diff options
Diffstat (limited to 'sound')
33 files changed, 693 insertions, 231 deletions
diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 6fc0ae90e5b1..fff7753e35c1 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -18,6 +18,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/io.h> +#include <linux/gpio.h> #include <sound/ac97_codec.h> #include <sound/pxa2xx-lib.h> @@ -148,6 +149,8 @@ static inline void pxa_ac97_warm_pxa27x(void) static inline void pxa_ac97_cold_pxa27x(void) { + unsigned int timeout; + GCR &= GCR_COLD_RST; /* clear everything but nCRST */ GCR &= ~GCR_COLD_RST; /* then assert nCRST */ @@ -157,8 +160,10 @@ static inline void pxa_ac97_cold_pxa27x(void) clk_enable(ac97conf_clk); udelay(5); clk_disable(ac97conf_clk); - GCR = GCR_COLD_RST; - udelay(50); + GCR = GCR_COLD_RST | GCR_WARM_RST; + timeout = 100; /* wait for the codec-ready bit to be set */ + while (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR)) && timeout--) + mdelay(1); } #endif @@ -340,8 +345,21 @@ int pxa2xx_ac97_hw_probe(struct platform_device *dev) } if (cpu_is_pxa27x()) { - /* Use GPIO 113 as AC97 Reset on Bulverde */ + /* + * This gpio is needed for a work-around to a bug in the ac97 + * controller during warm reset. The direction and level is set + * here so that it is an output driven high when switching from + * AC97_nRESET alt function to generic gpio. + */ + ret = gpio_request_one(reset_gpio, GPIOF_OUT_INIT_HIGH, + "pxa27x ac97 reset"); + if (ret < 0) { + pr_err("%s: gpio_request_one() failed: %d\n", + __func__, ret); + goto err_conf; + } pxa27x_assert_ac97reset(reset_gpio, 0); + ac97conf_clk = clk_get(&dev->dev, "AC97CONFCLK"); if (IS_ERR(ac97conf_clk)) { ret = PTR_ERR(ac97conf_clk); @@ -384,6 +402,8 @@ EXPORT_SYMBOL_GPL(pxa2xx_ac97_hw_probe); void pxa2xx_ac97_hw_remove(struct platform_device *dev) { + if (cpu_is_pxa27x()) + gpio_free(reset_gpio); GCR |= GCR_ACLINK_OFF; free_irq(IRQ_AC97, NULL); if (ac97conf_clk) { diff --git a/sound/oss/pas2_card.c b/sound/oss/pas2_card.c index dabf8a871dcc..7004e24d209f 100644 --- a/sound/oss/pas2_card.c +++ b/sound/oss/pas2_card.c @@ -333,6 +333,11 @@ static void __init attach_pas_card(struct address_info *hw_config) { char temp[100]; + if (pas_model < 0 || + pas_model >= ARRAY_SIZE(pas_model_names)) { + printk(KERN_ERR "pas2 unrecognized model.\n"); + return; + } sprintf(temp, "%s rev %d", pas_model_names[(int) pas_model], pas_read(0x2789)); diff --git a/sound/pci/au88x0/au88x0_synth.c b/sound/pci/au88x0/au88x0_synth.c index 2805e34bd41d..8bef47311e45 100644 --- a/sound/pci/au88x0/au88x0_synth.c +++ b/sound/pci/au88x0/au88x0_synth.c @@ -58,7 +58,7 @@ static void vortex_wt_setdsout(vortex_t * vortex, u32 wt, int en) if (en) temp |= (1 << (wt & 0x1f)); else - temp &= (1 << ~(wt & 0x1f)); + temp &= ~(1 << (wt & 0x1f)); hwwrite(vortex->mmio, WT_DSREG((wt >= 0x20) ? 1 : 0), temp); } diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 8353c77536ac..b8fb0a5adb9b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -2531,7 +2531,7 @@ static int vmaster_mute_mode_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char * const texts[] = { - "Off", "On", "Follow Master" + "On", "Off", "Follow Master" }; unsigned int index; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 0f3d3db0df71..0b6aebacc56b 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -573,9 +573,12 @@ enum { #define AZX_DCAPS_PM_RUNTIME (1 << 26) /* runtime PM support */ /* quirks for Intel PCH */ -#define AZX_DCAPS_INTEL_PCH \ +#define AZX_DCAPS_INTEL_PCH_NOPM \ (AZX_DCAPS_SCH_SNOOP | AZX_DCAPS_BUFSIZE | \ - AZX_DCAPS_COUNT_LPIB_DELAY | AZX_DCAPS_PM_RUNTIME) + AZX_DCAPS_COUNT_LPIB_DELAY) + +#define AZX_DCAPS_INTEL_PCH \ + (AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_PM_RUNTIME) /* quirks for ATI SB / AMD Hudson */ #define AZX_DCAPS_PRESET_ATI_SB \ @@ -2876,7 +2879,7 @@ static int azx_free(struct azx *chip) azx_notifier_unregister(chip); chip->init_failed = 1; /* to be sure */ - complete(&chip->probe_wait); + complete_all(&chip->probe_wait); if (use_vga_switcheroo(chip)) { if (chip->disabled && chip->bus) @@ -3504,7 +3507,7 @@ static int azx_probe(struct pci_dev *pci, pm_runtime_put_noidle(&pci->dev); dev++; - complete(&chip->probe_wait); + complete_all(&chip->probe_wait); return 0; out_free: @@ -3586,13 +3589,13 @@ static void azx_remove(struct pci_dev *pci) static DEFINE_PCI_DEVICE_TABLE(azx_ids) = { /* CPT */ { PCI_DEVICE(0x8086, 0x1c20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* PBG */ { PCI_DEVICE(0x8086, 0x1d20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Panther Point */ { PCI_DEVICE(0x8086, 0x1e20), - .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, + .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM }, /* Lynx Point */ { PCI_DEVICE(0x8086, 0x8c20), .driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH }, diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 60890bfecc19..dd798c3196ff 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -558,24 +558,12 @@ static int conexant_build_controls(struct hda_codec *codec) return 0; } -#ifdef CONFIG_PM -static int conexant_suspend(struct hda_codec *codec) -{ - snd_hda_shutup_pins(codec); - return 0; -} -#endif - static const struct hda_codec_ops conexant_patch_ops = { .build_controls = conexant_build_controls, .build_pcms = conexant_build_pcms, .init = conexant_init, .free = conexant_free, .set_power_state = conexant_set_power, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; #ifdef CONFIG_SND_HDA_INPUT_BEEP @@ -4405,10 +4393,6 @@ static const struct hda_codec_ops cx_auto_patch_ops = { .init = cx_auto_init, .free = conexant_free, .unsol_event = snd_hda_jack_unsol_event, -#ifdef CONFIG_PM - .suspend = conexant_suspend, -#endif - .reboot_notify = snd_hda_shutup_pins, }; /* diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 0fcfa6f406b8..807a2aa1ff38 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -431,9 +431,11 @@ static void hdmi_init_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (get_wcaps(codec, pin_nid) & AC_WCAP_OUT_AMP) snd_hda_codec_write(codec, pin_nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE); - /* Disable pin out until stream is active*/ + /* Enable pin out: some machines with GM965 gets broken output when + * the pin is disabled or changed while using with HDMI + */ snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, 0); + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT); } static int hdmi_get_channel_count(struct hda_codec *codec, hda_nid_t cvt_nid) @@ -1341,7 +1343,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, struct hdmi_spec *spec = codec->spec; int pin_idx = hinfo_to_pin_index(spec, hinfo); hda_nid_t pin_nid = spec->pins[pin_idx].pin_nid; - int pinctl; bool non_pcm; non_pcm = check_non_pcm_per_cvt(codec, cvt_nid); @@ -1350,11 +1351,6 @@ static int generic_hdmi_playback_pcm_prepare(struct hda_pcm_stream *hinfo, hdmi_setup_audio_infoframe(codec, pin_idx, non_pcm, substream); - pinctl = snd_hda_codec_read(codec, pin_nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, pinctl | PIN_OUT); - return hdmi_setup_stream(codec, cvt_nid, pin_nid, stream_tag, format); } @@ -1374,7 +1370,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, int cvt_idx, pin_idx; struct hdmi_spec_per_cvt *per_cvt; struct hdmi_spec_per_pin *per_pin; - int pinctl; if (hinfo->nid) { cvt_idx = cvt_nid_to_cvt_index(spec, hinfo->nid); @@ -1391,11 +1386,6 @@ static int hdmi_pcm_close(struct hda_pcm_stream *hinfo, return -EINVAL; per_pin = &spec->pins[pin_idx]; - pinctl = snd_hda_codec_read(codec, per_pin->pin_nid, 0, - AC_VERB_GET_PIN_WIDGET_CONTROL, 0); - snd_hda_codec_write(codec, per_pin->pin_nid, 0, - AC_VERB_SET_PIN_WIDGET_CONTROL, - pinctl & ~PIN_OUT); snd_hda_spdif_ctls_unassign(codec, pin_idx); per_pin->chmap_set = false; memset(per_pin->chmap, 0, sizeof(per_pin->chmap)); @@ -1512,7 +1502,7 @@ static int hdmi_chmap_ctl_put(struct snd_kcontrol *kcontrol, ctl_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); substream = snd_pcm_chmap_substream(info, ctl_idx); if (!substream || !substream->runtime) - return -EBADFD; + return 0; /* just for avoiding error from alsactl restore */ switch (substream->runtime->status->state) { case SNDRV_PCM_STATE_OPEN: case SNDRV_PCM_STATE_SETUP: @@ -1691,6 +1681,30 @@ static const struct hda_codec_ops generic_hdmi_patch_ops = { .unsol_event = hdmi_unsol_event, }; +static void intel_haswell_fixup_connect_list(struct hda_codec *codec) +{ + unsigned int vendor_param; + hda_nid_t list[3] = {0x2, 0x3, 0x4}; + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || vendor_param & 0x02) + return; + + /* enable DP1.2 mode */ + vendor_param |= 0x02; + snd_hda_codec_read(codec, 0x08, 0, 0x781, vendor_param); + + vendor_param = snd_hda_codec_read(codec, 0x08, 0, 0xf81, 0); + if (vendor_param == -1 || !(vendor_param & 0x02)) + return; + + /* override 3 pins connection list */ + snd_hda_override_conn_list(codec, 0x05, 3, list); + snd_hda_override_conn_list(codec, 0x06, 3, list); + snd_hda_override_conn_list(codec, 0x07, 3, list); +} + + static int patch_generic_hdmi(struct hda_codec *codec) { struct hdmi_spec *spec; @@ -1700,6 +1714,10 @@ static int patch_generic_hdmi(struct hda_codec *codec) return -ENOMEM; codec->spec = spec; + + if (codec->vendor_id == 0x80862807) + intel_haswell_fixup_connect_list(codec); + if (hdmi_parse_codec(codec) < 0) { codec->spec = NULL; kfree(spec); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 7743775f6abb..f5196277b6e9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4373,6 +4373,7 @@ static int alc_alloc_spec(struct hda_codec *codec, hda_nid_t mixer_nid) if (!spec) return -ENOMEM; codec->spec = spec; + codec->single_adc_amp = 1; spec->mixer_nid = mixer_nid; snd_hda_gen_init(&spec->gen); snd_array_init(&spec->kctls, sizeof(struct snd_kcontrol_new), 32); @@ -5816,6 +5817,9 @@ enum { ALC269_TYPE_ALC269VB, ALC269_TYPE_ALC269VC, ALC269_TYPE_ALC269VD, + ALC269_TYPE_ALC280, + ALC269_TYPE_ALC282, + ALC269_TYPE_ALC284, }; /* @@ -5832,10 +5836,13 @@ static int alc269_parse_auto_config(struct hda_codec *codec) switch (spec->codec_variant) { case ALC269_TYPE_ALC269VA: case ALC269_TYPE_ALC269VC: + case ALC269_TYPE_ALC280: + case ALC269_TYPE_ALC284: ssids = alc269va_ssids; break; case ALC269_TYPE_ALC269VB: case ALC269_TYPE_ALC269VD: + case ALC269_TYPE_ALC282: ssids = alc269_ssids; break; default: @@ -5991,6 +5998,30 @@ static void alc269_fixup_quanta_mute(struct hda_codec *codec, spec->automute_hook = alc269_quanta_automute; } +/* update mute-LED according to the speaker mute state via mic1 VREF pin */ +static void alc269_fixup_mic1_mute_hook(void *private_data, int enabled) +{ + struct hda_codec *codec = private_data; + unsigned int pinval = AC_PINCTL_IN_EN + (enabled ? + AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80); + snd_hda_set_pin_ctl_cache(codec, 0x18, pinval); +} + +static void alc269_fixup_mic1_mute(struct hda_codec *codec, + const struct alc_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + switch (action) { + case ALC_FIXUP_ACT_BUILD: + spec->vmaster_mute.hook = alc269_fixup_mic1_mute_hook; + snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, true); + /* fallthru */ + case ALC_FIXUP_ACT_INIT: + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + break; + } +} + /* update mute-LED according to the speaker mute state via mic2 VREF pin */ static void alc269_fixup_mic2_mute_hook(void *private_data, int enabled) { @@ -6042,6 +6073,7 @@ enum { ALC269_FIXUP_DMIC, ALC269VB_FIXUP_AMIC, ALC269VB_FIXUP_DMIC, + ALC269_FIXUP_MIC1_MUTE_LED, ALC269_FIXUP_MIC2_MUTE_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, @@ -6170,6 +6202,10 @@ static const struct alc_fixup alc269_fixups[] = { { } }, }, + [ALC269_FIXUP_MIC1_MUTE_LED] = { + .type = ALC_FIXUP_FUNC, + .v.func = alc269_fixup_mic1_mute, + }, [ALC269_FIXUP_MIC2_MUTE_LED] = { .type = ALC_FIXUP_FUNC, .v.func = alc269_fixup_mic2_mute, @@ -6214,6 +6250,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x029b, "Acer 1810TZ", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x1025, 0x0349, "Acer AOD260", ALC269_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_MIC2_MUTE_LED), + SND_PCI_QUIRK(0x103c, 0x1972, "HP Pavilion 17", ALC269_FIXUP_MIC1_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), @@ -6369,7 +6406,8 @@ static int patch_alc269(struct hda_codec *codec) alc_auto_parse_customize_define(codec); - if (codec->vendor_id == 0x10ec0269) { + switch (codec->vendor_id) { + case 0x10ec0269: spec->codec_variant = ALC269_TYPE_ALC269VA; switch (alc_get_coef0(codec) & 0x00f0) { case 0x0010: @@ -6394,6 +6432,20 @@ static int patch_alc269(struct hda_codec *codec) goto error; spec->init_hook = alc269_fill_coef; alc269_fill_coef(codec); + break; + + case 0x10ec0280: + case 0x10ec0290: + spec->codec_variant = ALC269_TYPE_ALC280; + break; + case 0x10ec0282: + case 0x10ec0283: + spec->codec_variant = ALC269_TYPE_ALC282; + break; + case 0x10ec0284: + case 0x10ec0292: + spec->codec_variant = ALC269_TYPE_ALC284; + break; } /* automatic parse from the BIOS config */ @@ -6569,8 +6621,8 @@ static void alc861vd_fixup_dallas(struct hda_codec *codec, const struct alc_fixup *fix, int action) { if (action == ALC_FIXUP_ACT_PRE_PROBE) { - snd_hda_override_pin_caps(codec, 0x18, 0x00001714); - snd_hda_override_pin_caps(codec, 0x19, 0x0000171c); + snd_hda_override_pin_caps(codec, 0x18, 0x00000734); + snd_hda_override_pin_caps(codec, 0x19, 0x0000073c); } } @@ -7098,6 +7150,7 @@ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0280, .name = "ALC280", .patch = patch_alc269 }, { .id = 0x10ec0282, .name = "ALC282", .patch = patch_alc269 }, { .id = 0x10ec0283, .name = "ALC283", .patch = patch_alc269 }, + { .id = 0x10ec0284, .name = "ALC284", .patch = patch_alc269 }, { .id = 0x10ec0290, .name = "ALC290", .patch = patch_alc269 }, { .id = 0x10ec0292, .name = "ALC292", .patch = patch_alc269 }, { .id = 0x10ec0861, .rev = 0x100340, .name = "ALC660", diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index df13c0f84899..a86547ca17c8 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1725,7 +1725,7 @@ static const struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1658, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x1659, - "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), + "HP Pavilion dv7", STAC_HP_DV7_4000), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165A, "HP", STAC_92HD83XXX_HP_cNB11_INTQUAD), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x165B, diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 6e02e064d7b4..223c3d9cc69e 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -441,6 +441,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); */ /* status */ #define HDSPM_AES32_wcLock 0x0200000 +#define HDSPM_AES32_wcSync 0x0100000 #define HDSPM_AES32_wcFreq_bit 22 /* (status >> HDSPM_AES32_wcFreq_bit) & 0xF gives WC frequency (cf function HDSPM_bit2freq */ @@ -3467,10 +3468,12 @@ static int hdspm_wc_sync_check(struct hdspm *hdspm) switch (hdspm->io_type) { case AES32: status = hdspm_read(hdspm, HDSPM_statusRegister); - if (status & HDSPM_wcSync) - return 2; - else if (status & HDSPM_wcLock) - return 1; + if (status & HDSPM_AES32_wcLock) { + if (status & HDSPM_AES32_wcSync) + return 2; + else + return 1; + } return 0; break; @@ -4658,6 +4661,7 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, unsigned int status; unsigned int status2; unsigned int timecode; + unsigned int wcLock, wcSync; int pref_syncref; char *autosync_ref; int x; @@ -4751,8 +4755,11 @@ snd_hdspm_proc_read_aes32(struct snd_info_entry * entry, snd_iprintf(buffer, "--- Status:\n"); + wcLock = status & HDSPM_AES32_wcLock; + wcSync = wcLock && (status & HDSPM_AES32_wcSync); + snd_iprintf(buffer, "Word: %s Frequency: %d\n", - (status & HDSPM_AES32_wcLock) ? "Sync " : "No Lock", + (wcLock) ? (wcSync ? "Sync " : "Lock ") : "No Lock", HDSPM_bit2freq((status >> HDSPM_AES32_wcFreq_bit) & 0xF)); for (x = 0; x < 8; x++) { diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index adf397b9d0e6..1d8bb5917594 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -446,15 +446,9 @@ static int arizona_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mode = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mode = 1; - break; case SND_SOC_DAIFMT_I2S: mode = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mode = 3; - break; default: arizona_aif_err(dai, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); @@ -714,7 +708,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, snd_soc_update_bits(codec, ARIZONA_ASYNC_SAMPLE_RATE_1, ARIZONA_ASYNC_SAMPLE_RATE_MASK, sr_val); snd_soc_update_bits(codec, base + ARIZONA_AIF_RATE_CTRL, - ARIZONA_AIF1_RATE_MASK, 8); + ARIZONA_AIF1_RATE_MASK, + 8 << ARIZONA_AIF1_RATE_SHIFT); break; default: arizona_aif_err(dai, "Invalid clock %d\n", dai_priv->clk); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 41dae1ed3b71..4deebeb07177 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -34,15 +34,15 @@ #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 -#define ARIZONA_FLL_SRC_SLIMCLK 2 -#define ARIZONA_FLL_SRC_FLL1 3 -#define ARIZONA_FLL_SRC_FLL2 4 -#define ARIZONA_FLL_SRC_AIF1BCLK 5 -#define ARIZONA_FLL_SRC_AIF2BCLK 6 -#define ARIZONA_FLL_SRC_AIF3BCLK 7 -#define ARIZONA_FLL_SRC_AIF1LRCLK 8 -#define ARIZONA_FLL_SRC_AIF2LRCLK 9 -#define ARIZONA_FLL_SRC_AIF3LRCLK 10 +#define ARIZONA_FLL_SRC_SLIMCLK 3 +#define ARIZONA_FLL_SRC_FLL1 4 +#define ARIZONA_FLL_SRC_FLL2 5 +#define ARIZONA_FLL_SRC_AIF1BCLK 8 +#define ARIZONA_FLL_SRC_AIF2BCLK 9 +#define ARIZONA_FLL_SRC_AIF3BCLK 10 +#define ARIZONA_FLL_SRC_AIF1LRCLK 12 +#define ARIZONA_FLL_SRC_AIF2LRCLK 13 +#define ARIZONA_FLL_SRC_AIF3LRCLK 14 #define ARIZONA_MIXER_VOL_MASK 0x00FE #define ARIZONA_MIXER_VOL_SHIFT 1 diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 4f1127935fdf..ac8742a1f25a 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -474,16 +474,16 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; int gpio_nreset = -EINVAL; - int amutec_eq_bmutec = 0; + bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { gpio_nreset = of_get_named_gpio(codec->dev->of_node, "reset-gpio", 0); - if (!of_get_property(codec->dev->of_node, + if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) - amutec_eq_bmutec = 1; + amutec_eq_bmutec = true; } #endif diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 99bb1c69499e..9811a5478c87 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -737,7 +737,7 @@ static const struct cs42l52_clk_para clk_map_table[] = { static int cs42l52_get_clk(int mclk, int rate) { - int i, ret = 0; + int i, ret = -EINVAL; u_int mclk1, mclk2 = 0; for (i = 0; i < ARRAY_SIZE(clk_map_table); i++) { @@ -749,8 +749,6 @@ static int cs42l52_get_clk(int mclk, int rate) } } } - if (ret > ARRAY_SIZE(clk_map_table)) - return -EINVAL; return ret; } diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index a0791ecf6d95..6361dab48bd1 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -40,6 +40,7 @@ struct cs42l73_private { u32 sysclk; u8 mclksel; u32 mclk; + int shutdwn_delay; }; static const struct reg_default cs42l73_reg_defaults[] = { @@ -588,7 +589,60 @@ static const struct snd_kcontrol_new cs42l73_snd_controls[] = { SOC_ENUM("XSPOUT Mono/Stereo Select", xsp_output_mux_enum), }; +static int cs42l73_spklo_spk_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMD: + /* 150 ms delay between setting PDN and MCLKDIS */ + priv->shutdwn_delay = 150; + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + +static int cs42l73_ear_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMD: + /* 50 ms delay between setting PDN and MCLKDIS */ + if (priv->shutdwn_delay < 50) + priv->shutdwn_delay = 50; + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + + +static int cs42l73_hp_amp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct cs42l73_private *priv = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMD: + /* 30 ms delay between setting PDN and MCLKDIS */ + if (priv->shutdwn_delay < 30) + priv->shutdwn_delay = 30; + break; + default: + pr_err("Invalid event = 0x%x\n", event); + } + return 0; +} + static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("DMICA"), + SND_SOC_DAPM_INPUT("DMICB"), SND_SOC_DAPM_INPUT("LINEINA"), SND_SOC_DAPM_INPUT("LINEINB"), SND_SOC_DAPM_INPUT("MIC1"), @@ -604,9 +658,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { CS42L73_PWRCTL2, 3, 1), SND_SOC_DAPM_AIF_OUT("ASPOUTR", NULL, 0, CS42L73_PWRCTL2, 3, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTL", NULL, 0, - CS42L73_PWRCTL2, 4, 1), - SND_SOC_DAPM_AIF_OUT("VSPOUTR", NULL, 0, + SND_SOC_DAPM_AIF_OUT("VSPINOUT", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_PGA("PGA Left", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -632,8 +684,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_MIXER("ASPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("XSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_MIXER("XSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("VSPL Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("VSPR Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("VSP Output Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_AIF_IN("XSPINL", NULL, 0, CS42L73_PWRCTL2, 0, 1), @@ -649,7 +700,7 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_AIF_IN("ASPINM", NULL, 0, CS42L73_PWRCTL2, 2, 1), - SND_SOC_DAPM_AIF_IN("VSPIN", NULL, 0, + SND_SOC_DAPM_AIF_IN("VSPINOUT", NULL, 0, CS42L73_PWRCTL2, 4, 1), SND_SOC_DAPM_MIXER("HL Left Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), @@ -674,16 +725,20 @@ static const struct snd_soc_dapm_widget cs42l73_dapm_widgets[] = { SND_SOC_DAPM_PGA("SPK DAC", SND_SOC_NOPM, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ESL DAC", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_SWITCH("HP Amp", CS42L73_PWRCTL3, 0, 1, - &hp_amp_ctl), + SND_SOC_DAPM_SWITCH_E("HP Amp", CS42L73_PWRCTL3, 0, 1, + &hp_amp_ctl, cs42l73_hp_amp_event, + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_SWITCH("LO Amp", CS42L73_PWRCTL3, 1, 1, &lo_amp_ctl), - SND_SOC_DAPM_SWITCH("SPK Amp", CS42L73_PWRCTL3, 2, 1, - &spk_amp_ctl), - SND_SOC_DAPM_SWITCH("EAR Amp", CS42L73_PWRCTL3, 3, 1, - &ear_amp_ctl), - SND_SOC_DAPM_SWITCH("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, - &spklo_amp_ctl), + SND_SOC_DAPM_SWITCH_E("SPK Amp", CS42L73_PWRCTL3, 2, 1, + &spk_amp_ctl, cs42l73_spklo_spk_amp_event, + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("EAR Amp", CS42L73_PWRCTL3, 3, 1, + &ear_amp_ctl, cs42l73_ear_amp_event, + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH_E("SPKLO Amp", CS42L73_PWRCTL3, 4, 1, + &spklo_amp_ctl, cs42l73_spklo_spk_amp_event, + SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_OUTPUT("HPOUTA"), SND_SOC_DAPM_OUTPUT("HPOUTB"), @@ -705,7 +760,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"ESL DAC", "ESL-ASP Mono Volume", "ESL Mixer"}, {"ESL DAC", "ESL-XSP Mono Volume", "ESL Mixer"}, - {"ESL DAC", "ESL-VSP Mono Volume", "VSPIN"}, + {"ESL DAC", "ESL-VSP Mono Volume", "VSPINOUT"}, /* Loopback */ {"ESL DAC", "ESL-IP Mono Volume", "Input Left Capture"}, {"ESL DAC", "ESL-IP Mono Volume", "Input Right Capture"}, @@ -727,7 +782,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"SPK DAC", "SPK-ASP Mono Volume", "SPK Mixer"}, {"SPK DAC", "SPK-XSP Mono Volume", "SPK Mixer"}, - {"SPK DAC", "SPK-VSP Mono Volume", "VSPIN"}, + {"SPK DAC", "SPK-VSP Mono Volume", "VSPINOUT"}, /* Loopback */ {"SPK DAC", "SPK-IP Mono Volume", "Input Left Capture"}, {"SPK DAC", "SPK-IP Mono Volume", "Input Right Capture"}, @@ -770,8 +825,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"HL Right Mixer", NULL, "ASPINR"}, {"HL Left Mixer", NULL, "XSPINL"}, {"HL Right Mixer", NULL, "XSPINR"}, - {"HL Left Mixer", NULL, "VSPIN"}, - {"HL Right Mixer", NULL, "VSPIN"}, + {"HL Left Mixer", NULL, "VSPINOUT"}, + {"HL Right Mixer", NULL, "VSPINOUT"}, {"ASPINL", NULL, "ASP Playback"}, {"ASPINM", NULL, "ASP Playback"}, @@ -779,7 +834,7 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"XSPINL", NULL, "XSP Playback"}, {"XSPINM", NULL, "XSP Playback"}, {"XSPINR", NULL, "XSP Playback"}, - {"VSPIN", NULL, "VSP Playback"}, + {"VSPINOUT", NULL, "VSP Playback"}, /* Capture Paths */ {"MIC1", NULL, "MIC1 Bias"}, @@ -795,6 +850,8 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"ADC Left", NULL, "PGA Left"}, {"ADC Right", NULL, "PGA Right"}, + {"DMIC Left", NULL, "DMICA"}, + {"DMIC Right", NULL, "DMICB"}, {"Input Left Capture", "ADC Left Input", "ADC Left"}, {"Input Right Capture", "ADC Right Input", "ADC Right"}, @@ -819,21 +876,18 @@ static const struct snd_soc_dapm_route cs42l73_audio_map[] = { {"XSPOUTR", NULL, "XSPR Output Mixer"}, /* Voice Capture */ - {"VSPL Output Mixer", NULL, "Input Left Capture"}, - {"VSPR Output Mixer", NULL, "Input Left Capture"}, + {"VSP Output Mixer", NULL, "Input Left Capture"}, + {"VSP Output Mixer", NULL, "Input Right Capture"}, - {"VSPOUTL", "VSP-IP Volume", "VSPL Output Mixer"}, - {"VSPOUTR", "VSP-IP Volume", "VSPR Output Mixer"}, + {"VSPINOUT", "VSP-IP Volume", "VSP Output Mixer"}, - {"VSPOUTL", NULL, "VSPL Output Mixer"}, - {"VSPOUTR", NULL, "VSPR Output Mixer"}, + {"VSPINOUT", NULL, "VSP Output Mixer"}, {"ASP Capture", NULL, "ASPOUTL"}, {"ASP Capture", NULL, "ASPOUTR"}, {"XSP Capture", NULL, "XSPOUTL"}, {"XSP Capture", NULL, "XSPOUTR"}, - {"VSP Capture", NULL, "VSPOUTL"}, - {"VSP Capture", NULL, "VSPOUTR"}, + {"VSP Capture", NULL, "VSPINOUT"}, }; struct cs42l73_mclk_div { @@ -1167,6 +1221,14 @@ static int cs42l73_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_OFF: snd_soc_update_bits(codec, CS42L73_PWRCTL1, PDN, 1); + if (cs42l73->shutdwn_delay > 0) { + mdelay(cs42l73->shutdwn_delay); + cs42l73->shutdwn_delay = 0; + } else { + mdelay(15); /* Min amount of time requred to power + * down. + */ + } snd_soc_update_bits(codec, CS42L73_DMMCC, MCLKDIS, 1); break; } diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index d75257d40a49..e19490cfb3a8 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -111,9 +111,9 @@ static struct reg_default lm49453_reg_defs[] = { { 101, 0x00 }, { 102, 0x00 }, { 103, 0x01 }, - { 105, 0x01 }, - { 106, 0x00 }, - { 107, 0x01 }, + { 104, 0x01 }, + { 105, 0x00 }, + { 106, 0x01 }, { 107, 0x00 }, { 108, 0x00 }, { 109, 0x00 }, @@ -163,56 +163,25 @@ static struct reg_default lm49453_reg_defs[] = { { 184, 0x00 }, { 185, 0x00 }, { 186, 0x00 }, - { 189, 0x00 }, + { 187, 0x00 }, { 188, 0x00 }, - { 194, 0x00 }, - { 195, 0x00 }, - { 196, 0x00 }, - { 197, 0x00 }, - { 200, 0x00 }, - { 201, 0x00 }, - { 202, 0x00 }, - { 203, 0x00 }, - { 204, 0x00 }, - { 205, 0x00 }, - { 208, 0x00 }, + { 189, 0x00 }, + { 208, 0x06 }, { 209, 0x00 }, - { 210, 0x00 }, - { 211, 0x00 }, - { 213, 0x00 }, - { 214, 0x00 }, - { 215, 0x00 }, - { 216, 0x00 }, - { 217, 0x00 }, - { 218, 0x00 }, - { 219, 0x00 }, + { 210, 0x08 }, + { 211, 0x54 }, + { 212, 0x14 }, + { 213, 0x0d }, + { 214, 0x0d }, + { 215, 0x14 }, + { 216, 0x60 }, { 221, 0x00 }, { 222, 0x00 }, + { 223, 0x00 }, { 224, 0x00 }, - { 225, 0x00 }, - { 226, 0x00 }, - { 227, 0x00 }, - { 228, 0x00 }, - { 229, 0x00 }, - { 230, 0x13 }, - { 231, 0x00 }, - { 232, 0x80 }, - { 233, 0x0C }, - { 234, 0xDD }, - { 235, 0x00 }, - { 236, 0x04 }, - { 237, 0x00 }, - { 238, 0x00 }, - { 239, 0x00 }, - { 240, 0x00 }, - { 241, 0x00 }, - { 242, 0x00 }, - { 243, 0x00 }, - { 244, 0x00 }, - { 245, 0x00 }, { 248, 0x00 }, { 249, 0x00 }, - { 254, 0x00 }, + { 250, 0x00 }, { 255, 0x00 }, }; @@ -525,36 +494,41 @@ SOC_DAPM_SINGLE("Port2_2 Switch", LM49453_P0_PORT2_TX2_REG, 7, 1, 0), }; /* TLV Declarations */ -static const DECLARE_TLV_DB_SCALE(digital_tlv, -7650, 150, 1); -static const DECLARE_TLV_DB_SCALE(port_tlv, 0, 600, 0); +static const DECLARE_TLV_DB_SCALE(adc_dac_tlv, -7650, 150, 1); +static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 200, 1); +static const DECLARE_TLV_DB_SCALE(port_tlv, -1800, 600, 0); +static const DECLARE_TLV_DB_SCALE(stn_tlv, -7200, 150, 0); static const struct snd_kcontrol_new lm49453_sidetone_mixer_controls[] = { /* Sidetone supports mono only */ SOC_DAPM_SINGLE_TLV("Sidetone ADCL Volume", LM49453_P0_STN_VOL_ADCL_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone ADCR Volume", LM49453_P0_STN_VOL_ADCR_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1L Volume", LM49453_P0_STN_VOL_DMIC1L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC1R Volume", LM49453_P0_STN_VOL_DMIC1R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2L Volume", LM49453_P0_STN_VOL_DMIC2L_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), SOC_DAPM_SINGLE_TLV("Sidetone DMIC2R Volume", LM49453_P0_STN_VOL_DMIC2R_REG, - 0, 0x3F, 0, digital_tlv), + 0, 0x3F, 0, stn_tlv), }; static const struct snd_kcontrol_new lm49453_snd_controls[] = { /* mic1 and mic2 supports mono only */ - SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_ADC_LEVELL_REG, 0, 6, - 0, digital_tlv), - SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_ADC_LEVELR_REG, 0, 6, - 0, digital_tlv), + SOC_SINGLE_TLV("Mic1 Volume", LM49453_P0_MICL_REG, 0, 15, 0, mic_tlv), + SOC_SINGLE_TLV("Mic2 Volume", LM49453_P0_MICR_REG, 0, 15, 0, mic_tlv), + + SOC_SINGLE_TLV("ADCL Volume", LM49453_P0_ADC_LEVELL_REG, 0, 63, + 0, adc_dac_tlv), + SOC_SINGLE_TLV("ADCR Volume", LM49453_P0_ADC_LEVELR_REG, 0, 63, + 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC1 Volume", LM49453_P0_DMIC1_LEVELL_REG, - LM49453_P0_DMIC1_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC1_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DMIC2 Volume", LM49453_P0_DMIC2_LEVELL_REG, - LM49453_P0_DMIC2_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DMIC2_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DAPM_ENUM("Mic2Mode", lm49453_mic2mode_enum), SOC_DAPM_ENUM("DMIC12 SRC", lm49453_dmic12_cfg_enum), @@ -569,16 +543,16 @@ static const struct snd_kcontrol_new lm49453_snd_controls[] = { 2, 1, 0), SOC_DOUBLE_R_TLV("DAC HP Volume", LM49453_P0_DAC_HP_LEVELL_REG, - LM49453_P0_DAC_HP_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HP_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LO Volume", LM49453_P0_DAC_LO_LEVELL_REG, - LM49453_P0_DAC_LO_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LO_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC LS Volume", LM49453_P0_DAC_LS_LEVELL_REG, - LM49453_P0_DAC_LS_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_LS_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_DOUBLE_R_TLV("DAC HA Volume", LM49453_P0_DAC_HA_LEVELL_REG, - LM49453_P0_DAC_HA_LEVELR_REG, 0, 6, 0, digital_tlv), + LM49453_P0_DAC_HA_LEVELR_REG, 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("EP Volume", LM49453_P0_DAC_LS_LEVELL_REG, - 0, 6, 0, digital_tlv), + 0, 63, 0, adc_dac_tlv), SOC_SINGLE_TLV("PORT1_1_RX_LVL Volume", LM49453_P0_PORT1_RX_LVL1_REG, 0, 3, 0, port_tlv), @@ -1218,7 +1192,7 @@ static int lm49453_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) } snd_soc_update_bits(codec, LM49453_P0_AUDIO_PORT1_BASIC_REG, - LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(1)|BIT(5), + LM49453_AUDIO_PORT1_BASIC_FMT_MASK|BIT(0)|BIT(5), (aif_val | mode | clk_phase)); snd_soc_write(codec, LM49453_P0_AUDIO_PORT1_RX_MSB_REG, clk_shift); diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index cb1675cd8e1c..92bbfec9b107 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -401,7 +401,7 @@ static const struct snd_kcontrol_new sgtl5000_snd_controls[] = { 5, 1, 0), SOC_SINGLE_TLV("Mic Volume", SGTL5000_CHIP_MIC_CTRL, - 0, 4, 0, mic_gain_tlv), + 0, 3, 0, mic_gain_tlv), }; /* mute the codec used by alsa core */ @@ -1344,7 +1344,7 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) SGTL5000_HP_ZCD_EN | SGTL5000_ADC_ZCD_EN); - snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 0); + snd_soc_write(codec, SGTL5000_CHIP_MIC_CTRL, 2); /* * disable DAP diff --git a/sound/soc/codecs/sigmadsp.c b/sound/soc/codecs/sigmadsp.c index 5be42bf56996..4068f2491232 100644 --- a/sound/soc/codecs/sigmadsp.c +++ b/sound/soc/codecs/sigmadsp.c @@ -225,7 +225,7 @@ EXPORT_SYMBOL(process_sigma_firmware); static int sigma_action_write_regmap(void *control_data, const struct sigma_action *sa, size_t len) { - return regmap_raw_write(control_data, le16_to_cpu(sa->addr), + return regmap_raw_write(control_data, be16_to_cpu(sa->addr), sa->payload, len - 2); } diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index ab355c4f0b2d..40c07be9b581 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -74,9 +74,10 @@ SNDRV_PCM_FMTBIT_S32_LE) #define S2PC_VALUE 0x98 #define CLOCK_OUT 0x60 -#define LEFT_J_DATA_FORMAT 0x10 -#define I2S_DATA_FORMAT 0x12 -#define RIGHT_J_DATA_FORMAT 0x14 +#define DATA_FORMAT_MSK 0x0E +#define LEFT_J_DATA_FORMAT 0x00 +#define I2S_DATA_FORMAT 0x02 +#define RIGHT_J_DATA_FORMAT 0x04 #define CODEC_MUTE_VAL 0x80 #define POWER_CNTLMSAK 0x40 @@ -289,7 +290,7 @@ static int sta529_set_dai_fmt(struct snd_soc_dai *codec_dai, u32 fmt) return -EINVAL; } - snd_soc_update_bits(codec, STA529_S2PCFG0, 0x0D, mode); + snd_soc_update_bits(codec, STA529_S2PCFG0, DATA_FORMAT_MSK, mode); return 0; } diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8d75aa152c8c..c58bee8346ce 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -398,7 +398,8 @@ static int tpa6130a2_probe(struct i2c_client *client, TPA6130A2_MUTE_L; if (data->power_gpio >= 0) { - ret = gpio_request(data->power_gpio, "tpa6130a2 enable"); + ret = devm_gpio_request(dev, data->power_gpio, + "tpa6130a2 enable"); if (ret < 0) { dev_err(dev, "Failed to request power GPIO (%d)\n", data->power_gpio); @@ -419,16 +420,16 @@ static int tpa6130a2_probe(struct i2c_client *client, break; } - data->supply = regulator_get(dev, regulator); + data->supply = devm_regulator_get(dev, regulator); if (IS_ERR(data->supply)) { ret = PTR_ERR(data->supply); dev_err(dev, "Failed to request supply: %d\n", ret); - goto err_regulator; + goto err_gpio; } ret = tpa6130a2_power(1); if (ret != 0) - goto err_power; + goto err_gpio; /* Read version */ @@ -440,15 +441,10 @@ static int tpa6130a2_probe(struct i2c_client *client, /* Disable the chip */ ret = tpa6130a2_power(0); if (ret != 0) - goto err_power; + goto err_gpio; return 0; -err_power: - regulator_put(data->supply); -err_regulator: - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); err_gpio: tpa6130a2_client = NULL; @@ -457,14 +453,7 @@ err_gpio: static int tpa6130a2_remove(struct i2c_client *client) { - struct tpa6130a2_data *data = i2c_get_clientdata(client); - tpa6130a2_power(0); - - if (data->power_gpio >= 0) - gpio_free(data->power_gpio); - - regulator_put(data->supply); tpa6130a2_client = NULL; return 0; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 1cbe88f01d63..12bcae63a7f0 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -209,9 +209,9 @@ static int wm2000_power_up(struct i2c_client *i2c, int analogue) ret = wm2000_read(i2c, WM2000_REG_SPEECH_CLARITY); if (wm2000->speech_clarity) - ret &= ~WM2000_SPEECH_CLARITY; - else ret |= WM2000_SPEECH_CLARITY; + else + ret &= ~WM2000_SPEECH_CLARITY; wm2000_write(i2c, WM2000_REG_SPEECH_CLARITY, ret); wm2000_write(i2c, WM2000_REG_SYS_START0, 0x33); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index afcf31df77e0..e6cefe1ac677 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1566,15 +1566,9 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: fmt_val = 0; break; - case SND_SOC_DAIFMT_DSP_B: - fmt_val = 1; - break; case SND_SOC_DAIFMT_I2S: fmt_val = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - fmt_val = 3; - break; default: dev_err(codec->dev, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); @@ -1626,7 +1620,7 @@ static int wm2200_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) WM2200_AIF1TX_LRCLK_MSTR | WM2200_AIF1TX_LRCLK_INV, lrclk); snd_soc_update_bits(codec, WM2200_AUDIO_IF_1_5, - WM2200_AIF1_FMT_MASK << 1, fmt_val << 1); + WM2200_AIF1_FMT_MASK, fmt_val); return 0; } diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 5a5f36936235..54397a508073 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -1279,15 +1279,9 @@ static int wm5100_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) case SND_SOC_DAIFMT_DSP_A: mask = 0; break; - case SND_SOC_DAIFMT_DSP_B: - mask = 1; - break; case SND_SOC_DAIFMT_I2S: mask = 2; break; - case SND_SOC_DAIFMT_LEFT_J: - mask = 3; - break; default: dev_err(codec->dev, "Unsupported DAI format %d\n", fmt & SND_SOC_DAIFMT_FORMAT_MASK); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 688ade080589..7a9048dad1cd 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -36,6 +36,9 @@ struct wm5102_priv { struct arizona_priv core; struct arizona_fll fll[2]; + + unsigned int spk_ena:2; + unsigned int spk_ena_pending:1; }; static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); @@ -787,6 +790,47 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), }; +static int wm5102_spk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec); + + if (arizona->rev < 1) + return 0; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (!wm5102->spk_ena) { + snd_soc_write(codec, 0x4f5, 0x25a); + wm5102->spk_ena_pending = true; + } + break; + case SND_SOC_DAPM_POST_PMU: + if (wm5102->spk_ena_pending) { + msleep(75); + snd_soc_write(codec, 0x4f5, 0xda); + wm5102->spk_ena_pending = false; + wm5102->spk_ena++; + } + break; + case SND_SOC_DAPM_PRE_PMD: + wm5102->spk_ena--; + if (!wm5102->spk_ena) + snd_soc_write(codec, 0x4f5, 0x25a); + break; + case SND_SOC_DAPM_POST_PMD: + if (!wm5102->spk_ena) + snd_soc_write(codec, 0x4f5, 0x0da); + break; + } + + return 0; +} + + ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); @@ -1034,10 +1078,10 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index ffc89fab96fb..7b198c38f3ef 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -169,6 +169,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) const struct wm_adsp_region *mem; const char *region_name; char *file, *text; + void *buf; unsigned int reg; int regions = 0; int ret, offset, type, sizes; @@ -322,8 +323,18 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - ret = regmap_raw_write(regmap, reg, region->data, + buf = kmemdup(region->data, le32_to_cpu(region->len), + GFP_KERNEL); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + + ret = regmap_raw_write(regmap, reg, buf, le32_to_cpu(region->len)); + + kfree(buf); + if (ret != 0) { adsp_err(dsp, "%s.%d: Failed to write %d bytes at %d in %s: %d\n", @@ -359,6 +370,7 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) const char *region_name; int ret, pos, blocks, type, offset, reg; char *file; + void *buf; file = kzalloc(PAGE_SIZE, GFP_KERNEL); if (file == NULL) @@ -426,6 +438,13 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) } if (reg) { + buf = kmemdup(blk->data, le32_to_cpu(blk->len), + GFP_KERNEL); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + return -ENOMEM; + } + ret = regmap_raw_write(regmap, reg, blk->data, le32_to_cpu(blk->len)); if (ret != 0) { @@ -433,6 +452,8 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) "%s.%d: Failed to write to %x in %s\n", file, blocks, reg, region_name); } + + kfree(buf); } pos += le32_to_cpu(blk->len) + sizeof(*blk); diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 967d0e173e1b..5fbfb06e8083 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -113,7 +113,7 @@ static int soc_compr_free(struct snd_compr_stream *cstream) SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } else - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } else { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 9c768bcb98a6..2370063b5824 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1255,6 +1255,8 @@ static int soc_post_component_init(struct snd_soc_card *card, INIT_LIST_HEAD(&rtd->dpcm[SNDRV_PCM_STREAM_CAPTURE].fe_clients); ret = device_add(rtd->dev); if (ret < 0) { + /* calling put_device() here to free the rtd->dev */ + put_device(rtd->dev); dev_err(card->dev, "ASoC: failed to register runtime device: %d\n", ret); return ret; @@ -1554,7 +1556,7 @@ static void soc_remove_aux_dev(struct snd_soc_card *card, int num) /* unregister the rtd device */ if (rtd->dev_registered) { device_remove_file(rtd->dev, &dev_attr_codec_reg); - device_del(rtd->dev); + device_unregister(rtd->dev); rtd->dev_registered = 0; } @@ -2917,7 +2919,7 @@ int snd_soc_info_volsw_range(struct snd_kcontrol *kcontrol, platform_max = mc->platform_max; uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; + uinfo->count = snd_soc_volsw_is_stereo(mc) ? 2 : 1; uinfo->value.integer.min = 0; uinfo->value.integer.max = platform_max - min; @@ -2941,12 +2943,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; + int ret; val = ((ucontrol->value.integer.value[0] + min) & mask); if (invert) @@ -2954,7 +2958,21 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, val_mask = mask << shift; val = val << shift; - return snd_soc_update_bits_locked(codec, reg, val_mask, val); + ret = snd_soc_update_bits_locked(codec, reg, val_mask, val); + if (ret != 0) + return ret; + + if (snd_soc_volsw_is_stereo(mc)) { + val = ((ucontrol->value.integer.value[1] + min) & mask); + if (invert) + val = max - val; + val_mask = mask << shift; + val = val << shift; + + ret = snd_soc_update_bits_locked(codec, rreg, val_mask, val); + } + + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_range); @@ -2974,6 +2992,7 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, (struct soc_mixer_control *)kcontrol->private_value; struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int reg = mc->reg; + unsigned int rreg = mc->rreg; unsigned int shift = mc->shift; int min = mc->min; int max = mc->max; @@ -2988,6 +3007,16 @@ int snd_soc_get_volsw_range(struct snd_kcontrol *kcontrol, ucontrol->value.integer.value[0] = ucontrol->value.integer.value[0] - min; + if (snd_soc_volsw_is_stereo(mc)) { + ucontrol->value.integer.value[1] = + (snd_soc_read(codec, rreg) >> shift) & mask; + if (invert) + ucontrol->value.integer.value[1] = + max - ucontrol->value.integer.value[1]; + ucontrol->value.integer.value[1] = + ucontrol->value.integer.value[1] - min; + } + return 0; } EXPORT_SYMBOL_GPL(snd_soc_get_volsw_range); @@ -4155,9 +4184,9 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, ret = of_property_read_string_index(np, propname, 2 * i, &routes[i].sink); if (ret) { - dev_err(card->dev, "ASoC: Property '%s' index %d" - " could not be read: %d\n", propname, 2 * i, - ret); + dev_err(card->dev, + "ASoC: Property '%s' index %d could not be read: %d\n", + propname, 2 * i, ret); kfree(routes); return -EINVAL; } @@ -4165,8 +4194,8 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, (2 * i) + 1, &routes[i].source); if (ret) { dev_err(card->dev, - "ASoC: Property '%s' index %d could not be" - " read: %d\n", propname, (2 * i) + 1, ret); + "ASoC: Property '%s' index %d could not be read: %d\n", + propname, (2 * i) + 1, ret); kfree(routes); return -EINVAL; } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 5c3ca2a34661..cf191e6aebbe 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -334,11 +334,11 @@ static void close_delayed_work(struct work_struct *work) dev_dbg(rtd->dev, "ASoC: pop wq checking: %s status: %s waiting: %s\n", codec_dai->driver->playback.stream_name, codec_dai->playback_active ? "active" : "inactive", - codec_dai->pop_wait ? "yes" : "no"); + rtd->pop_wait ? "yes" : "no"); /* are we waiting on this codec DAI stream */ - if (codec_dai->pop_wait == 1) { - codec_dai->pop_wait = 0; + if (rtd->pop_wait == 1) { + rtd->pop_wait = 0; snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, SND_SOC_DAPM_STREAM_STOP); } @@ -408,7 +408,7 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) SND_SOC_DAPM_STREAM_STOP); } else { /* start delayed pop wq here for playback streams */ - codec_dai->pop_wait = 1; + rtd->pop_wait = 1; schedule_delayed_work(&rtd->delayed_work, msecs_to_jiffies(rtd->pmdown_time)); } @@ -480,8 +480,8 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) /* cancel any delayed stream shutdown that is pending */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && - codec_dai->pop_wait) { - codec_dai->pop_wait = 0; + rtd->pop_wait) { + rtd->pop_wait = 0; cancel_delayed_work(&rtd->delayed_work); } @@ -1243,6 +1243,7 @@ static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_FREE) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) continue; diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index e71fe55cebef..0e2ed3d05c45 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -179,6 +179,15 @@ static struct usbmix_name_map audigy2nx_map[] = { { 0 } /* terminator */ }; +static struct usbmix_selector_map c400_selectors[] = { + { + .id = 0x80, + .count = 2, + .names = (const char*[]) {"Internal", "SPDIF"} + }, + { 0 } /* terminator */ +}; + static struct usbmix_selector_map audigy2nx_selectors[] = { { .id = 14, /* Capture Source */ @@ -367,6 +376,10 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .map = hercules_usb51_map, }, { + .id = USB_ID(0x0763, 0x2030), + .selector_map = c400_selectors, + }, + { .id = USB_ID(0x08bb, 0x2702), .map = linex_map, .ignore_ctl_error = 1, diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 0422b1360af3..15520de1df56 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1206,7 +1206,7 @@ static int snd_c400_create_mixer(struct usb_mixer_interface *mixer) * are valid they presents mono controls as L and R channels of * stereo. So we provide a good mixer here. */ -struct std_mono_table ebox44_table[] = { +static struct std_mono_table ebox44_table[] = { { .unitid = 4, .control = 1, diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index c6593101c049..d82e378d37cb 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -511,6 +511,16 @@ static int configure_sync_endpoint(struct snd_usb_substream *subs) struct snd_usb_substream *sync_subs = &subs->stream->substream[subs->direction ^ 1]; + if (subs->sync_endpoint->type != SND_USB_ENDPOINT_TYPE_DATA || + !subs->stream) + return snd_usb_endpoint_set_params(subs->sync_endpoint, + subs->pcm_format, + subs->channels, + subs->period_bytes, + subs->cur_rate, + subs->cur_audiofmt, + NULL); + /* Try to find the best matching audioformat. */ list_for_each_entry(fp, &sync_subs->fmt_list, list) { int score = match_endpoint_audioformats(fp, subs->cur_audiofmt, diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 49f9af995d7a..64d25a7a4d59 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -50,6 +50,28 @@ } }, +{ + /* Creative BT-D1 */ + USB_DEVICE(0x041e, 0x0005), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .ifnum = 1, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .channels = 2, + .iface = 1, + .altsetting = 1, + .altset_idx = 1, + .endpoint = 0x03, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .attributes = 0, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 48000, + .rate_max = 48000, + } + } +}, + /* Creative/Toshiba Multimedia Center SB-0500 */ { USB_DEVICE(0x041e, 0x3048), @@ -99,6 +121,42 @@ }, /* + * HP Wireless Audio + * When not ignored, causes instability issues for some users, forcing them to + * blacklist the entire module. + */ +{ + USB_DEVICE(0x0424, 0xb832), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Standard Microsystems Corp.", + .product_name = "HP Wireless Audio", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + /* Mixer */ + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Playback */ + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* Capture */ + { + .ifnum = 2, + .type = QUIRK_IGNORE_INTERFACE, + }, + /* HID Device, .ifnum = 3 */ + { + .ifnum = -1, + } + } + } +}, + +/* * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface * class matches do not take effect without an explicit ID match. */ @@ -2231,7 +2289,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 }, - .clock = 0x81, + .clock = 0x80, } }, /* Capture */ @@ -2257,7 +2315,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .rate_table = (unsigned int[]) { 44100, 48000, 88200, 96000 }, - .clock = 0x81, + .clock = 0x80, } }, /* MIDI */ @@ -2885,6 +2943,93 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, + +/* DIGIDESIGN MBOX 2 */ +{ + USB_DEVICE(0x0dba, 0x3000), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "Digidesign", + .product_name = "Mbox 2", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 1, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 2, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 2, + .iface = 2, + .altsetting = 2, + .altset_idx = 1, + .attributes = 0x00, + .endpoint = 0x03, + .ep_attr = USB_ENDPOINT_SYNC_ASYNC, + .maxpacksize = 0x128, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 3, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 4, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 2, + .iface = 4, + .altsetting = 2, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x85, + .ep_attr = USB_ENDPOINT_SYNC_SYNC, + .maxpacksize = 0x128, + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 48000, + .rate_max = 48000, + .nr_rates = 1, + .rate_table = (unsigned int[]) { + 48000 + } + } + }, + { + .ifnum = 5, + .type = QUIRK_IGNORE_INTERFACE + }, + { + .ifnum = 6, + .type = QUIRK_MIDI_MIDIMAN, + .data = &(const struct snd_usb_midi_endpoint_info) { + .out_ep = 0x02, + .out_cables = 0x0001, + .in_ep = 0x81, + .in_interval = 0x01, + .in_cables = 0x0001 + } + }, + { + .ifnum = -1 + } + } + } +}, { /* Tascam US122 MKII - playback-only support */ .match_flags = USB_DEVICE_ID_MATCH_DEVICE, diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 007fcecdf5cd..2c971858d6b7 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -387,11 +387,13 @@ static int snd_usb_fasttrackpro_boot_quirk(struct usb_device *dev) * rules */ err = usb_driver_set_configuration(dev, 2); - if (err < 0) { + if (err < 0) snd_printdd("error usb_driver_set_configuration: %d\n", err); - return -ENODEV; - } + /* Always return an error, so that we stop creating a device + that will just be destroyed and recreated with a new + configuration */ + return -ENODEV; } else snd_printk(KERN_INFO "usb-audio: Fast Track Pro config OK\n"); @@ -497,6 +499,92 @@ static int snd_usb_nativeinstruments_boot_quirk(struct usb_device *dev) return -EAGAIN; } +static void mbox2_setup_48_24_magic(struct usb_device *dev) +{ + u8 srate[3]; + u8 temp[12]; + + /* Choose 48000Hz permanently */ + srate[0] = 0x80; + srate[1] = 0xbb; + srate[2] = 0x00; + + /* Send the magic! */ + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + 0x01, 0x22, 0x0100, 0x0085, &temp, 0x0003); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0085, &srate, 0x0003); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0086, &srate, 0x0003); + snd_usb_ctl_msg(dev, usb_sndctrlpipe(dev, 0), + 0x81, 0xa2, 0x0100, 0x0003, &srate, 0x0003); + return; +} + +/* Digidesign Mbox 2 needs to load firmware onboard + * and driver must wait a few seconds for initialisation. + */ + +#define MBOX2_FIRMWARE_SIZE 646 +#define MBOX2_BOOT_LOADING 0x01 /* Hard coded into the device */ +#define MBOX2_BOOT_READY 0x02 /* Hard coded into the device */ + +static int snd_usb_mbox2_boot_quirk(struct usb_device *dev) +{ + struct usb_host_config *config = dev->actconfig; + int err; + u8 bootresponse[12]; + int fwsize; + int count; + + fwsize = le16_to_cpu(get_cfg_desc(config)->wTotalLength); + + if (fwsize != MBOX2_FIRMWARE_SIZE) { + snd_printk(KERN_ERR "usb-audio: Invalid firmware size=%d.\n", fwsize); + return -ENODEV; + } + + snd_printd("usb-audio: Sending Digidesign Mbox 2 boot sequence...\n"); + + count = 0; + bootresponse[0] = MBOX2_BOOT_LOADING; + while ((bootresponse[0] == MBOX2_BOOT_LOADING) && (count < 10)) { + msleep(500); /* 0.5 second delay */ + snd_usb_ctl_msg(dev, usb_rcvctrlpipe(dev, 0), + /* Control magic - load onboard firmware */ + 0x85, 0xc0, 0x0001, 0x0000, &bootresponse, 0x0012); + if (bootresponse[0] == MBOX2_BOOT_READY) + break; + snd_printd("usb-audio: device not ready, resending boot sequence...\n"); + count++; + } + + if (bootresponse[0] != MBOX2_BOOT_READY) { + snd_printk(KERN_ERR "usb-audio: Unknown bootresponse=%d, or timed out, ignoring device.\n", bootresponse[0]); + return -ENODEV; + } + + snd_printdd("usb-audio: device initialised!\n"); + + err = usb_get_descriptor(dev, USB_DT_DEVICE, 0, + &dev->descriptor, sizeof(dev->descriptor)); + config = dev->actconfig; + if (err < 0) + snd_printd("error usb_get_descriptor: %d\n", err); + + err = usb_reset_configuration(dev); + if (err < 0) + snd_printd("error usb_reset_configuration: %d\n", err); + snd_printdd("mbox2_boot: new boot length = %d\n", + le16_to_cpu(get_cfg_desc(config)->wTotalLength)); + + mbox2_setup_48_24_magic(dev); + + snd_printk(KERN_INFO "usb-audio: Digidesign Mbox 2: 24bit 48kHz"); + + return 0; /* Successful boot */ +} + /* * Setup quirks */ @@ -573,7 +661,6 @@ static int audiophile_skip_setting_quirk(struct snd_usb_audio *chip, return 0; /* keep this altsetting */ } - static int fasttrackpro_skip_setting_quirk(struct snd_usb_audio *chip, int iface, int altno) { @@ -655,6 +742,10 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, case USB_ID(0x0ccd, 0x00b1): /* Terratec Aureon 7.1 USB */ return snd_usb_cm6206_boot_quirk(dev); + case USB_ID(0x0dba, 0x3000): + /* Digidesign Mbox 2 */ + return snd_usb_mbox2_boot_quirk(dev); + case USB_ID(0x133e, 0x0815): /* Access Music VirusTI Desktop */ return snd_usb_accessmusic_boot_quirk(dev); @@ -770,6 +861,17 @@ void snd_usb_endpoint_start_quirk(struct snd_usb_endpoint *ep) if ((le16_to_cpu(ep->chip->dev->descriptor.idVendor) == 0x23ba) && ep->type == SND_USB_ENDPOINT_TYPE_SYNC) ep->skip_packets = 4; + + /* + * M-Audio Fast Track C400 - when packets are not skipped, real world + * latency varies by approx. +/- 50 frames (at 96KHz) each time the + * stream is (re)started. When skipping packets 16 at endpoint start + * up, the real world latency is stable within +/- 1 frame (also + * across power cycles). + */ + if (ep->chip->usb_id == USB_ID(0x0763, 0x2030) && + ep->type == SND_USB_ENDPOINT_TYPE_DATA) + ep->skip_packets = 16; } void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, |