP + depends on ISA_DMA_API && PNP select SND_RAWMIDI select SND_WSS_LIB help @@ -255,6 +264,7 @@ config SND_INTERWAVE_STB config SND_JAZZ16 tristate "Media Vision Jazz16 card and compatibles" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_SB8_DSP @@ -271,6 +281,7 @@ config SND_JAZZ16 config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB @@ -283,6 +294,7 @@ config SND_OPL3SA2 config SND_OPTI92X_AD1848 tristate "OPTi 82C92x - AD1848" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART @@ -296,6 +308,7 @@ config SND_OPTI92X_AD1848 config SND_OPTI92X_CS4231 tristate "OPTi 82C92x - CS4231" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_OPL4_LIB select SND_MPU401_UART @@ -309,6 +322,7 @@ config SND_OPTI92X_CS4231 config SND_OPTI93X tristate "OPTi 82C93x" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_WSS_LIB @@ -321,6 +335,7 @@ config SND_OPTI93X config SND_MIRO tristate "Miro miroSOUND PCM1pro/PCM12/PCM20radio driver" + depends on ISA_DMA_API select SND_OPL4_LIB select SND_WSS_LIB select SND_MPU401_UART @@ -334,6 +349,7 @@ config SND_MIRO config SND_SB8 tristate "Sound Blaster 1.0/2.0/Pro (8-bit)" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_RAWMIDI select SND_SB8_DSP @@ -346,6 +362,7 @@ config SND_SB8 config SND_SB16 tristate "Sound Blaster 16 (PnP)" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP @@ -358,6 +375,7 @@ config SND_SB16 config SND_SBAWE tristate "Sound Blaster AWE (32,64) (PnP)" + depends on ISA_DMA_API select SND_OPL3_LIB select SND_MPU401_UART select SND_SB16_DSP @@ -370,7 +388,7 @@ config SND_SBAWE config SND_SB16_CSP bool "Sound Blaster 16/AWE CSP support" - depends on (SND_SB16 || SND_SBAWE) && (BROKEN || !PPC) + depends on ISA_DMA_API && (SND_SB16 || SND_SBAWE) && (BROKEN || !PPC) select FW_LOADER help Say Y here to include support for the CSP core. This special @@ -379,6 +397,7 @@ config SND_SB16_CSP config SND_SSCAPE tristate "Ensoniq SoundScape driver" + depends on ISA_DMA_API select SND_MPU401_UART select SND_WSS_LIB select FW_LOADER @@ -400,6 +419,7 @@ config SND_SSCAPE config SND_WAVEFRONT tristate "Turtle Beach Maui,Tropez,Tropez+ (Wavefront)" + depends on ISA_DMA_API select FW_LOADER select SND_OPL3_LIB select SND_MPU401_UART

2 2

[alsa-devel] [PATCH] MEDIA: Fix non-ISA_DMA_API link failure of sound code
by Ralf Baechle 24 Jun '11

24 Jun '11

A build with ISA && ISA_DMA && !ISA_DMA_API results in: CC sound/isa/es18xx.o sound/isa/es18xx.c: In function ‘snd_es18xx_playback1_prepare’: sound/isa/es18xx.c:501:9: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/es18xx.c: In function ‘snd_es18xx_playback_pointer’: sound/isa/es18xx.c:818:3: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[2]: *** [sound/isa/es18xx.o] Error 1 CC sound/isa/sscape.o sound/isa/sscape.c: In function ‘upload_dma_data’: sound/isa/sscape.c:481:3: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[2]: *** [sound/isa/sscape.o] Error 1 CC sound/isa/ad1816a/ad1816a_lib.o sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_playback_prepare’: sound/isa/ad1816a/ad1816a_lib.c:244:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_playback_pointer’: sound/isa/ad1816a/ad1816a_lib.c:302:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] sound/isa/ad1816a/ad1816a_lib.c: In function ‘snd_ad1816a_free’: sound/isa/ad1816a/ad1816a_lib.c:544:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/ad1816a/ad1816a_lib.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/ad1816a] Error 2 CC sound/isa/es1688/es1688_lib.o sound/isa/es1688/es1688_lib.c: In function ‘snd_es1688_playback_prepare’: sound/isa/es1688/es1688_lib.c:417:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/es1688/es1688_lib.c: In function ‘snd_es1688_playback_pointer’: sound/isa/es1688/es1688_lib.c:509:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/es1688/es1688_lib.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/es1688] Error 2 CC sound/isa/gus/gus_dma.o sound/isa/gus/gus_dma.c: In function ‘snd_gf1_dma_program’: sound/isa/gus/gus_dma.c:79:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/gus/gus_dma.c: In function ‘snd_gf1_dma_done’: sound/isa/gus/gus_dma.c:177:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/gus/gus_dma.o] Error 1 CC sound/isa/gus/gus_pcm.o sound/isa/gus/gus_pcm.c: In function ‘snd_gf1_pcm_capture_prepare’: sound/isa/gus/gus_pcm.c:591:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/gus/gus_pcm.c: In function ‘snd_gf1_pcm_capture_pointer’: sound/isa/gus/gus_pcm.c:619:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/gus/gus_pcm.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/gus] Error 2 CC sound/isa/sb/sb16_csp.o sound/isa/sb/sb16_csp.c: In function ‘snd_sb_csp_ioctl’: sound/isa/sb/sb16_csp.c:228:227: error: case label does not reduce to an integer constant make[3]: *** [sound/isa/sb/sb16_csp.o] Error 1 CC sound/isa/sb/sb16_main.o sound/isa/sb/sb16_main.c: In function ‘snd_sb16_playback_prepare’: sound/isa/sb/sb16_main.c:276:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/sb/sb16_main.c: In function ‘snd_sb16_playback_pointer’: sound/isa/sb/sb16_main.c:456:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/sb/sb16_main.o] Error 1 CC sound/isa/sb/sb8_main.o sound/isa/sb/sb8_main.c: In function ‘snd_sb8_playback_prepare’: sound/isa/sb/sb8_main.c:172:3: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/sb/sb8_main.c: In function ‘snd_sb8_playback_pointer’: sound/isa/sb/sb8_main.c:425:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/sb/sb8_main.o] Error 1 make[3]: Target `__build' not remade because of errors. make[2]: *** [sound/isa/sb] Error 2 CC sound/isa/wss/wss_lib.o sound/isa/wss/wss_lib.c: In function ‘snd_wss_playback_prepare’: sound/isa/wss/wss_lib.c:1025:2: error: implicit declaration of function ‘snd_dma_program’ [-Werror=implicit-function-declaration] sound/isa/wss/wss_lib.c: In function ‘snd_wss_playback_pointer’: sound/isa/wss/wss_lib.c:1160:2: error: implicit declaration of function ‘snd_dma_pointer’ [-Werror=implicit-function-declaration] sound/isa/wss/wss_lib.c: In function ‘snd_wss_free’: sound/isa/wss/wss_lib.c:1695:3: error: implicit declaration of function ‘snd_dma_disable’ [-Werror=implicit-function-declaration] cc1: some warnings being treated as errors make[3]: *** [sound/isa/wss/wss_lib.o] Error 1 The root cause for this is hidden in this Kconfig warning: warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) Adding a dependency on ISA_DMA_API to RADIO_MIROPCM20 fixes these issues. Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> drivers/media/radio/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/drivers/media/radio/Kconfig b/drivers/media/radio/Kconfig index e4c97fd..0aeed28 100644 --- a/drivers/media/radio/Kconfig +++ b/drivers/media/radio/Kconfig @@ -168,7 +168,7 @@ config RADIO_MAXIRADIO config RADIO_MIROPCM20 tristate "miroSOUND PCM20 radio" - depends on ISA && VIDEO_V4L2 && SND + depends on ISA && ISA_DMA_API && VIDEO_V4L2 && SND select SND_ISA select SND_MIRO ---help---

1 0

Re: [alsa-devel] [PATCH] SOUND: Fix non-ISA_DMA_API build failure
by Ralf Baechle 24 Jun '11

24 Jun '11

On Fri, Jun 24, 2011 at 10:26:13AM +0200, Takashi Iwai wrote: > Hrm... I still don't understand why ES18XX or others were selected at > the first place. Isn't it covered by the conditional in > sound/isa/Kconfig like below? > > ================================================================ > menuconfig SND_ISA > bool "ISA sound devices" > depends on ISA && ISA_DMA_API > ... > if SND_ISA > ... > config SND_ES18XX > tristate "Generic ESS ES18xx driver" > ... > endif # SND_ISA > ================================================================ > > Isn't SND_ISA=n in your case although ISA_DMA_API=n? The answer is hidden in this Kconfig warning: warning: (RADIO_MIROPCM20) selects SND_ISA which has unmet direct dependencies (SOUND && !M68K && SND && ISA && ISA_DMA_API) This is due to the following in drivers/media/radio/Kconfig: config RADIO_MIROPCM20 tristate "miroSOUND PCM20 radio" depends on ISA && VIDEO_V4L2 && SND select SND_ISA select SND_MIRO So SND_ISA gets forced on even though the dependency on ISA_DMA_API is not fulfilled. That's solved by adding the dependency on ISA_DMA_API to RADIO_MIROPCM20. > Also, adlib driver is really only for ISA, so I see no big reason to > allow this built for non-ISA. With the patch applied: [...] menuconfig SND_ISA bool "ISA sound devices" depends on ISA [...] if SND_ISA config SND_ADLIB tristate "AdLib FM card" select SND_OPL3_LIB [...] So the Adlib driver will still only be built with ISA enabled. The only thing that makes the Adlib driver different from all the others in the ifdef SND_ISA ... endif bracket is that it does not directly or indirectly use the ISA DMA API and that's in the end the reason why sound/isa/Kconfig needs to be changed. I originally approach this a different way but now that I'm explaining the details I notice that it probably makes sense to split this patch into two: o The drivers/media/radio/Kconfig part should be applied for 3.0 and maybe -stable. o The sound/isa/Kconfig part is basically only fixing the dependency for the Adlib driver allowing it to be built on non-ISA_DMA_API system and is material for the next release after 3.0. If you agree I'm going to repost the patch with aproprite log messages. Ralf

3 5

[alsa-devel] SND_SOC_CUSTOM and how to use it
by Raffaele Recalcati 24 Jun '11

24 Jun '11

Hi everibody, I'm integrating the zl38005 Zarling hw codec in ASoC. I'll use analog audio path to send and receive audio from cpu analog port (there are also other ways to interface hw, but that one is implemented in my board). I'll use spi cpu port to read and write registers, but with a little complex protocol. I have this spi commands in a char driver and I suppose I could leave the protocol there also for making easier the parameteres tuning from user-space and also I could call that commands from snd_soc_codec_set_cache_io function. Any comment? Here you are spi protocol that I need to integrate (and that I'll make better befor publishing...) --------------------- drivers/char/zl38005.c ----------------------- /* * Copyright (C) 2011 Bticino S.p.A. * * This program is free software you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option)any later version. * * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * * Zarlink zl38005 (Acoustic Echo Canceller) * Some not public information, delivered by Zarlink, * has been used to implement the following code. * */ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/device.h> #include <linux/sysfs.h> #include <linux/spi/spi.h> #include <linux/slab.h> #include <linux/kthread.h> #include <linux/fs.h> #include <linux/list.h> #include <linux/smp_lock.h> #include <linux/cdev.h> #include <linux/uaccess.h> #include "zl38005.h" MODULE_DESCRIPTION("zl38005 codec driver"); MODULE_AUTHOR("Raffaele Recalcati <raffaele.recalcati(a)bticino.it>"); MODULE_LICENSE("GPL"); #ifdef DEBUG #define deb(args...) printk(KERN_DEBUG "%s: " __func__, ##args); #else #define deb(args...) #endif /* ZL38005 driver private data */ struct zl38005 { dev_t dev; struct spi_device *spi; struct list_head device_entry; }; /* For registeration of charatcer device */ static struct cdev c_dev; int module_usage_count; struct zl38005 zl38005_data; static DEFINE_MUTEX(zl38005_list_lock); static LIST_HEAD(zl38005_list); struct zl38005_ioctl_par { unsigned int addr; unsigned int data; }; /* ioctl() calls that are permitted to the /dev/zl38005 interface. */ #define ZL_MAGIC 'G' #define ZL_RD_REG _IOR(ZL_MAGIC, 0x09, struct zl38005_ioctl_par) #define ZL_WR_REG _IOR(ZL_MAGIC, 0x0a, struct zl38005_ioctl_par) static struct zl38005_ioctl_par zl38005_ioctl_par; static int zl38005_rd_ctrl_reg(struct device *dev, u8 *val) { int err; u8 cmd; u8 rx_buf[3]; struct zl38005 *zl38005 = dev_get_drvdata(dev); cmd = (CMD_VALID | CMD_READ | CMD_LEN_16_BIT | CMD_TYPE_CTRL); deb("%s TX cmd=%X\n", cmd); err = spi_write_then_read(zl38005->spi, &cmd, sizeof cmd, rx_buf, 2); if (err < 0) { dev_err(dev, "spi_sync failed with %d\n", err); return err; } *val = rx_buf[1]; return 0; } static int zl38005_wr_ctrl_reg(struct device *dev, u8 val, u8 write) { struct spi_message msg; u8 buf[2]; struct spi_transfer xfer = { .len = 2, .tx_buf = buf, }; int err; u8 cmd = (CMD_VALID | CMD_WRITE | CMD_LEN_8_BIT | CMD_TYPE_CTRL); struct zl38005 *zl38005 = dev_get_drvdata(dev); spi_message_init(&msg); buf[0] = cmd; buf[1] = val | (write == 1 ? CTRL_OP_WR : CTRL_OP_RD) | CTRL_WIDTH_16; deb("buf[0]=%02X\n", buf[0]); deb("buf[1]=%02X\n", buf[1]); spi_message_add_tail(&xfer, &msg); err = spi_sync(zl38005->spi, &msg); if (err < 0) { dev_err(dev, "spi_sync failed with %d\n", err); return err; } return 0; } static int zl38005_wr_addr(struct device *dev, u16 val) { struct spi_message msg; u8 buf[3]; struct spi_transfer xfer = { .len = 3, .tx_buf = buf, }; int err; u8 cmd = (CMD_VALID | CMD_WRITE | CMD_LEN_16_BIT | CMD_TYPE_ADDR_WORD); struct zl38005 *zl38005 = dev_get_drvdata(dev); spi_message_init(&msg); buf[0] = cmd; buf[1] = ((val >> 8) & 0xff); buf[2] = (val & 0xff); deb("buf[0]=%02X\n", buf[0]); deb("buf[1]=%02X\n", buf[1]); deb("buf[2]=%02X\n", buf[2]); spi_message_add_tail(&xfer, &msg); err = spi_sync(zl38005->spi, &msg); if (err < 0) { dev_err(dev, "spi_sync failed with %d\n", err); return err; } return 0; } static int zl38005_wr_data(struct device *dev, u16 val) { struct spi_message msg; u8 buf[5]; struct spi_transfer xfer = { .len = 5, .tx_buf = buf, }; int err; u8 cmd = (CMD_VALID | CMD_WRITE | CMD_LEN_32_BIT | CMD_TYPE_WR_DATA); struct zl38005 *zl38005 = dev_get_drvdata(dev); spi_message_init(&msg); buf[0] = cmd; buf[1] = 0; buf[2] = 0; buf[3] = ((val >> 8) & 0xff); buf[4] = (val & 0xff); deb("buf[0]=%02X\n", buf[0]); deb("buf[1]=%02X\n", buf[1]); deb("buf[2]=%02X\n", buf[2]); deb("buf[3]=%02X\n", buf[3]); deb("buf[4]=%02X\n", buf[4]); spi_message_add_tail(&xfer, &msg); err = spi_sync(zl38005->spi, &msg); if (err < 0) { dev_err(dev, "spi_sync failed with %d\n", err); return err; } return 0; } static int zl38005_rd_data(struct device *dev, char *pval) { int err; u8 cmd = (CMD_VALID | CMD_READ | CMD_LEN_32_BIT | CMD_TYPE_RD_DATA); u8 rx_buf[5]; struct zl38005 *zl38005 = dev_get_drvdata(dev); deb("tx cmd=%X\n", cmd); err = spi_write_then_read(zl38005->spi, &cmd, sizeof cmd, rx_buf, sizeof rx_buf); if (err < 0) { dev_err(dev, "spi_write_then_read failed with %d\n", err); return err; } deb("rx_buf[0]=%X\n", rx_buf[0]); deb("rx_buf[1]=%X\n", rx_buf[1]); deb("rx_buf[2]=%X\n", rx_buf[2]); deb("rx_buf[3]=%X\n", rx_buf[3]); deb("rx_buf[4]=%X\n", rx_buf[4]); memcpy(pval, &rx_buf[3], 2); return 0; } static int zl38005_wr_reg(struct device *dev, u16 addr, u16 val) { int t = 0; u8 ctrl_val = CTRL_PENDING; u8 v; /* See if there is a pending operation */ while (t++ < COUNT) { if (!(zl38005_rd_ctrl_reg(dev, &ctrl_val))) { v = ctrl_val & CTRL_PENDING; if (v == 0) break; } schedule(); } if (t == COUNT) { printk(KERN_ERR "zl38005: TIMEOUT for pending operations\n"); return -1; } /* Write Address */ if (zl38005_wr_addr(dev, addr)) return -ENODEV; /* Write Data */ if (zl38005_wr_data(dev, val)) return -ENODEV; /* Writing Ctrl reg */ if (zl38005_wr_ctrl_reg(dev, CTRL_PENDING, 1)) return -ENODEV; return 0; } static int zl38005_rd_reg(struct device *dev, u16 addr, u8 *pval) { int t = 0; u8 ctrl_val = CTRL_PENDING; u8 v; /* See if there is a pending operation */ while (t++ < COUNT) { if (!(zl38005_rd_ctrl_reg(dev, &ctrl_val))) { v = ctrl_val & CTRL_PENDING; if (v == 0) break; } schedule(); } if (t == COUNT) return -1; /* Write Address */ if (zl38005_wr_addr(dev, addr)) return -ENODEV; /* Writing Ctrl reg */ if (zl38005_wr_ctrl_reg(dev, CTRL_PENDING, 0)) return -ENODEV; /* See if there is a pending operation */ while (t++ < COUNT) { if (!(zl38005_rd_ctrl_reg(dev, &ctrl_val))) { v = ctrl_val & CTRL_PENDING; if (v == 0) break; } schedule(); } if (t == COUNT) return -1; /* Read Data */ if (zl38005_rd_data(dev, pval)) return -ENODEV; return 0; } static int zl38005_spi_port_test(struct device *dev) { int t = 0; u8 buf[2]; zl38005_wr_reg(dev, REG_0403, REG_0403_ACK); while (t++ < COUNT) { zl38005_wr_reg(dev, REG_0402, TX_START); if (!(zl38005_rd_reg(dev, REG_0403, buf)) && (buf[0] == RX_START_OK)) break; } if (t < COUNT) return 0; else return -1; } static int zl38005_open(struct inode *inode, struct file *filp) { if (module_usage_count) { printk(KERN_ERR "zl38005 device alrady opened\n"); return -EBUSY; } module_usage_count++; return 0; } static int zl38005_close(struct inode *inode, struct file *filp) { if (module_usage_count) { module_usage_count--; } return 0; } static int zl38005_ioctl(struct inode *inode, struct file *file, unsigned int cmd, unsigned long arg) { struct zl38005 *zl38005 = &zl38005_data; int ret; u8 buf[2]; switch (cmd) { case ZL_WR_REG: if (copy_from_user(&zl38005_ioctl_par, (struct zl38005_ioctl_par *)arg, sizeof(struct zl38005_ioctl_par))) return -EFAULT; deb("ioctl wr ADDR=%X\n", zl38005_ioctl_par.addr); deb("ioctl wr DATA=%X\n", zl38005_ioctl_par.data); ret = zl38005_wr_reg(&zl38005->spi->dev, zl38005_ioctl_par.addr, zl38005_ioctl_par.data); break; case ZL_RD_REG: if (copy_from_user(&zl38005_ioctl_par, (struct zl38005_ioctl_par *)arg, sizeof(struct zl38005_ioctl_par))) { return -EFAULT; } deb("ioctl wr ADDR=%X\n", zl38005_ioctl_par.addr); ret = zl38005_rd_reg(&zl38005->spi->dev, (u16)zl38005_ioctl_par.addr, buf); zl38005_ioctl_par.data = (buf[0] << 8) | buf[1]; if (!ret) { deb("ioctl rd DATA=%X\n", zl38005_ioctl_par.data); if (copy_to_user((struct zl38005_ioctl_par *)arg, &zl38005_ioctl_par, sizeof(struct zl38005_ioctl_par))) return -EFAULT; } else return -EAGAIN; break; default: printk(KERN_DEBUG "ioctl: Invalid Command Value"); ret = -EINVAL; } return ret; } static const struct file_operations zl38005_fops = { .owner = THIS_MODULE, .open = zl38005_open, .release = zl38005_close, .ioctl = zl38005_ioctl, }; /*-------------------------------------------------------------------------*/ static struct class *zl38005_class; static int zl38005_spi_probe(struct spi_device *spi) { struct zl38005 *zl38005 = &zl38005_data; int err; dev_dbg(&spi->dev, "probing zl38005 spi device\n"); spi->bits_per_word = 8; spi->mode = SPI_MODE_0; err = spi_setup(spi); if (err < 0) return err; /* Initialize the driver data */ zl38005->spi = spi; spi_set_drvdata(spi, zl38005); #if 0 /* Trying to see if zl38005 is live */ if (zl38005_spi_port_test(&spi->dev)) { dev_err(&spi->dev, "ZL38005 spi port test NOT passed\n"); } dev_info(&spi->dev, "ZL38005 spi port test passed\n"); #endif return 0; } static int zl38005_spi_remove(struct spi_device *spi) { device_destroy(zl38005_class, zl38005_data.dev); class_destroy(zl38005_class); kfree(spi_get_drvdata(spi)); return 0; } static struct spi_driver zl38005_spi = { .driver = { .name = "zl38005", .owner = THIS_MODULE, }, .probe = zl38005_spi_probe, .remove = zl38005_spi_remove, }; static int __init zl38005_init(void) { struct zl38005 *zl38005 = &zl38005_data; int result; result = alloc_chrdev_region(&zl38005->dev, 0, 1, "zl38005"); if (result < 0) { printk(KERN_ERR "\nzl38005: Module intialization failed.\ could not register character device"); return -ENODEV; } /* Initialize of character device */ cdev_init(&c_dev, &zl38005_fops); c_dev.owner = THIS_MODULE; c_dev.ops = &zl38005_fops; /* addding character device */ result = cdev_add(&c_dev, zl38005->dev, 1); if (result) { printk("NOTICE \nzl38005:Error %d adding zl38005\ ..error no:", result); unregister_chrdev_region(zl38005->dev, 1); } /* registration of character device */ register_chrdev(MAJOR(zl38005->dev), DRIVER_NAME, &zl38005_fops); zl38005_class = class_create(THIS_MODULE, "zl38005"); if (!zl38005_class) { cdev_del(&c_dev); unregister_chrdev(MAJOR(zl38005->dev), DRIVER_NAME); } device_create(zl38005_class, NULL, zl38005->dev, NULL, "zl38005"); result = spi_register_driver(&zl38005_spi); if (result < 0) { class_destroy(zl38005_class); unregister_chrdev(MAJOR(zl38005->dev), zl38005_spi.driver.name); } return result; } module_init(zl38005_init); static void __exit zl38005_exit(void) { spi_unregister_driver(&zl38005_spi); } module_exit(zl38005_exit); /* Module information */ MODULE_AUTHOR("Raffaele Recalcati <raffaele.recalcati(a)bticino.it>"); MODULE_DESCRIPTION("ZL38005 acoustic echo canceller"); MODULE_LICENSE("GPLv2"); MODULE_VERSION(DRIVER_VERSION); --------------------- drivers/char/zl38005.h ----------------------- #ifndef __ZL38005_H #define __ZL38005_H #define DRIVER_NAME "zl38005" #define CMD_VALID (2<<6) #define CMD_READ (0<<5) #define CMD_WRITE (1<<5) #define CMD_LEN_8_BIT (0<<3) #define CMD_LEN_16_BIT (1<<3) #define CMD_LEN_24_BIT (2<<3) #define CMD_LEN_32_BIT (3<<3) #define CMD_TYPE_CTRL (1<<0) #define CMD_TYPE_ADDR_WORD (2<<0) #define CMD_TYPE_RD_DATA (3<<0) #define CMD_TYPE_WR_DATA (4<<0) #define CTRL_OP_RD (0<<0) #define CTRL_OP_WR (1<<0) #define CTRL_WIDTH_8 (0<<1) #define CTRL_WIDTH_16 (1<<1) #define CTRL_WIDTH_24 (2<<1) #define CTRL_WIDTH_32 (3<<1) #define CTRL_PENDING (1<<3) #define CTRL_MEMORY_SEL (1<<8) #define REG_0402 0x0402 #define REG_0403 0x0403 #define TX_START 0x7e01 #define RX_START_OK 0x5e01 #define REG_0403_ACK 0 // x1 #define COUNT 10 #define ZL38005_RX_BUF_SIZE 4 #define ZL38005_TX_BUF_SIZE 4 #endif /* __ZL38005_H */ -- www.opensurf.it

2 1

[alsa-devel] ALSA SoC on Android problems
by Marco Sinigaglia 24 Jun '11

24 Jun '11

Hi, I am trying to enable the WM9714 attached to a S3C6410 through the AC97 Bus. I am using the Android Gingerbread running the kernel linux-2.6.35. I am not sure if this is the right mailing list, please redirect me if it is not. I can see, on the kernel source, that there is the WM9713 codec but not the WM9714. Is this a problem? I am quite sure that the kernel does not find/setup the WM9714. Assuming that is the kernel that has to configure it, do I have to setup something? I can see that, on linux-2.6.35/sound/soc/codecs/wm9713.c, the wm9713_soc_probe function is not call. Nevertheless, the smdk_wm9713.c does run the smdk_init without errors. The kernel log is following: No device for DAI AD73311 Asahi Kasei AK4104 ALSA SoC Codec Driver No device for DAI AK4535 Cirrus Logic CS4270 ALSA SoC Codec Driver No device for DAI PCM3008 HiFi No device for DAI SSM2602 No device for DAI tlv320aic23 No device for DAI UDA134X No device for DAI WM8510 HiFi No device for DAI WM8728 No device for DAI WM8971 No device for DAI WM8990 ADC/DAC Primary ALSA device list: No soundcards found. Indeed, the user space is complaining: E/ALSALib ( 53): external/alsa-lib/src/control/control.c:902:(snd_ctl_open_noupdate) Invalid CTL AndroidOut W/AudioHardwareALSA( 53): Unable to attach mixer to device AndroidOut: No such file or directory E/AudioHardwareALSA( 53): Unable to attach mixer to device default: No such file or directory E/ALSALib ( 53): external/alsa-lib/src/control/control.c:902:(snd_ctl_open_noupdate) Invalid CTL AndroidIn Kernel config : # CONFIG_SOUND_OSS_CORE is not set CONFIG_SND=y CONFIG_SND_TIMER=y CONFIG_SND_PCM=y CONFIG_SND_JACK=y CONFIG_SND_SEQUENCER=y CONFIG_SND_SEQ_DUMMY=y # CONFIG_SND_MIXER_OSS is not set # CONFIG_SND_PCM_OSS is not set # CONFIG_SND_SEQUENCER_OSS is not set CONFIG_SND_DYNAMIC_MINORS=y CONFIG_SND_SUPPORT_OLD_API=y CONFIG_SND_VERBOSE_PROCFS=y CONFIG_SND_VERBOSE_PRINTK=y CONFIG_SND_DEBUG=y # CONFIG_SND_DEBUG_VERBOSE is not set # CONFIG_SND_PCM_XRUN_DEBUG is not set CONFIG_SND_VMASTER=y # CONFIG_SND_RAWMIDI_SEQ is not set # CONFIG_SND_OPL3_LIB_SEQ is not set # CONFIG_SND_OPL4_LIB_SEQ is not set # CONFIG_SND_SBAWE_SEQ is not set # CONFIG_SND_EMU10K1_SEQ is not set CONFIG_SND_AC97_CODEC=y CONFIG_SND_DRIVERS=y # CONFIG_SND_DUMMY is not set # CONFIG_SND_VIRMIDI is not set # CONFIG_SND_MTPAV is not set # CONFIG_SND_SERIAL_U16550 is not set # CONFIG_SND_MPU401 is not set # CONFIG_SND_AC97_POWER_SAVE is not set CONFIG_SND_ARM=y # CONFIG_SND_SPI is not set CONFIG_SND_SOC=y CONFIG_SND_SOC_AC97_BUS=y CONFIG_SND_S3C24XX_SOC=y CONFIG_SND_S3C_I2SV2_SOC=y CONFIG_SND_S3C64XX_SOC_I2S_V4=y CONFIG_SND_S3C_SOC_AC97=y CONFIG_SND_S3C64XX_SOC_WM8580=y CONFIG_SND_SOC_SMDK_WM9713=y CONFIG_SND_SOC_I2C_AND_SPI=y CONFIG_SND_SOC_ALL_CODECS=y CONFIG_SND_SOC_WM_HUBS=y CONFIG_SND_SOC_AC97_CODEC=y CONFIG_SND_SOC_AD1836=y CONFIG_SND_SOC_AD193X=y CONFIG_SND_SOC_AD1980=y CONFIG_SND_SOC_AD73311=y CONFIG_SND_SOC_ADS117X=y CONFIG_SND_SOC_AK4104=y CONFIG_SND_SOC_AK4535=y CONFIG_SND_SOC_AK4642=y CONFIG_SND_SOC_AK4671=y CONFIG_SND_SOC_CS4270=y CONFIG_SND_SOC_DA7210=y CONFIG_SND_SOC_L3=y CONFIG_SND_SOC_PCM3008=y CONFIG_SND_SOC_SPDIF=y CONFIG_SND_SOC_SSM2602=y CONFIG_SND_SOC_STAC9766=y CONFIG_SND_SOC_TLV320AIC23=y CONFIG_SND_SOC_TLV320AIC26=y CONFIG_SND_SOC_TLV320AIC3X=y CONFIG_SND_SOC_TLV320DAC33=y CONFIG_SND_SOC_UDA134X=y CONFIG_SND_SOC_UDA1380=y CONFIG_SND_SOC_WM8510=y CONFIG_SND_SOC_WM8523=y CONFIG_SND_SOC_WM8580=y CONFIG_SND_SOC_WM8711=y CONFIG_SND_SOC_WM8727=y CONFIG_SND_SOC_WM8728=y CONFIG_SND_SOC_WM8731=y CONFIG_SND_SOC_WM8750=y CONFIG_SND_SOC_WM8753=y CONFIG_SND_SOC_WM8776=y CONFIG_SND_SOC_WM8900=y CONFIG_SND_SOC_WM8903=y CONFIG_SND_SOC_WM8904=y CONFIG_SND_SOC_WM8940=y CONFIG_SND_SOC_WM8955=y CONFIG_SND_SOC_WM8960=y CONFIG_SND_SOC_WM8961=y CONFIG_SND_SOC_WM8971=y CONFIG_SND_SOC_WM8974=y CONFIG_SND_SOC_WM8978=y CONFIG_SND_SOC_WM8988=y CONFIG_SND_SOC_WM8990=y CONFIG_SND_SOC_WM8993=y CONFIG_SND_SOC_WM9081=y CONFIG_SND_SOC_WM9705=y CONFIG_SND_SOC_WM9712=y CONFIG_SND_SOC_WM9713=y CONFIG_SND_SOC_MAX9877=y CONFIG_SND_SOC_TPA6130A2=y CONFIG_SND_SOC_WM2000=y CONFIG_SND_SOC_WM9090=y # CONFIG_SOUND_PRIME is not set CONFIG_AC97_BUS=y CONFIG_HID_SUPPORT=y CONFIG_HID=y # CONFIG_HIDRAW is not set # CONFIG_HID_PID is not set Cheers Marco Member of the CSR plc group of companies. CSR plc registered in England and Wales, registered number 4187346, registered office Churchill House, Cambridge Business Park, Cowley Road, Cambridge, CB4 0WZ, United Kingdom More information can be found at www.csr.com. Follow CSR on Twitter at http://twitter.com/CSR_PLC and read our blog at www.csr.com/blog

2 1

[alsa-devel] [PATCH 1/3] ASoC: Add ADAV80x codec driver
by Lars-Peter Clausen 24 Jun '11

24 Jun '11

This patch adds support for the Analog Devices ADAV801 and ADAV803 audio codec. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- Changes since v1: * Sync register on resume * Don't use arrays for the mux enums and controls * Rework PLL handling. Use DAPM for powering the PLL and OSC down and use set_sysclk to select the system clock instead of doing this implicitly in set_pll. --- MAINTAINERS | 1 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/adav80x.c | 908 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/adav80x.h | 31 ++ 5 files changed, 946 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/adav80x.c create mode 100644 sound/soc/codecs/adav80x.h diff --git a/MAINTAINERS b/MAINTAINERS index a26e9ec..4f7665c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -534,6 +534,7 @@ L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) W: http://wiki.analog.com/ S: Supported F: sound/soc/codecs/adau* +F: sound/soc/codecs/adav* F: sound/soc/codecs/ad1* F: sound/soc/codecs/ssm* diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2998e65..ff43405 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,6 +17,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 + select SND_SOC_ADAV80X select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C @@ -137,6 +138,9 @@ config SND_SOC_ADAU1701 select SIGMA tristate +config SND_SOC_ADAV80X + tristate + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 51cd3d4..4957431 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -5,6 +5,7 @@ snd-soc-ad193x-objs := ad193x.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o +snd-soc-adav80x-objs := adav80x.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -99,6 +100,7 @@ obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o +obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c new file mode 100644 index 0000000..60f5d8c --- /dev/null +++ b/sound/soc/codecs/adav80x.c @@ -0,0 +1,908 @@ +/* + * ADAV80X Audio Codec driver supporting ADAV801, ADAV803 + * + * Copyright 2011 Analog Devices Inc. + * Author: Yi Li <yi.li(a)analog.com> + * Author: Lars-Peter Clausen <lars(a)metafoo.de> + * + * Licensed under the GPL-2 or later. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/kernel.h> +#include <linux/i2c.h> +#include <linux/spi/spi.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/tlv.h> +#include <sound/soc.h> + +#include "adav80x.h" + +#define ADAV80X_PLAYBACK_CTRL 0x04 +#define ADAV80X_AUX_IN_CTRL 0x05 +#define ADAV80X_REC_CTRL 0x06 +#define ADAV80X_AUX_OUT_CTRL 0x07 +#define ADAV80X_DPATH_CTRL1 0x62 +#define ADAV80X_DPATH_CTRL2 0x63 +#define ADAV80X_DAC_CTRL1 0x64 +#define ADAV80X_DAC_CTRL2 0x65 +#define ADAV80X_DAC_CTRL3 0x66 +#define ADAV80X_DAC_L_VOL 0x68 +#define ADAV80X_DAC_R_VOL 0x69 +#define ADAV80X_PGA_L_VOL 0x6c +#define ADAV80X_PGA_R_VOL 0x6d +#define ADAV80X_ADC_CTRL1 0x6e +#define ADAV80X_ADC_CTRL2 0x6f +#define ADAV80X_ADC_L_VOL 0x70 +#define ADAV80X_ADC_R_VOL 0x71 +#define ADAV80X_PLL_CTRL1 0x74 +#define ADAV80X_PLL_CTRL2 0x75 +#define ADAV80X_ICLK_CTRL1 0x76 +#define ADAV80X_ICLK_CTRL2 0x77 +#define ADAV80X_PLL_CLK_SRC 0x78 +#define ADAV80X_PLL_OUTE 0x7a + +#define ADAV80X_PLL_CLK_SRC_PLL_XIN(pll) 0x00 +#define ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll) (0x40 << (pll)) +#define ADAV80X_PLL_CLK_SRC_PLL_MASK(pll) (0x40 << (pll)) + +#define ADAV80X_ICLK_CTRL1_DAC_SRC(src) ((src) << 5) +#define ADAV80X_ICLK_CTRL1_ADC_SRC(src) ((src) << 2) +#define ADAV80X_ICLK_CTRL1_ICLK2_SRC(src) (src) +#define ADAV80X_ICLK_CTRL2_ICLK1_SRC(src) ((src) << 3) + +#define ADAV80X_PLL_CTRL1_PLLDIV 0x10 +#define ADAV80X_PLL_CTRL1_PLLPD(pll) (0x04 << (pll)) +#define ADAV80X_PLL_CTRL1_XTLPD 0x02 + +#define ADAV80X_PLL_CTRL2_FIELD(pll, x) ((x) << ((pll) * 4)) + +#define ADAV80X_PLL_CTRL2_FS_48(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x00) +#define ADAV80X_PLL_CTRL2_FS_32(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x08) +#define ADAV80X_PLL_CTRL2_FS_44(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0c) + +#define ADAV80X_PLL_CTRL2_SEL(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x02) +#define ADAV80X_PLL_CTRL2_DOUB(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x01) +#define ADAV80X_PLL_CTRL2_PLL_MASK(pll) ADAV80X_PLL_CTRL2_FIELD((pll), 0x0f) + +#define ADAV80X_ADC_CTRL1_MODULATOR_MASK 0x80 +#define ADAV80X_ADC_CTRL1_MODULATOR_128FS 0x00 +#define ADAV80X_ADC_CTRL1_MODULATOR_64FS 0x80 + +#define ADAV80X_DAC_CTRL1_PD 0x80 + +#define ADAV80X_DAC_CTRL2_DIV1 0x00 +#define ADAV80X_DAC_CTRL2_DIV1_5 0x10 +#define ADAV80X_DAC_CTRL2_DIV2 0x20 +#define ADAV80X_DAC_CTRL2_DIV3 0x30 +#define ADAV80X_DAC_CTRL2_DIV_MASK 0x30 + +#define ADAV80X_DAC_CTRL2_INTERPOL_256FS 0x00 +#define ADAV80X_DAC_CTRL2_INTERPOL_128FS 0x40 +#define ADAV80X_DAC_CTRL2_INTERPOL_64FS 0x80 +#define ADAV80X_DAC_CTRL2_INTERPOL_MASK 0xc0 + +#define ADAV80X_DAC_CTRL2_DEEMPH_NONE 0x00 +#define ADAV80X_DAC_CTRL2_DEEMPH_44 0x01 +#define ADAV80X_DAC_CTRL2_DEEMPH_32 0x02 +#define ADAV80X_DAC_CTRL2_DEEMPH_48 0x03 +#define ADAV80X_DAC_CTRL2_DEEMPH_MASK 0x01 + +#define ADAV80X_CAPTURE_MODE_MASTER 0x20 +#define ADAV80X_CAPTURE_WORD_LEN24 0x00 +#define ADAV80X_CAPTURE_WORD_LEN20 0x04 +#define ADAV80X_CAPTRUE_WORD_LEN18 0x08 +#define ADAV80X_CAPTURE_WORD_LEN16 0x0c +#define ADAV80X_CAPTURE_WORD_LEN_MASK 0x0c + +#define ADAV80X_CAPTURE_MODE_LEFT_J 0x00 +#define ADAV80X_CAPTURE_MODE_I2S 0x01 +#define ADAV80X_CAPTURE_MODE_RIGHT_J 0x03 +#define ADAV80X_CAPTURE_MODE_MASK 0x03 + +#define ADAV80X_PLAYBACK_MODE_MASTER 0x10 +#define ADAV80X_PLAYBACK_MODE_LEFT_J 0x00 +#define ADAV80X_PLAYBACK_MODE_I2S 0x01 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_24 0x04 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_20 0x05 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_18 0x06 +#define ADAV80X_PLAYBACK_MODE_RIGHT_J_16 0x07 +#define ADAV80X_PLAYBACK_MODE_MASK 0x07 + +static u8 adav80x_default_regs[] = { + 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x02, 0x01, 0x80, 0x26, 0x00, 0x00, + 0x02, 0x40, 0x20, 0x00, 0x09, 0x08, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x04, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0xd1, 0x92, 0xb1, 0x37, + 0x48, 0xd2, 0xfb, 0xca, 0xd2, 0x15, 0xe8, 0x29, 0xb9, 0x6a, 0xda, 0x2b, + 0xb7, 0xc0, 0x11, 0x65, 0x5c, 0xf6, 0xff, 0x8d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xa5, 0x00, 0x00, + 0x00, 0xe8, 0x46, 0xe1, 0x5b, 0xd3, 0x43, 0x77, 0x93, 0xa7, 0x44, 0xee, + 0x32, 0x12, 0xc0, 0x11, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x3f, 0x3f, + 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0x00, 0x1d, 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x00, 0x00, 0x52, 0x00, +}; + +struct adav80x { + enum snd_soc_control_type control_type; + + enum adav80x_clk_src clk_src; + unsigned int sysclk; + enum adav80x_pll_src pll_src; + + unsigned int dai_fmt[2]; + unsigned int rate; + bool deemph; +}; + +static const char *adav80x_mux_text[] = { + "ADC", + "Playback", + "Aux Playback", +}; + +static const unsigned int adav80x_mux_values[] = { + 0, 2, 3, +}; + +#define ADAV80X_MUX_ENUM_DECL(name, reg, shift) \ + SOC_VALUE_ENUM_DOUBLE_DECL(name, reg, shift, 7, \ + ARRAY_SIZE(adav80x_mux_text), adav80x_mux_text, \ + adav80x_mux_values) + +static ADAV80X_MUX_ENUM_DECL(adav80x_aux_capture_enum, ADAV80X_DPATH_CTRL1, 0); +static ADAV80X_MUX_ENUM_DECL(adav80x_capture_enum, ADAV80X_DPATH_CTRL1, 3); +static ADAV80X_MUX_ENUM_DECL(adav80x_dac_enum, ADAV80X_DPATH_CTRL2, 3); + +static const struct snd_kcontrol_new adav80x_aux_capture_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_aux_capture_enum); +static const struct snd_kcontrol_new adav80x_capture_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_capture_enum); +static const struct snd_kcontrol_new adav80x_dac_mux_ctrl = + SOC_DAPM_VALUE_ENUM("Route", adav80x_dac_enum); + +#define ADAV80X_MUX(name, ctrl) \ + SND_SOC_DAPM_VALUE_MUX(name, SND_SOC_NOPM, 0, 0, ctrl) + +static const struct snd_soc_dapm_widget adav80x_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC", NULL, ADAV80X_DAC_CTRL1, 7, 1), + SND_SOC_DAPM_ADC("ADC", NULL, ADAV80X_ADC_CTRL1, 5, 1), + + SND_SOC_DAPM_PGA("Right PGA", ADAV80X_ADC_CTRL1, 0, 1, NULL, 0), + SND_SOC_DAPM_PGA("Left PGA", ADAV80X_ADC_CTRL1, 1, 1, NULL, 0), + + SND_SOC_DAPM_AIF_OUT("AIFOUT", "HiFi Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFIN", "HiFi Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_AIF_OUT("AIFAUXOUT", "Aux Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIFAUXIN", "Aux Playback", 0, SND_SOC_NOPM, 0, 0), + + ADAV80X_MUX("Aux Capture Select", &adav80x_aux_capture_mux_ctrl), + ADAV80X_MUX("Capture Select", &adav80x_capture_mux_ctrl), + ADAV80X_MUX("DAC Select", &adav80x_dac_mux_ctrl), + + SND_SOC_DAPM_INPUT("VINR"), + SND_SOC_DAPM_INPUT("VINL"), + SND_SOC_DAPM_OUTPUT("VOUTR"), + SND_SOC_DAPM_OUTPUT("VOUTL"), + + SND_SOC_DAPM_SUPPLY("SYSCLK", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", ADAV80X_PLL_CTRL1, 2, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2", ADAV80X_PLL_CTRL1, 3, 1, NULL, 0), + SND_SOC_DAPM_SUPPLY("OSC", ADAV80X_PLL_CTRL1, 1, 1, NULL, 0), +}; + +static int adav80x_dapm_sysclk_check(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + const char *clk; + + switch (adav80x->clk_src) { + case ADAV80X_CLK_PLL1: + clk = "PLL1"; + break; + case ADAV80X_CLK_PLL2: + clk = "PLL2"; + break; + case ADAV80X_CLK_XTAL: + clk = "OSC"; + break; + default: + return 0; + } + + return strcmp(source->name, clk) == 0; +} + +static int adav80x_dapm_pll_check(struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_codec *codec = source->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + return adav80x->pll_src == ADAV80X_PLL_SRC_XTAL; +} + + +static const struct snd_soc_dapm_route adav80x_dapm_routes[] = { + { "DAC Select", "ADC", "ADC" }, + { "DAC Select", "Playback", "AIFIN" }, + { "DAC Select", "Aux Playback", "AIFAUXIN" }, + { "DAC", NULL, "DAC Select" }, + + { "Capture Select", "ADC", "ADC" }, + { "Capture Select", "Playback", "AIFIN" }, + { "Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFOUT", NULL, "Capture Select" }, + + { "Aux Capture Select", "ADC", "ADC" }, + { "Aux Capture Select", "Playback", "AIFIN" }, + { "Aux Capture Select", "Aux Playback", "AIFAUXIN" }, + { "AIFAUXOUT", NULL, "Aux Capture Select" }, + + { "VOUTR", NULL, "DAC" }, + { "VOUTL", NULL, "DAC" }, + + { "Left PGA", NULL, "VINL" }, + { "Right PGA", NULL, "VINR" }, + { "ADC", NULL, "Left PGA" }, + { "ADC", NULL, "Right PGA" }, + + { "SYSCLK", NULL, "PLL1", adav80x_dapm_sysclk_check }, + { "SYSCLK", NULL, "PLL2", adav80x_dapm_sysclk_check }, + { "SYSCLK", NULL, "OSC", adav80x_dapm_sysclk_check }, + { "PLL1", NULL, "OSC", adav80x_dapm_pll_check }, + { "PLL2", NULL, "OSC", adav80x_dapm_pll_check }, + + { "ADC", NULL, "SYSCLK" }, + { "DAC", NULL, "SYSCLK" }, + { "AIFOUT", NULL, "SYSCLK" }, + { "AIFAUXOUT", NULL, "SYSCLK" }, + { "AIFIN", NULL, "SYSCLK" }, + { "AIFAUXIN", NULL, "SYSCLK" }, +}; + +static int adav80x_set_deemph(struct snd_soc_codec *codec) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->deemph) { + switch (adav80x->rate) { + case 32000: + val = ADAV80X_DAC_CTRL2_DEEMPH_32; + break; + case 44100: + val = ADAV80X_DAC_CTRL2_DEEMPH_44; + break; + case 48000: + case 64000: + case 88200: + case 96000: + val = ADAV80X_DAC_CTRL2_DEEMPH_48; + break; + default: + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + break; + } + } else { + val = ADAV80X_DAC_CTRL2_DEEMPH_NONE; + } + + return snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DEEMPH_MASK, val); +} + +static int adav80x_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int deemph = ucontrol->value.enumerated.item[0]; + + if (deemph > 1) + return -EINVAL; + + adav80x->deemph = deemph; + + return adav80x_set_deemph(codec); +} + +static int adav80x_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = adav80x->deemph; + return 0; +}; + +static const DECLARE_TLV_DB_SCALE(adav80x_inpga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_MINMAX(adav80x_digital_tlv, -9563, 0); + +static const struct snd_kcontrol_new adav80x_controls[] = { + SOC_DOUBLE_R_TLV("Master Playback Volume", ADAV80X_DAC_L_VOL, + ADAV80X_DAC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + SOC_DOUBLE_R_TLV("Master Capture Volume", ADAV80X_ADC_L_VOL, + ADAV80X_ADC_R_VOL, 0, 0xff, 0, adav80x_digital_tlv), + + SOC_DOUBLE_R_TLV("PGA Capture Volume", ADAV80X_PGA_L_VOL, + ADAV80X_PGA_R_VOL, 0, 0x30, 0, adav80x_inpga_tlv), + + SOC_DOUBLE("Master Playback Switch", ADAV80X_DAC_CTRL1, 0, 1, 1, 0), + SOC_DOUBLE("Master Capture Switch", ADAV80X_ADC_CTRL1, 2, 3, 1, 1), + + SOC_SINGLE("ADC High Pass Filter Switch", ADAV80X_ADC_CTRL1, 6, 1, 0), + + SOC_SINGLE_BOOL_EXT("Playback De-emphasis Switch", 0, + adav80x_get_deemph, adav80x_put_deemph), +}; + +static unsigned int adav80x_port_ctrl_regs[2][2] = { + { ADAV80X_REC_CTRL, ADAV80X_PLAYBACK_CTRL, }, + { ADAV80X_AUX_OUT_CTRL, ADAV80X_AUX_IN_CTRL }, +}; + +static int adav80x_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int capture = 0x00; + unsigned int playback = 0x00; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + capture |= ADAV80X_CAPTURE_MODE_MASTER; + playback |= ADAV80X_PLAYBACK_MODE_MASTER; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + capture |= ADAV80X_CAPTURE_MODE_I2S; + playback |= ADAV80X_PLAYBACK_MODE_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + capture |= ADAV80X_CAPTURE_MODE_LEFT_J; + playback |= ADAV80X_PLAYBACK_MODE_LEFT_J; + break; + case SND_SOC_DAIFMT_RIGHT_J: + capture |= ADAV80X_CAPTURE_MODE_RIGHT_J; + playback |= ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_MODE_MASK | ADAV80X_CAPTURE_MODE_MASTER, + capture); + snd_soc_write(codec, adav80x_port_ctrl_regs[dai->id][1], playback); + + adav80x->dai_fmt[dai->id] = fmt & SND_SOC_DAIFMT_FORMAT_MASK; + + return 0; +} + +static int adav80x_set_adc_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_ADC_CTRL1_MODULATOR_128FS; + else + val = ADAV80X_ADC_CTRL1_MODULATOR_64FS; + + snd_soc_update_bits(codec, ADAV80X_ADC_CTRL1, + ADAV80X_ADC_CTRL1_MODULATOR_MASK, val); + + return 0; +} + +static int adav80x_set_dac_clock(struct snd_soc_codec *codec, + unsigned int sample_rate) +{ + unsigned int val; + + if (sample_rate <= 48000) + val = ADAV80X_DAC_CTRL2_DIV1 | ADAV80X_DAC_CTRL2_INTERPOL_256FS; + else + val = ADAV80X_DAC_CTRL2_DIV2 | ADAV80X_DAC_CTRL2_INTERPOL_128FS; + + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL2, + ADAV80X_DAC_CTRL2_DIV_MASK | ADAV80X_DAC_CTRL2_INTERPOL_MASK, + val); + + return 0; +} + +static int adav80x_set_capture_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + unsigned int val; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_CAPTURE_WORD_LEN16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_CAPTRUE_WORD_LEN18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_CAPTURE_WORD_LEN20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_CAPTURE_WORD_LEN24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][0], + ADAV80X_CAPTURE_WORD_LEN_MASK, val); + + return 0; +} + +static int adav80x_set_playback_pcm_format(struct snd_soc_codec *codec, + struct snd_soc_dai *dai, snd_pcm_format_t format) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int val; + + if (adav80x->dai_fmt[dai->id] != SND_SOC_DAIFMT_RIGHT_J) + return 0; + + switch (format) { + case SNDRV_PCM_FORMAT_S16_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_16; + break; + case SNDRV_PCM_FORMAT_S18_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_18; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_20; + break; + case SNDRV_PCM_FORMAT_S24_LE: + val = ADAV80X_PLAYBACK_MODE_RIGHT_J_24; + break; + default: + break; + } + + snd_soc_update_bits(codec, adav80x_port_ctrl_regs[dai->id][1], + ADAV80X_PLAYBACK_MODE_MASK, val); + + return 0; +} + +static int adav80x_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + + if (rate * 256 != adav80x->sysclk) + return -EINVAL; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + adav80x_set_playback_pcm_format(codec, dai, + params_format(params)); + adav80x_set_dac_clock(codec, rate); + } else { + adav80x_set_capture_pcm_format(codec, dai, + params_format(params)); + adav80x_set_adc_clock(codec, rate); + } + adav80x->rate = rate; + adav80x_set_deemph(codec); + + return 0; +} + +static int adav80x_set_sysclk(struct snd_soc_codec *codec, + int clk_id, unsigned int freq, int dir) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case ADAV80X_CLK_XIN: + case ADAV80X_CLK_XTAL: + case ADAV80X_CLK_MCLKI: + case ADAV80X_CLK_PLL1: + case ADAV80X_CLK_PLL2: + break; + default: + return -EINVAL; + } + + adav80x->sysclk = freq; + + if (adav80x->clk_src != clk_id) { + unsigned int iclk_ctrl1, iclk_ctrl2; + + adav80x->clk_src = clk_id; + if (clk_id == ADAV80X_CLK_XTAL) + clk_id = ADAV80X_CLK_XIN; + + iclk_ctrl1 = ADAV80X_ICLK_CTRL1_DAC_SRC(clk_id) | + ADAV80X_ICLK_CTRL1_ADC_SRC(clk_id) | + ADAV80X_ICLK_CTRL1_ICLK2_SRC(clk_id); + iclk_ctrl2 = ADAV80X_ICLK_CTRL2_ICLK1_SRC(clk_id); + + snd_soc_write(codec, ADAV80X_ICLK_CTRL1, iclk_ctrl1); + snd_soc_write(codec, ADAV80X_ICLK_CTRL2, iclk_ctrl2); + + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int adav80x_set_pll(struct snd_soc_codec *codec, int pll_id, + int source, unsigned int freq_in, unsigned int freq_out) +{ + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + unsigned int pll_ctrl1 = 0; + unsigned int pll_ctrl2 = 0; + unsigned int pll_src; + + switch (source) { + case ADAV80X_PLL_SRC_XTAL: + case ADAV80X_PLL_SRC_XIN: + case ADAV80X_PLL_SRC_MCLKI: + break; + default: + return -EINVAL; + } + + if (!freq_out) + return 0; + + switch (freq_in) { + case 27000000: + break; + case 54000000: + if (source == ADAV80X_PLL_SRC_XIN) { + pll_ctrl1 |= ADAV80X_PLL_CTRL1_PLLDIV; + break; + } + default: + return -EINVAL; + } + + if (freq_out > 12288000) { + pll_ctrl2 |= ADAV80X_PLL_CTRL2_DOUB(pll_id); + freq_out /= 2; + } + + /* freq_out = sample_rate * 256 */ + switch (freq_out) { + case 8192000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_32(pll_id); + break; + case 11289600: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_44(pll_id); + break; + case 12288000: + pll_ctrl2 |= ADAV80X_PLL_CTRL2_FS_48(pll_id); + break; + default: + return -EINVAL; + } + + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL1, ADAV80X_PLL_CTRL1_PLLDIV, + pll_ctrl1); + snd_soc_update_bits(codec, ADAV80X_PLL_CTRL2, + ADAV80X_PLL_CTRL2_PLL_MASK(pll_id), pll_ctrl2); + + if (source != adav80x->pll_src) { + if (source == ADAV80X_PLL_SRC_MCLKI) + pll_src = ADAV80X_PLL_CLK_SRC_PLL_MCLKI(pll_id); + else + pll_src = ADAV80X_PLL_CLK_SRC_PLL_XIN(pll_id); + + snd_soc_update_bits(codec, ADAV80X_PLL_CLK_SRC, + ADAV80X_PLL_CLK_SRC_PLL_MASK(pll_id), pll_src); + + adav80x->pll_src = source; + + snd_soc_dapm_sync(&codec->dapm); + } + + return 0; +} + +static int adav80x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + unsigned int mask = ADAV80X_DAC_CTRL1_PD; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, 0x00); + break; + case SND_SOC_BIAS_OFF: + snd_soc_update_bits(codec, ADAV80X_DAC_CTRL1, mask, mask); + break; + } + + codec->dapm.bias_level = level; + return 0; +} + +/* Enforce the same sample rate on all audio interfaces */ +static int adav80x_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active || !adav80x->rate) + return 0; + + return snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, adav80x->rate, adav80x->rate); +} + +static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + if (!codec->active) + adav80x->rate = 0; +} + +static const struct snd_soc_dai_ops adav80x_dai_ops = { + .set_fmt = adav80x_set_dai_fmt, + .hw_params = adav80x_hw_params, + .startup = adav80x_dai_startup, + .shutdown = adav80x_dai_shutdown, +}; + +#define ADAV80X_PLAYBACK_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_64000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000) + +#define ADAV80X_CAPTURE_RATES (SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000) + +#define ADAV80X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S18_3LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver adav80x_dais[] = { + { + .name = "adav80x-hifi", + .id = 0, + .playback = { + .stream_name = "HiFi Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "HiFi Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, + { + .name = "adav80x-aux", + .id = 1, + .playback = { + .stream_name = "Aux Playback", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_PLAYBACK_RATES, + .formats = ADAV80X_FORMATS, + }, + .capture = { + .stream_name = "Aux Capture", + .channels_min = 2, + .channels_max = 2, + .rates = ADAV80X_CAPTURE_RATES, + .formats = ADAV80X_FORMATS, + }, + .ops = &adav80x_dai_ops, + }, +}; + +static int adav80x_probe(struct snd_soc_codec *codec) +{ + int ret; + struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + + ret = snd_soc_codec_set_cache_io(codec, 7, 9, adav80x->control_type); + if (ret) { + dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); + return ret; + } + + /* Power down SYSCLK output, power down S/PDIF receiver */ + snd_soc_write(codec, ADAV80X_PLL_OUTE, 0x27); + /* Disable DAC zero flag */ + snd_soc_write(codec, ADAV80X_DAC_CTRL3, 0x6); + + return adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); +} + +static int adav80x_suspend(struct snd_soc_codec *codec, pm_message_t state) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static int adav80x_resume(struct snd_soc_codec *codec) +{ + adav80x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + codec->cache_sync = 1; + snd_soc_cache_sync(codec); + + return 0; +} + +static int adav80x_remove(struct snd_soc_codec *codec) +{ + return adav80x_set_bias_level(codec, SND_SOC_BIAS_OFF); +} + +static struct snd_soc_codec_driver adav80x_codec_driver = { + .probe = adav80x_probe, + .remove = adav80x_remove, + .suspend = adav80x_suspend, + .resume = adav80x_resume, + .set_bias_level = adav80x_set_bias_level, + + .set_pll = adav80x_set_pll, + .set_sysclk = adav80x_set_sysclk, + + .reg_word_size = sizeof(u8), + .reg_cache_size = ARRAY_SIZE(adav80x_default_regs), + .reg_cache_default = adav80x_default_regs, + + .controls = adav80x_controls, + .num_controls = ARRAY_SIZE(adav80x_controls), + .dapm_widgets = adav80x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adav80x_dapm_widgets), + .dapm_routes = adav80x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), +}; + +static int __devinit adav80x_bus_probe(struct device *dev, + enum snd_soc_control_type control_type) +{ + struct adav80x *adav80x; + int ret; + + adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); + if (!adav80x) + return -ENOMEM; + + dev_set_drvdata(dev, adav80x); + adav80x->control_type = control_type; + + ret = snd_soc_register_codec(dev, &adav80x_codec_driver, + adav80x_dais, ARRAY_SIZE(adav80x_dais)); + if (ret) + kfree(adav80x); + + return ret; +} + +static int __devexit adav80x_bus_remove(struct device *dev) +{ + snd_soc_unregister_codec(dev); + kfree(dev_get_drvdata(dev)); + return 0; +} + +#if defined(CONFIG_SPI_MASTER) +static int __devinit adav80x_spi_probe(struct spi_device *spi) +{ + return adav80x_bus_probe(&spi->dev, SND_SOC_SPI); +} + +static int __devexit adav80x_spi_remove(struct spi_device *spi) +{ + return adav80x_bus_remove(&spi->dev); +} + +static struct spi_driver adav80x_spi_driver = { + .driver = { + .name = "adav801", + .owner = THIS_MODULE, + }, + .probe = adav80x_spi_probe, + .remove = __devexit_p(adav80x_spi_remove), +}; +#endif + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +static const struct i2c_device_id adav80x_id[] = { + { "adav803", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adav80x_id); + +static int __devinit adav80x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return adav80x_bus_probe(&client->dev, SND_SOC_I2C); +} + +static int __devexit adav80x_i2c_remove(struct i2c_client *client) +{ + return adav80x_bus_remove(&client->dev); +} + +static struct i2c_driver adav80x_i2c_driver = { + .driver = { + .name = "adav803", + .owner = THIS_MODULE, + }, + .probe = adav80x_i2c_probe, + .remove = __devexit_p(adav80x_i2c_remove), + .id_table = adav80x_id, +}; +#endif + +static int __init adav80x_init(void) +{ + int ret = 0; + +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + ret = i2c_add_driver(&adav80x_i2c_driver); + if (ret) + return ret; +#endif + +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&adav80x_spi_driver); +#endif + + return ret; +} +module_init(adav80x_init); + +static void __exit adav80x_exit(void) +{ +#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) + i2c_del_driver(&adav80x_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&adav80x_spi_driver); +#endif +} +module_exit(adav80x_exit); + +MODULE_DESCRIPTION("ASoC ADAV80x driver"); +MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); +MODULE_AUTHOR("Yi Li <yi.li(a)analog.com>>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h new file mode 100644 index 0000000..a13aefe --- /dev/null +++ b/sound/soc/codecs/adav80x.h @@ -0,0 +1,31 @@ +/* + * header file for ADAV80X parts + * + * Copyright 2011 Analog Devices Inc. + * + * Licensed under the GPL-2 or later. + */ + +#ifndef _ADAV80X_H +#define _ADAV80X_H + +enum adav80x_pll_src { + ADAV80X_PLL_SRC_XIN, + ADAV80X_PLL_SRC_XTAL, + ADAV80X_PLL_SRC_MCLKI, +}; + +enum adav80x_pll { + ADAV80X_PLL1 = 0, + ADAV80X_PLL2 = 1, +}; + +enum adav80x_clk_src { + ADAV80X_CLK_XIN = 0, + ADAV80X_CLK_MCLKI = 1, + ADAV80X_CLK_PLL1 = 2, + ADAV80X_CLK_PLL2 = 3, + ADAV80X_CLK_XTAL = 6, +}; + +#endif -- 1.7.2.5

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[alsa-devel] ASoC updates for 3.1
by Mark Brown 24 Jun '11

24 Jun '11

The following changes since commit f192c0ab2420aa59d6fae26564b14adf97456063: ASoC: SMDKV310: Enable SPDIF device (2011-06-10 11:05:09 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.1 Arnaud Patard (2): ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry ASoC: imx: add missing module informations Barry Song (1): ASoC: AD1836: rename suspend/resume funcs Dan Carpenter (1): ASoC: adau1701: signedness bug in adau1701_write() Daniel Mack (1): ASoC: pxa-ssp: Correct check for stream presence Johannes Stezenbach (2): ASoC: add STA32X codec driver ASoC: add WM8782 ADC Codec Driver Lars-Peter Clausen (2): ASoC: Add ADAU1701 codec driver ASoC: Blackfin: Add bf5xx-adau1701 machine driver Liam Girdwood (1): ASoC: dapm - Refactor widget IO functions in preparation for platform widgets. Mark Brown (15): ASoC: Factor out redundant read() functions ASoC: Move register I/O code into a separate file ASoC: Add weak routes for sidetone style paths ASoC: Don't use codec->control_data in bulk write ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus Merge branch 'for-3.0' into for-3.1 ASoC: Fix mismerge with release branch ASoC: Clean up some coding style nits in the bf5xx-i2s-pcm driver ASoC: Add adau* drivers to the Analog Devices file patterns ASoC: Remove adau1701 from SND_SOC_ALL_CODECS due to Sigma dependency ASoC: Fix DAPM sequence run for per-widget I/O methods ASoC: Trigger wm_hubs series update startup off a separate flag ASoC: Allow suppression of series updates on wm_hubs devices ASoC: Add basic WM8918 support Merge branch 'for-3.0' into for-3.1 Mike Frysinger (5): ASoC: AD1836: clean up comment headers ASoC: AD1836: drop unnecessary spi register check ASoC: AD1836: fix intermixed tab/space indentation ASoC: AD1836: fix codec name ASoC: Blackfin: allow SPI for SSM2602 parts Taylor Hutt (1): ASoC: codecs: Max98095: Fix logging of hardware revision. MAINTAINERS | 1 + include/sound/soc-dapm.h | 3 + include/sound/soc.h | 3 +- sound/soc/Makefile | 3 +- sound/soc/blackfin/Kconfig | 14 +- sound/soc/blackfin/Makefile | 2 + sound/soc/blackfin/bf5xx-i2s-pcm.c | 6 +- sound/soc/blackfin/bfin-eval-adau1701.c | 139 ++++++ sound/soc/codecs/Kconfig | 14 +- sound/soc/codecs/Makefile | 6 + sound/soc/codecs/ad1836.c | 78 ++-- sound/soc/codecs/ad1836.h | 21 +- sound/soc/codecs/adau1701.c | 549 ++++++++++++++++++++++ sound/soc/codecs/adau1701.h | 17 + sound/soc/codecs/max98095.c | 8 +- sound/soc/codecs/sta32x.c | 777 +++++++++++++++++++++++++++++++ sound/soc/codecs/sta32x.h | 210 +++++++++ sound/soc/codecs/wm8782.c | 80 ++++ sound/soc/codecs/wm8904.c | 1 + sound/soc/codecs/wm8993.c | 1 + sound/soc/codecs/wm8994.c | 1 + sound/soc/codecs/wm_hubs.c | 5 +- sound/soc/codecs/wm_hubs.h | 2 + sound/soc/imx/Kconfig | 7 - sound/soc/imx/imx-pcm-dma-mx2.c | 2 + sound/soc/imx/imx-ssi.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 4 +- sound/soc/soc-cache.c | 416 ----------------- sound/soc/soc-dapm.c | 142 ++++++- sound/soc/soc-io.c | 396 ++++++++++++++++ 30 files changed, 2399 insertions(+), 511 deletions(-) create mode 100644 sound/soc/blackfin/bfin-eval-adau1701.c create mode 100644 sound/soc/codecs/adau1701.c create mode 100644 sound/soc/codecs/adau1701.h create mode 100644 sound/soc/codecs/sta32x.c create mode 100644 sound/soc/codecs/sta32x.h create mode 100644 sound/soc/codecs/wm8782.c create mode 100644 sound/soc/soc-io.c

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[alsa-devel] ASoC updates for 3.0
by Mark Brown 24 Jun '11

24 Jun '11

The following changes since commit 33195500edf260e8c8809ab9dfc67f50e0ce031f: ASoC: SAMSUNG: Fix the incorrect referencing of I2SCON register (2011-06-10 10:54:46 +0100) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-3.0 Arnaud Patard (2): ASoC: imx: Remove unused Kconfig SND_MXC_SOC_SSI entry ASoC: imx: add missing module informations Daniel Mack (1): ASoC: pxa-ssp: Correct check for stream presence Mark Brown (1): ASoC: Remove unused and about to be broken SND_SOC_CUSTOM I/O bus include/sound/soc.h | 3 +-- sound/soc/imx/Kconfig | 7 ------- sound/soc/imx/imx-pcm-dma-mx2.c | 2 ++ sound/soc/imx/imx-ssi.c | 2 +- sound/soc/pxa/pxa2xx-pcm.c | 4 ++-- sound/soc/soc-cache.c | 3 --- 6 files changed, 6 insertions(+), 15 deletions(-)

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Re: [alsa-devel] [PATCH 01/37] Remove unneeded version.h includes from sound/
by Takashi Iwai 24 Jun '11

24 Jun '11

At Thu, 23 Jun 2011 23:54:40 +0200 (CEST), Jesper Juhl wrote: > > In the sound/ directory there are two files (flagged by 'make > versioncheck'); sound/pci/asihpi/asihpi.c and > sound/soc/codecs/wm8991.c that include linux/version.h although they > don't need it. This patch removes the unneeded includes. > > Signed-off-by: Jesper Juhl <jj(a)chaosbits.net> Thanks, applied to sound git tree now. Takashi > --- > sound/pci/asihpi/asihpi.c | 1 - > sound/soc/codecs/wm8991.c | 1 - > 2 files changed, 0 insertions(+), 2 deletions(-) > > diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c > index 2ca6f4f..e3569bd 100644 > --- a/sound/pci/asihpi/asihpi.c > +++ b/sound/pci/asihpi/asihpi.c > @@ -27,7 +27,6 @@ > #include "hpioctl.h" > > #include <linux/pci.h> > -#include <linux/version.h> > #include <linux/init.h> > #include <linux/jiffies.h> > #include <linux/slab.h> > diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c > index 3c2ee1b..6af23d0 100644 > --- a/sound/soc/codecs/wm8991.c > +++ b/sound/soc/codecs/wm8991.c > @@ -13,7 +13,6 @@ > > #include <linux/module.h> > #include <linux/moduleparam.h> > -#include <linux/version.h> > #include <linux/kernel.h> > #include <linux/init.h> > #include <linux/delay.h> > -- > 1.7.5.2 > > > -- > Jesper Juhl <jj(a)chaosbits.net> http://www.chaosbits.net/ > Don't top-post http://www.catb.org/jargon/html/T/top-post.html > Plain text mails only, please. >

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