diff options
Diffstat (limited to 'sound')
39 files changed, 393 insertions, 142 deletions
diff --git a/sound/core/compress_offload.c b/sound/core/compress_offload.c index 98969541cbcc..bea523a5d852 100644 --- a/sound/core/compress_offload.c +++ b/sound/core/compress_offload.c @@ -139,6 +139,18 @@ static int snd_compr_open(struct inode *inode, struct file *f) static int snd_compr_free(struct inode *inode, struct file *f) { struct snd_compr_file *data = f->private_data; + struct snd_compr_runtime *runtime = data->stream.runtime; + + switch (runtime->state) { + case SNDRV_PCM_STATE_RUNNING: + case SNDRV_PCM_STATE_DRAINING: + case SNDRV_PCM_STATE_PAUSED: + data->stream.ops->trigger(&data->stream, SNDRV_PCM_TRIGGER_STOP); + break; + default: + break; + } + data->stream.ops->free(&data->stream); kfree(data->stream.runtime->buffer); kfree(data->stream.runtime); @@ -837,7 +849,8 @@ static int snd_compress_dev_disconnect(struct snd_device *device) struct snd_compr *compr; compr = device->device_data; - snd_unregister_device(compr->direction, compr->card, compr->device); + snd_unregister_device(SNDRV_DEVICE_TYPE_COMPRESS, compr->card, + compr->device); return 0; } diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 17f45e8aa89c..e1e9e0c999fe 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -49,6 +49,8 @@ static struct snd_pcm *snd_pcm_get(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->internal) + continue; if (pcm->card == card && pcm->device == device) return pcm; } @@ -60,6 +62,8 @@ static int snd_pcm_next(struct snd_card *card, int device) struct snd_pcm *pcm; list_for_each_entry(pcm, &snd_pcm_devices, list) { + if (pcm->internal) + continue; if (pcm->card == card && pcm->device > device) return pcm->device; else if (pcm->card->number > card->number) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 445ca481d8d3..bf578ba2677e 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,6 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, { 0x54524108, 0xffffffff, "TR28028", patch_tritech_tr28028, NULL }, // added by xin jin [07/09/99] { 0x54524123, 0xffffffff, "TR28602", NULL, NULL }, // only guess --jk [TR28023 = eMicro EM28023 (new CT1297)] +{ 0x54584e03, 0xffffffff, "TLV320AIC27", NULL, NULL }, { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 5b6c4e3c92ca..748c6a941963 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -4864,8 +4864,8 @@ static void hda_power_work(struct work_struct *work) spin_unlock(&codec->power_lock); state = hda_call_codec_suspend(codec, true); - codec->pm_down_notified = 0; - if (!bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { + if (!codec->pm_down_notified && + !bus->power_keep_link_on && (state & AC_PWRST_CLK_STOP_OK)) { codec->pm_down_notified = 1; hda_call_pm_notify(bus, false); } diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index ac41e9cdc976..b7c89dff7066 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -3531,7 +3531,7 @@ static int create_capture_mixers(struct hda_codec *codec) if (!multi) err = create_single_cap_vol_ctl(codec, n, vol, sw, inv_dmic); - else if (!multi_cap_vol) + else if (!multi_cap_vol && !inv_dmic) err = create_bind_cap_vol_ctl(codec, n, vol, sw); else err = create_multi_cap_vol_ctl(codec); @@ -4475,9 +4475,11 @@ int snd_hda_gen_build_controls(struct hda_codec *codec) true, &spec->vmaster_mute.sw_kctl); if (err < 0) return err; - if (spec->vmaster_mute.hook) + if (spec->vmaster_mute.hook) { snd_hda_add_vmaster_hook(codec, &spec->vmaster_mute, spec->vmaster_mute_enum); + snd_hda_sync_vmaster_hook(&spec->vmaster_mute); + } } free_kctls(spec); /* no longer needed */ diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 0cbdd87dde6d..2aa2f579b4d6 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -968,6 +968,15 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, } } +static void ad1884_fixup_thinkpad(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct ad198x_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->gen.keep_eapd_on = 1; +} + /* set magic COEFs for dmic */ static const struct hda_verb ad1884_dmic_init_verbs[] = { {0x01, AC_VERB_SET_COEF_INDEX, 0x13f7}, @@ -979,6 +988,7 @@ enum { AD1884_FIXUP_AMP_OVERRIDE, AD1884_FIXUP_HP_EAPD, AD1884_FIXUP_DMIC_COEF, + AD1884_FIXUP_THINKPAD, AD1884_FIXUP_HP_TOUCHSMART, }; @@ -997,6 +1007,12 @@ static const struct hda_fixup ad1884_fixups[] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, }, + [AD1884_FIXUP_THINKPAD] = { + .type = HDA_FIXUP_FUNC, + .v.func = ad1884_fixup_thinkpad, + .chained = true, + .chain_id = AD1884_FIXUP_DMIC_COEF, + }, [AD1884_FIXUP_HP_TOUCHSMART] = { .type = HDA_FIXUP_VERBS, .v.verbs = ad1884_dmic_init_verbs, @@ -1008,7 +1024,7 @@ static const struct hda_fixup ad1884_fixups[] = { static const struct snd_pci_quirk ad1884_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x2a82, "HP Touchsmart", AD1884_FIXUP_HP_TOUCHSMART), SND_PCI_QUIRK_VENDOR(0x103c, "HP", AD1884_FIXUP_HP_EAPD), - SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_DMIC_COEF), + SND_PCI_QUIRK_VENDOR(0x17aa, "Lenovo Thinkpad", AD1884_FIXUP_THINKPAD), {} }; diff --git a/sound/pci/hda/patch_cirrus.c b/sound/pci/hda/patch_cirrus.c index b524f89a1f13..18d972501585 100644 --- a/sound/pci/hda/patch_cirrus.c +++ b/sound/pci/hda/patch_cirrus.c @@ -111,6 +111,9 @@ enum { /* 0x0009 - 0x0014 -> 12 test regs */ /* 0x0015 - visibility reg */ +/* Cirrus Logic CS4208 */ +#define CS4208_VENDOR_NID 0x24 + /* * Cirrus Logic CS4210 * @@ -223,6 +226,16 @@ static const struct hda_verb cs_coef_init_verbs[] = { {} /* terminator */ }; +static const struct hda_verb cs4208_coef_init_verbs[] = { + {0x01, AC_VERB_SET_POWER_STATE, 0x00}, /* AFG: D0 */ + {0x24, AC_VERB_SET_PROC_STATE, 0x01}, /* VPW: processing on */ + {0x24, AC_VERB_SET_COEF_INDEX, 0x0033}, + {0x24, AC_VERB_SET_PROC_COEF, 0x0001}, /* A1 ICS */ + {0x24, AC_VERB_SET_COEF_INDEX, 0x0034}, + {0x24, AC_VERB_SET_PROC_COEF, 0x1C01}, /* A1 Enable, A Thresh = 300mV */ + {} /* terminator */ +}; + /* Errata: CS4207 rev C0/C1/C2 Silicon * * http://www.cirrus.com/en/pubs/errata/ER880C3.pdf @@ -295,6 +308,8 @@ static int cs_init(struct hda_codec *codec) /* init_verb sequence for C0/C1/C2 errata*/ snd_hda_sequence_write(codec, cs_errata_init_verbs); snd_hda_sequence_write(codec, cs_coef_init_verbs); + } else if (spec->vendor_nid == CS4208_VENDOR_NID) { + snd_hda_sequence_write(codec, cs4208_coef_init_verbs); } snd_hda_gen_init(codec); @@ -434,6 +449,29 @@ static const struct hda_pintbl mba42_pincfgs[] = { {} /* terminator */ }; +static const struct hda_pintbl mba6_pincfgs[] = { + { 0x10, 0x032120f0 }, /* HP */ + { 0x11, 0x500000f0 }, + { 0x12, 0x90100010 }, /* Speaker */ + { 0x13, 0x500000f0 }, + { 0x14, 0x500000f0 }, + { 0x15, 0x770000f0 }, + { 0x16, 0x770000f0 }, + { 0x17, 0x430000f0 }, + { 0x18, 0x43ab9030 }, /* Mic */ + { 0x19, 0x770000f0 }, + { 0x1a, 0x770000f0 }, + { 0x1b, 0x770000f0 }, + { 0x1c, 0x90a00090 }, + { 0x1d, 0x500000f0 }, + { 0x1e, 0x500000f0 }, + { 0x1f, 0x500000f0 }, + { 0x20, 0x500000f0 }, + { 0x21, 0x430000f0 }, + { 0x22, 0x430000f0 }, + {} /* terminator */ +}; + static void cs420x_fixup_gpio_13(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -556,22 +594,23 @@ static int patch_cs420x(struct hda_codec *codec) /* * CS4208 support: - * Its layout is no longer compatible with CS4206/CS4207, and the generic - * parser seems working fairly well, except for trivial fixups. + * Its layout is no longer compatible with CS4206/CS4207 */ enum { + CS4208_MBA6, CS4208_GPIO0, }; static const struct hda_model_fixup cs4208_models[] = { { .id = CS4208_GPIO0, .name = "gpio0" }, + { .id = CS4208_MBA6, .name = "mba6" }, {} }; static const struct snd_pci_quirk cs4208_fixup_tbl[] = { /* codec SSID */ - SND_PCI_QUIRK(0x106b, 0x7100, "MacBookPro 6,1", CS4208_GPIO0), - SND_PCI_QUIRK(0x106b, 0x7200, "MacBookPro 6,2", CS4208_GPIO0), + SND_PCI_QUIRK(0x106b, 0x7100, "MacBookAir 6,1", CS4208_MBA6), + SND_PCI_QUIRK(0x106b, 0x7200, "MacBookAir 6,2", CS4208_MBA6), {} /* terminator */ }; @@ -588,18 +627,35 @@ static void cs4208_fixup_gpio0(struct hda_codec *codec, } static const struct hda_fixup cs4208_fixups[] = { + [CS4208_MBA6] = { + .type = HDA_FIXUP_PINS, + .v.pins = mba6_pincfgs, + .chained = true, + .chain_id = CS4208_GPIO0, + }, [CS4208_GPIO0] = { .type = HDA_FIXUP_FUNC, .v.func = cs4208_fixup_gpio0, }, }; +/* correct the 0dB offset of input pins */ +static void cs4208_fix_amp_caps(struct hda_codec *codec, hda_nid_t adc) +{ + unsigned int caps; + + caps = query_amp_caps(codec, adc, HDA_INPUT); + caps &= ~(AC_AMPCAP_OFFSET); + caps |= 0x02; + snd_hda_override_amp_caps(codec, adc, HDA_INPUT, caps); +} + static int patch_cs4208(struct hda_codec *codec) { struct cs_spec *spec; int err; - spec = cs_alloc_spec(codec, 0); /* no specific w/a */ + spec = cs_alloc_spec(codec, CS4208_VENDOR_NID); if (!spec) return -ENOMEM; @@ -609,6 +665,12 @@ static int patch_cs4208(struct hda_codec *codec) cs4208_fixups); snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_PRE_PROBE); + snd_hda_override_wcaps(codec, 0x18, + get_wcaps(codec, 0x18) | AC_WCAP_STEREO); + cs4208_fix_amp_caps(codec, 0x18); + cs4208_fix_amp_caps(codec, 0x1b); + cs4208_fix_amp_caps(codec, 0x1c); + err = cs_parse_auto_config(codec); if (err < 0) goto error; diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 4edd2d0f9a3c..ec68eaea0336 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -3231,6 +3231,7 @@ enum { CXT_FIXUP_INC_MIC_BOOST, CXT_FIXUP_HEADPHONE_MIC_PIN, CXT_FIXUP_HEADPHONE_MIC, + CXT_FIXUP_GPIO1, }; static void cxt_fixup_stereo_dmic(struct hda_codec *codec, @@ -3375,6 +3376,15 @@ static const struct hda_fixup cxt_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cxt_fixup_headphone_mic, }, + [CXT_FIXUP_GPIO1] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + { 0x01, AC_VERB_SET_GPIO_MASK, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DIRECTION, 0x01 }, + { 0x01, AC_VERB_SET_GPIO_DATA, 0x01 }, + { } + }, + }, }; static const struct snd_pci_quirk cxt5051_fixups[] = { @@ -3384,6 +3394,7 @@ static const struct snd_pci_quirk cxt5051_fixups[] = { static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x1025, 0x0543, "Acer Aspire One 522", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_GPIO1), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), SND_PCI_QUIRK(0x17aa, 0x215e, "Lenovo T410", CXT_PINCFG_LENOVO_TP410), diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3d8cd04455a6..50173d412ac5 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -937,6 +937,14 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, } /* + * always configure channel mapping, it may have been changed by the + * user in the meantime + */ + hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, + channels, per_pin->chmap, + per_pin->chmap_set); + + /* * sizeof(ai) is used instead of sizeof(*hdmi_ai) or * sizeof(*dp_ai) to avoid partial match/update problems when * the user switches between HDMI/DP monitors. @@ -947,20 +955,10 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, "pin=%d channels=%d\n", pin_nid, channels); - hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap, - per_pin->chmap_set); hdmi_stop_infoframe_trans(codec, pin_nid); hdmi_fill_audio_infoframe(codec, pin_nid, ai.bytes, sizeof(ai)); hdmi_start_infoframe_trans(codec, pin_nid); - } else { - /* For non-pcm audio switch, setup new channel mapping - * accordingly */ - if (per_pin->non_pcm != non_pcm) - hdmi_setup_channel_mapping(codec, pin_nid, non_pcm, ca, - channels, per_pin->chmap, - per_pin->chmap_set); } per_pin->non_pcm = non_pcm; @@ -1149,32 +1147,43 @@ static int hdmi_choose_cvt(struct hda_codec *codec, } static void haswell_config_cvts(struct hda_codec *codec, - int pin_id, int mux_id) + hda_nid_t pin_nid, int mux_idx) { struct hdmi_spec *spec = codec->spec; - struct hdmi_spec_per_pin *per_pin; - int pin_idx, mux_idx; - int curr; - int err; + hda_nid_t nid, end_nid; + int cvt_idx, curr; + struct hdmi_spec_per_cvt *per_cvt; - for (pin_idx = 0; pin_idx < spec->num_pins; pin_idx++) { - per_pin = get_pin(spec, pin_idx); + /* configure all pins, including "no physical connection" ones */ + end_nid = codec->start_nid + codec->num_nodes; + for (nid = codec->start_nid; nid < end_nid; nid++) { + unsigned int wid_caps = get_wcaps(codec, nid); + unsigned int wid_type = get_wcaps_type(wid_caps); + + if (wid_type != AC_WID_PIN) + continue; - if (pin_idx == pin_id) + if (nid == pin_nid) continue; - curr = snd_hda_codec_read(codec, per_pin->pin_nid, 0, + curr = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_CONNECT_SEL, 0); + if (curr != mux_idx) + continue; - /* Choose another unused converter */ - if (curr == mux_id) { - err = hdmi_choose_cvt(codec, pin_idx, NULL, &mux_idx); - if (err < 0) - return; - snd_printdd("HDMI: choose converter %d for pin %d\n", mux_idx, pin_idx); - snd_hda_codec_write_cache(codec, per_pin->pin_nid, 0, + /* choose an unassigned converter. The conveters in the + * connection list are in the same order as in the codec. + */ + for (cvt_idx = 0; cvt_idx < spec->num_cvts; cvt_idx++) { + per_cvt = get_cvt(spec, cvt_idx); + if (!per_cvt->assigned) { + snd_printdd("choose cvt %d for pin nid %d\n", + cvt_idx, nid); + snd_hda_codec_write_cache(codec, nid, 0, AC_VERB_SET_CONNECT_SEL, - mux_idx); + cvt_idx); + break; + } } } } @@ -1216,7 +1225,7 @@ static int hdmi_pcm_open(struct hda_pcm_stream *hinfo, /* configure unused pins to choose other converters */ if (is_haswell(codec)) - haswell_config_cvts(codec, pin_idx, mux_idx); + haswell_config_cvts(codec, per_pin->pin_nid, mux_idx); snd_hda_spdif_ctls_assign(codec, pin_idx, per_cvt->cvt_nid); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index bc07d369fac4..8ad554312b69 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2819,6 +2819,15 @@ static void alc269_fixup_hweq(struct hda_codec *codec, alc_write_coef_idx(codec, 0x1e, coef | 0x80); } +static void alc269_fixup_headset_mic(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) + spec->parse_flags |= HDA_PINCFG_HEADSET_MIC; +} + static void alc271_fixup_dmic(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3439,6 +3448,9 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, /* Set to manual mode */ val = alc_read_coef_idx(codec, 0x06); alc_write_coef_idx(codec, 0x06, val & ~0x000c); + /* Enable Line1 input control by verb */ + val = alc_read_coef_idx(codec, 0x1a); + alc_write_coef_idx(codec, 0x1a, val | (1 << 4)); break; } } @@ -3493,6 +3505,15 @@ static void alc282_fixup_asus_tx300(struct hda_codec *codec, } } +static void alc290_fixup_mono_speakers(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) + /* Remove DAC node 0x03, as it seems to be + giving mono output */ + snd_hda_override_wcaps(codec, 0x03, 0); +} + enum { ALC269_FIXUP_SONY_VAIO, ALC275_FIXUP_SONY_VAIO_GPIO2, @@ -3504,6 +3525,7 @@ enum { ALC271_FIXUP_DMIC, ALC269_FIXUP_PCM_44K, ALC269_FIXUP_STEREO_DMIC, + ALC269_FIXUP_HEADSET_MIC, ALC269_FIXUP_QUANTA_MUTE, ALC269_FIXUP_LIFEBOOK, ALC269_FIXUP_AMIC, @@ -3516,9 +3538,11 @@ enum { ALC269_FIXUP_HP_GPIO_LED, ALC269_FIXUP_INV_DMIC, ALC269_FIXUP_LENOVO_DOCK, + ALC286_FIXUP_SONY_MIC_NO_PRESENCE, ALC269_FIXUP_PINCFG_NO_HP_TO_LINEOUT, ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASUS_X101_FUNC, @@ -3531,6 +3555,8 @@ enum { ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, ALC282_FIXUP_ASUS_TX300, + ALC283_FIXUP_INT_MIC, + ALC290_FIXUP_MONO_SPEAKERS, }; static const struct hda_fixup alc269_fixups[] = { @@ -3599,6 +3625,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_stereo_dmic, }, + [ALC269_FIXUP_HEADSET_MIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_headset_mic, + }, [ALC269_FIXUP_QUANTA_MUTE] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_quanta_mute, @@ -3708,6 +3738,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC }, + [ALC269_FIXUP_DELL3_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1a, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, [ALC269_FIXUP_HEADSET_MODE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode, @@ -3716,6 +3755,15 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_headset_mode_no_hp_mic, }, + [ALC286_FIXUP_SONY_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x18, 0x01a1913c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MIC + }, [ALC269_FIXUP_ASUS_X101_FUNC] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_x101_headset_mic, @@ -3790,6 +3838,22 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc282_fixup_asus_tx300, }, + [ALC283_FIXUP_INT_MIC] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + {0x20, AC_VERB_SET_COEF_INDEX, 0x1a}, + {0x20, AC_VERB_SET_PROC_COEF, 0x0011}, + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST + }, + [ALC290_FIXUP_MONO_SPEAKERS] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc290_fixup_mono_speakers, + .chained = true, + .chain_id = ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -3831,6 +3895,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0608, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0609, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x15cc, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x15cd, "Dell X5 Precision", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), @@ -3853,6 +3918,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x8398, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x83ce, "ASUS P1005", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x8516, "ASUS X101CH", ALC269_FIXUP_ASUS_X101), + SND_PCI_QUIRK(0x104d, 0x90b6, "Sony VAIO Pro 13", ALC286_FIXUP_SONY_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x104d, 0x9073, "Sony VAIO", ALC275_FIXUP_SONY_VAIO_GPIO2), SND_PCI_QUIRK(0x104d, 0x907b, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), SND_PCI_QUIRK(0x104d, 0x9084, "Sony VAIO", ALC275_FIXUP_SONY_HWEQ), @@ -3874,7 +3940,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x2214, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x2215, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), + SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), SND_PCI_QUIRK(0x17aa, 0x5026, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x5109, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), @@ -3938,6 +4004,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_STEREO_DMIC, .name = "alc269-dmic"}, {.id = ALC271_FIXUP_DMIC, .name = "alc271-dmic"}, {.id = ALC269_FIXUP_INV_DMIC, .name = "inv-dmic"}, + {.id = ALC269_FIXUP_HEADSET_MIC, .name = "headset-mic"}, {.id = ALC269_FIXUP_LENOVO_DOCK, .name = "lenovo-dock"}, {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, @@ -4555,6 +4622,8 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x05d8, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05db, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), + SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_ASUS_MODE4), + SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_ASUS_MODE4), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_FIXUP_ASUS_MODE2), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 4f255dfee450..f59a321a6d6a 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -4845,6 +4845,7 @@ static int snd_hdsp_hwdep_ioctl(struct snd_hwdep *hw, struct file *file, unsigne if ((err = hdsp_get_iobox_version(hdsp)) < 0) return err; } + memset(&hdsp_version, 0, sizeof(hdsp_version)); hdsp_version.io_type = hdsp->io_type; hdsp_version.firmware_rev = hdsp->firmware_rev; if ((err = copy_to_user(argp, &hdsp_version, sizeof(hdsp_version)))) diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c index c02405cc007d..5810a0603f2f 100644 --- a/sound/soc/blackfin/bf6xx-i2s.c +++ b/sound/soc/blackfin/bf6xx-i2s.c @@ -88,6 +88,7 @@ static int bfin_i2s_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_FORMAT_S8: param.spctl |= 0x70; sport->wdsize = 1; + break; case SNDRV_PCM_FORMAT_S16_LE: param.spctl |= 0xf0; sport->wdsize = 2; diff --git a/sound/soc/cirrus/ep93xx-pcm.c b/sound/soc/cirrus/ep93xx-pcm.c index 0e9f56e0d4b2..cfe517e68009 100644 --- a/sound/soc/cirrus/ep93xx-pcm.c +++ b/sound/soc/cirrus/ep93xx-pcm.c @@ -57,9 +57,22 @@ static bool ep93xx_pcm_dma_filter(struct dma_chan *chan, void *filter_param) return false; } +static struct dma_chan *ep93xx_compat_request_channel( + struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_substream *substream) +{ + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + return snd_dmaengine_pcm_request_channel(ep93xx_pcm_dma_filter, + dma_data); +} + static const struct snd_dmaengine_pcm_config ep93xx_dmaengine_pcm_config = { .pcm_hardware = &ep93xx_pcm_hardware, .compat_filter_fn = ep93xx_pcm_dma_filter, + .compat_request_channel = ep93xx_compat_request_channel, .prealloc_buffer_size = 131072, }; diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 4633e51b1500..75d0ad5d2dcb 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -309,6 +309,9 @@ static int snd_soc_put_volsw_2r_st(struct snd_kcontrol *kcontrol, val = ucontrol->value.integer.value[0]; val2 = ucontrol->value.integer.value[1]; + if (val >= ARRAY_SIZE(st_table) || val2 >= ARRAY_SIZE(st_table)) + return -EINVAL; + err = snd_soc_update_bits(codec, reg, 0x3f, st_table[val].m); if (err < 0) return err; diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index b8ba0adacfce..80555d7551e6 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -1225,13 +1225,18 @@ static int anc_status_control_put(struct snd_kcontrol *kcontrol, struct ab8500_codec_drvdata *drvdata = dev_get_drvdata(codec->dev); struct device *dev = codec->dev; bool apply_fir, apply_iir; - int req, status; + unsigned int req; + int status; dev_dbg(dev, "%s: Enter.\n", __func__); mutex_lock(&drvdata->anc_lock); req = ucontrol->value.integer.value[0]; + if (req >= ARRAY_SIZE(enum_anc_state)) { + status = -EINVAL; + goto cleanup; + } if (req != ANC_APPLY_FIR_IIR && req != ANC_APPLY_FIR && req != ANC_APPLY_IIR) { dev_err(dev, "%s: ERROR: Unsupported status to set '%s'!\n", diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 2d0378709702..687565d08d9c 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -257,7 +257,7 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - snd_soc_write(codec, SG_SL1, PMMP | MGAIN0); + snd_soc_update_bits(codec, SG_SL1, PMMP | MGAIN0, PMMP | MGAIN0); snd_soc_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); snd_soc_write(codec, ALC_CTL1, ALC | LMTH0); snd_soc_update_bits(codec, PW_MGMT1, PMADL, PMADL); diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 41cdd1642970..8dbcacd44e6a 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1863,7 +1863,7 @@ static int max98095_put_eq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_eq_channel(kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_eq_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; @@ -2016,7 +2016,7 @@ static int max98095_put_bq_enum(struct snd_kcontrol *kcontrol, struct max98095_pdata *pdata = max98095->pdata; int channel = max98095_get_bq_channel(codec, kcontrol->id.name); struct max98095_cdata *cdata; - int sel = ucontrol->value.integer.value[0]; + unsigned int sel = ucontrol->value.integer.value[0]; struct max98095_biquad_cfg *coef_set; int fs, best, best_val, i; int regmask, regsave; diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 651ce0923675..c91eba504f92 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -270,7 +270,7 @@ MODULE_DEVICE_TABLE(of, pcm1681_dt_ids); static const struct regmap_config pcm1681_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = ARRAY_SIZE(pcm1681_reg_defaults) + 1, + .max_register = 0x13, .reg_defaults = pcm1681_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm1681_reg_defaults), .writeable_reg = pcm1681_writeable_reg, diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 2a8eccf64c76..7613181123fe 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -188,7 +188,7 @@ MODULE_DEVICE_TABLE(of, pcm1792a_of_match); static const struct regmap_config pcm1792a_regmap = { .reg_bits = 8, .val_bits = 8, - .max_register = 24, + .max_register = 23, .reg_defaults = pcm1792a_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm1792a_reg_defaults), .writeable_reg = pcm1792a_writeable_reg, diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6e3f269243e0..64ad84d8a306 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -674,6 +674,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Left Input */ {"Left Line1L Mux", "single-ended", "LINE1L"}, {"Left Line1L Mux", "differential", "LINE1L"}, + {"Left Line1R Mux", "single-ended", "LINE1R"}, + {"Left Line1R Mux", "differential", "LINE1R"}, {"Left Line2L Mux", "single-ended", "LINE2L"}, {"Left Line2L Mux", "differential", "LINE2L"}, @@ -690,6 +692,8 @@ static const struct snd_soc_dapm_route intercon[] = { /* Right Input */ {"Right Line1R Mux", "single-ended", "LINE1R"}, {"Right Line1R Mux", "differential", "LINE1R"}, + {"Right Line1L Mux", "single-ended", "LINE1L"}, + {"Right Line1L Mux", "differential", "LINE1L"}, {"Right Line2R Mux", "single-ended", "LINE2R"}, {"Right Line2R Mux", "differential", "LINE2R"}, diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b38f3506418f..60b6b593c407 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1062,6 +1062,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].dm); region->len -= be32_to_cpu(adsp1_alg[i].dm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region DM with ID %x\n", @@ -1079,6 +1080,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp1_alg[i + 1].zm); region->len -= be32_to_cpu(adsp1_alg[i].zm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", @@ -1108,6 +1110,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].xm); region->len -= be32_to_cpu(adsp2_alg[i].xm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region XM with ID %x\n", @@ -1125,6 +1128,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].ym); region->len -= be32_to_cpu(adsp2_alg[i].ym); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region YM with ID %x\n", @@ -1142,6 +1146,7 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) if (i + 1 < algs) { region->len = be32_to_cpu(adsp2_alg[i + 1].zm); region->len -= be32_to_cpu(adsp2_alg[i].zm); + region->len *= 4; wm_adsp_create_control(dsp, region); } else { adsp_warn(dsp, "Missing length info for region ZM with ID %x\n", diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 8b50e5958de5..01daf655e20b 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -530,6 +530,7 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w, hubs->hp_startup_mode); break; } + break; case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(codec, WM8993_CHARGE_PUMP_1, diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c6b743978d5e..6b81d0ce2c44 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -936,7 +936,7 @@ static int fsl_ssi_probe(struct platform_device *pdev) ssi_private->ssi_phys = res.start; ssi_private->irq = irq_of_parse_and_map(np, 0); - if (ssi_private->irq == NO_IRQ) { + if (ssi_private->irq == 0) { dev_err(&pdev->dev, "no irq for node %s\n", np->full_name); return -ENXIO; } diff --git a/sound/soc/fsl/imx-mc13783.c b/sound/soc/fsl/imx-mc13783.c index a3d60d4bea4c..a2fd7321b5a9 100644 --- a/sound/soc/fsl/imx-mc13783.c +++ b/sound/soc/fsl/imx-mc13783.c @@ -112,7 +112,7 @@ static int imx_mc13783_probe(struct platform_device *pdev) return ret; } - if (machine_is_mx31_3ds()) { + if (machine_is_mx31_3ds() || machine_is_mx31moboard()) { imx_audmux_v2_configure_port(MX31_AUDMUX_PORT4_SSI_PINS_4, IMX_AUDMUX_V2_PTCR_SYN, IMX_AUDMUX_V2_PDCR_RXDSEL(MX31_AUDMUX_PORT1_SSI0) | diff --git a/sound/soc/fsl/imx-pcm-dma.c b/sound/soc/fsl/imx-pcm-dma.c index 4dc1296688e9..aee23077080a 100644 --- a/sound/soc/fsl/imx-pcm-dma.c +++ b/sound/soc/fsl/imx-pcm-dma.c @@ -25,12 +25,10 @@ static bool filter(struct dma_chan *chan, void *param) { - struct snd_dmaengine_dai_dma_data *dma_data = param; - if (!imx_dma_is_general_purpose(chan)) return false; - chan->private = dma_data->filter_data; + chan->private = param; return true; } diff --git a/sound/soc/fsl/imx-pcm-fiq.c b/sound/soc/fsl/imx-pcm-fiq.c index 34043c55f2a6..10e330514ed8 100644 --- a/sound/soc/fsl/imx-pcm-fiq.c +++ b/sound/soc/fsl/imx-pcm-fiq.c @@ -39,8 +39,6 @@ struct imx_pcm_runtime_data { unsigned int period; int periods; unsigned long offset; - unsigned long last_offset; - unsigned long size; struct hrtimer hrt; int poll_time_ns; struct snd_pcm_substream *substream; @@ -52,9 +50,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) struct imx_pcm_runtime_data *iprtd = container_of(hrt, struct imx_pcm_runtime_data, hrt); struct snd_pcm_substream *substream = iprtd->substream; - struct snd_pcm_runtime *runtime = substream->runtime; struct pt_regs regs; - unsigned long delta; if (!atomic_read(&iprtd->running)) return HRTIMER_NORESTART; @@ -66,19 +62,7 @@ static enum hrtimer_restart snd_hrtimer_callback(struct hrtimer *hrt) else iprtd->offset = regs.ARM_r9 & 0xffff; - /* How much data have we transferred since the last period report? */ - if (iprtd->offset >= iprtd->last_offset) - delta = iprtd->offset - iprtd->last_offset; - else - delta = runtime->buffer_size + iprtd->offset - - iprtd->last_offset; - - /* If we've transferred at least a period then report it and - * reset our poll time */ - if (delta >= iprtd->period) { - snd_pcm_period_elapsed(substream); - iprtd->last_offset = iprtd->offset; - } + snd_pcm_period_elapsed(substream); hrtimer_forward_now(hrt, ns_to_ktime(iprtd->poll_time_ns)); @@ -95,11 +79,9 @@ static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime = substream->runtime; struct imx_pcm_runtime_data *iprtd = runtime->private_data; - iprtd->size = params_buffer_bytes(params); iprtd->periods = params_periods(params); - iprtd->period = params_period_bytes(params) ; + iprtd->period = params_period_bytes(params); iprtd->offset = 0; - iprtd->last_offset = 0; iprtd->poll_time_ns = 1000000000 / params_rate(params) * params_period_size(params); snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c index 46c5b4fdfc52..ca1be1d9dcf0 100644 --- a/sound/soc/fsl/imx-sgtl5000.c +++ b/sound/soc/fsl/imx-sgtl5000.c @@ -62,7 +62,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) struct device_node *ssi_np, *codec_np; struct platform_device *ssi_pdev; struct i2c_client *codec_dev; - struct imx_sgtl5000_data *data; + struct imx_sgtl5000_data *data = NULL; int int_port, ext_port; int ret; @@ -128,7 +128,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) goto fail; } - data->codec_clk = devm_clk_get(&codec_dev->dev, NULL); + data->codec_clk = clk_get(&codec_dev->dev, NULL); if (IS_ERR(data->codec_clk)) { ret = PTR_ERR(data->codec_clk); goto fail; @@ -172,6 +172,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) return 0; fail: + if (data && !IS_ERR(data->codec_clk)) + clk_put(data->codec_clk); if (ssi_np) of_node_put(ssi_np); if (codec_np) @@ -185,6 +187,7 @@ static int imx_sgtl5000_remove(struct platform_device *pdev) struct imx_sgtl5000_data *data = platform_get_drvdata(pdev); snd_soc_unregister_card(&data->card); + clk_put(data->codec_clk); return 0; } diff --git a/sound/soc/fsl/imx-ssi.c b/sound/soc/fsl/imx-ssi.c index f58bcd85c07f..57d6941676ff 100644 --- a/sound/soc/fsl/imx-ssi.c +++ b/sound/soc/fsl/imx-ssi.c @@ -600,19 +600,17 @@ static int imx_ssi_probe(struct platform_device *pdev) ssi->fiq_params.dma_params_rx = &ssi->dma_params_rx; ssi->fiq_params.dma_params_tx = &ssi->dma_params_tx; - ret = imx_pcm_fiq_init(pdev, &ssi->fiq_params); - if (ret) - goto failed_pcm_fiq; + ssi->fiq_init = imx_pcm_fiq_init(pdev, &ssi->fiq_params); + ssi->dma_init = imx_pcm_dma_init(pdev); - ret = imx_pcm_dma_init(pdev); - if (ret) - goto failed_pcm_dma; + if (ssi->fiq_init && ssi->dma_init) { + ret = ssi->fiq_init; + goto failed_pcm; + } return 0; -failed_pcm_dma: - imx_pcm_fiq_exit(pdev); -failed_pcm_fiq: +failed_pcm: snd_soc_unregister_component(&pdev->dev); failed_register: release_mem_region(res->start, resource_size(res)); @@ -628,8 +626,11 @@ static int imx_ssi_remove(struct platform_device *pdev) struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); struct imx_ssi *ssi = platform_get_drvdata(pdev); - imx_pcm_dma_exit(pdev); - imx_pcm_fiq_exit(pdev); + if (!ssi->dma_init) + imx_pcm_dma_exit(pdev); + + if (!ssi->fiq_init) + imx_pcm_fiq_exit(pdev); snd_soc_unregister_component(&pdev->dev); diff --git a/sound/soc/fsl/imx-ssi.h b/sound/soc/fsl/imx-ssi.h index fb1616ba8c59..560c40fc9ebb 100644 --- a/sound/soc/fsl/imx-ssi.h +++ b/sound/soc/fsl/imx-ssi.h @@ -211,6 +211,8 @@ struct imx_ssi { struct imx_dma_data filter_data_rx; struct imx_pcm_fiq_params fiq_params; + int fiq_init; + int dma_init; int enabled; }; diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index daa78a0095fa..4a07f7179690 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -1,6 +1,6 @@ config SND_OMAP_SOC tristate "SoC Audio for the Texas Instruments OMAP chips" - depends on (ARCH_OMAP && DMA_OMAP) || (ARCH_ARM && COMPILE_TEST) + depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST) select SND_DMAENGINE_PCM config SND_OMAP_SOC_DMIC @@ -26,7 +26,7 @@ config SND_OMAP_SOC_N810 config SND_OMAP_SOC_RX51 tristate "SoC Audio support for Nokia RX-51" - depends on SND_OMAP_SOC && ARCH_ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) + depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) select SND_OMAP_SOC_MCBSP select SND_SOC_TLV320AIC3X select SND_SOC_TPA6130A2 diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index b302f3b7a587..3e08b6c0f7ba 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -702,13 +702,6 @@ static int i2s_hw_params(struct snd_pcm_substream *substream, } writel(mod, i2s->addr + I2SMOD); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - snd_soc_dai_set_dma_data(dai, substream, - (void *)&i2s->dma_playback); - else - snd_soc_dai_set_dma_data(dai, substream, - (void *)&i2s->dma_capture); - i2s->frmclk = params_rate(params); return 0; @@ -970,6 +963,8 @@ static int samsung_i2s_dai_probe(struct snd_soc_dai *dai) } clk_prepare_enable(i2s->clk); + snd_soc_dai_init_dma_data(dai, &i2s->dma_playback, &i2s->dma_capture); + if (other) { other->addr = i2s->addr; other->clk = i2s->clk; diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 9cc6986a8cfb..5dd87f4c919e 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -220,8 +220,8 @@ int rsnd_gen_path_exit(struct rsnd_priv *priv, void __iomem *rsnd_gen_reg_get(struct rsnd_priv *priv, struct rsnd_mod *mod, enum rsnd_reg reg); -#define rsnd_is_gen1(s) ((s)->info->flags & RSND_GEN1) -#define rsnd_is_gen2(s) ((s)->info->flags & RSND_GEN2) +#define rsnd_is_gen1(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN1) +#define rsnd_is_gen2(s) (((s)->info->flags & RSND_GEN_MASK) == RSND_GEN2) /* * R-Car ADG diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4d0561312f3b..1a38be0d0ca8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1380,7 +1380,6 @@ static int soc_probe_link_dais(struct snd_soc_card *card, int num, int order) return -ENODEV; list_add(&cpu_dai->dapm.list, &card->dapm_list); - snd_soc_dapm_new_dai_widgets(&cpu_dai->dapm, cpu_dai); } if (cpu_dai->driver->probe) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c17c14c394df..b2949aed1ac2 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1949,7 +1949,7 @@ static ssize_t dapm_widget_power_read_file(struct file *file, w->active ? "active" : "inactive"); list_for_each_entry(p, &w->sources, list_sink) { - if (p->connected && !p->connected(w, p->sink)) + if (p->connected && !p->connected(w, p->source)) continue; if (p->connect) @@ -3495,6 +3495,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->playback.stream_name); + return -ENOMEM; } w->priv = dai; @@ -3513,6 +3514,7 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->capture.stream_name); + return -ENOMEM; } w->priv = dai; diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index e29ec3cd84b1..6ad4c7a47f5d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -25,7 +25,7 @@ #include <sound/dmaengine_pcm.h> struct dmaengine_pcm { - struct dma_chan *chan[SNDRV_PCM_STREAM_CAPTURE + 1]; + struct dma_chan *chan[SNDRV_PCM_STREAM_LAST + 1]; const struct snd_dmaengine_pcm_config *config; struct snd_soc_platform platform; unsigned int flags; @@ -36,6 +36,15 @@ static struct dmaengine_pcm *soc_platform_to_pcm(struct snd_soc_platform *p) return container_of(p, struct dmaengine_pcm, platform); } +static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, + struct snd_pcm_substream *substream) +{ + if (!pcm->chan[substream->stream]) + return NULL; + + return pcm->chan[substream->stream]->device->dev; +} + /** * snd_dmaengine_pcm_prepare_slave_config() - Generic prepare_slave_config callback * @substream: PCM substream @@ -75,12 +84,21 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); struct dma_chan *chan = snd_dmaengine_pcm_get_chan(substream); + int (*prepare_slave_config)(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct dma_slave_config *slave_config); struct dma_slave_config slave_config; int ret; - if (pcm->config->prepare_slave_config) { - ret = pcm->config->prepare_slave_config(substream, params, - &slave_config); + memset(&slave_config, 0, sizeof(slave_config)); + + if (!pcm->config) + prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; + else + prepare_slave_config = pcm->config->prepare_slave_config; + + if (prepare_slave_config) { + ret = prepare_slave_config(substream, params, &slave_config); if (ret) return ret; @@ -92,28 +110,54 @@ static int dmaengine_pcm_hw_params(struct snd_pcm_substream *substream, return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); } -static int dmaengine_pcm_open(struct snd_pcm_substream *substream) +static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct device *dma_dev = dmaengine_dma_dev(pcm, substream); struct dma_chan *chan = pcm->chan[substream->stream]; + struct snd_dmaengine_dai_dma_data *dma_data; + struct dma_slave_caps dma_caps; + struct snd_pcm_hardware hw; int ret; - ret = snd_soc_set_runtime_hwparams(substream, + if (pcm->config && pcm->config->pcm_hardware) + return snd_soc_set_runtime_hwparams(substream, pcm->config->pcm_hardware); - if (ret) - return ret; - return snd_dmaengine_pcm_open(substream, chan); + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + memset(&hw, 0, sizeof(hw)); + hw.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED; + hw.periods_min = 2; + hw.periods_max = UINT_MAX; + hw.period_bytes_min = 256; + hw.period_bytes_max = dma_get_max_seg_size(dma_dev); + hw.buffer_bytes_max = SIZE_MAX; + hw.fifo_size = dma_data->fifo_size; + + ret = dma_get_slave_caps(chan, &dma_caps); + if (ret == 0) { + if (dma_caps.cmd_pause) + hw.info |= SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME; + } + + return snd_soc_set_runtime_hwparams(substream, &hw); } -static struct device *dmaengine_dma_dev(struct dmaengine_pcm *pcm, - struct snd_pcm_substream *substream) +static int dmaengine_pcm_open(struct snd_pcm_substream *substream) { - if (!pcm->chan[substream->stream]) - return NULL; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct dma_chan *chan = pcm->chan[substream->stream]; + int ret; - return pcm->chan[substream->stream]->device->dev; + ret = dmaengine_pcm_set_runtime_hwparams(substream); + if (ret) + return ret; + + return snd_dmaengine_pcm_open(substream, chan); } static void dmaengine_pcm_free(struct snd_pcm *pcm) @@ -126,6 +170,9 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( struct snd_pcm_substream *substream) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); + struct snd_dmaengine_dai_dma_data *dma_data; + + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; @@ -134,22 +181,42 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( return pcm->config->compat_request_channel(rtd, substream); return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream)); + dma_data->filter_data); } static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); const struct snd_dmaengine_pcm_config *config = pcm->config; + struct device *dev = rtd->platform->dev; + struct snd_dmaengine_dai_dma_data *dma_data; struct snd_pcm_substream *substream; + size_t prealloc_buffer_size; + size_t max_buffer_size; unsigned int i; int ret; + if (config && config->prealloc_buffer_size) { + prealloc_buffer_size = config->prealloc_buffer_size; + max_buffer_size = config->pcm_hardware->buffer_bytes_max; + } else { + prealloc_buffer_size = 512 * 1024; + max_buffer_size = SIZE_MAX; + } + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { substream = rtd->pcm->streams[i].substream; if (!substream) continue; + dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); + + if (!pcm->chan[i] && + (pcm->flags & SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) + pcm->chan[i] = dma_request_slave_channel(dev, + dma_data->chan_name); + if (!pcm->chan[i] && (pcm->flags & SND_DMAENGINE_PCM_FLAG_COMPAT)) { pcm->chan[i] = dmaengine_pcm_compat_request_channel(rtd, substream); @@ -165,8 +232,8 @@ static int dmaengine_pcm_new(struct snd_soc_pcm_runtime *rtd) ret = snd_pcm_lib_preallocate_pages(substream, SNDRV_DMA_TYPE_DEV, dmaengine_dma_dev(pcm, substream), - config->prealloc_buffer_size, - config->pcm_hardware->buffer_bytes_max); + prealloc_buffer_size, + max_buffer_size); if (ret) goto err_free; } @@ -222,7 +289,9 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, { unsigned int i; - if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_DT) || !dev->of_node) + if ((pcm->flags & (SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_CUSTOM_CHANNEL_NAME)) || + !dev->of_node) return; if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) { diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index f056f632557c..7b2d23ba69b3 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -56,7 +56,6 @@ static const struct snd_pcm_hardware tegra_pcm_hardware = { static const struct snd_dmaengine_pcm_config tegra_dmaengine_pcm_config = { .pcm_hardware = &tegra_pcm_hardware, .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, - .compat_filter_fn = NULL, .prealloc_buffer_size = PAGE_SIZE * 8, }; diff --git a/sound/usb/usx2y/us122l.c b/sound/usb/usx2y/us122l.c index d0323a693ba2..999550bbad40 100644 --- a/sound/usb/usx2y/us122l.c +++ b/sound/usb/usx2y/us122l.c @@ -262,7 +262,9 @@ static int usb_stream_hwdep_mmap(struct snd_hwdep *hw, } area->vm_ops = &usb_stream_hwdep_vm_ops; - area->vm_flags |= VM_DONTEXPAND | VM_DONTDUMP; + area->vm_flags |= VM_DONTDUMP; + if (!read) + area->vm_flags |= VM_DONTEXPAND; area->vm_private_data = us122l; atomic_inc(&us122l->mmap_count); out: diff --git a/sound/usb/usx2y/usbusx2yaudio.c b/sound/usb/usx2y/usbusx2yaudio.c index 63fb5219f0f8..6234a51625b1 100644 --- a/sound/usb/usx2y/usbusx2yaudio.c +++ b/sound/usb/usx2y/usbusx2yaudio.c @@ -299,19 +299,6 @@ static void usX2Y_error_urb_status(struct usX2Ydev *usX2Y, usX2Y_clients_stop(usX2Y); } -static void usX2Y_error_sequence(struct usX2Ydev *usX2Y, - struct snd_usX2Y_substream *subs, struct urb *urb) -{ - snd_printk(KERN_ERR -"Sequence Error!(hcd_frame=%i ep=%i%s;wait=%i,frame=%i).\n" -"Most probably some urb of usb-frame %i is still missing.\n" -"Cause could be too long delays in usb-hcd interrupt handling.\n", - usb_get_current_frame_number(usX2Y->dev), - subs->endpoint, usb_pipein(urb->pipe) ? "in" : "out", - usX2Y->wait_iso_frame, urb->start_frame, usX2Y->wait_iso_frame); - usX2Y_clients_stop(usX2Y); -} - static void i_usX2Y_urb_complete(struct urb *urb) { struct snd_usX2Y_substream *subs = urb->context; @@ -328,12 +315,9 @@ static void i_usX2Y_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + + subs->completed_urb = urb; + { struct snd_usX2Y_substream *capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE], *playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; diff --git a/sound/usb/usx2y/usx2yhwdeppcm.c b/sound/usb/usx2y/usx2yhwdeppcm.c index f2a1acdc4d83..814d0e887c62 100644 --- a/sound/usb/usx2y/usx2yhwdeppcm.c +++ b/sound/usb/usx2y/usx2yhwdeppcm.c @@ -244,13 +244,8 @@ static void i_usX2Y_usbpcm_urb_complete(struct urb *urb) usX2Y_error_urb_status(usX2Y, subs, urb); return; } - if (likely((urb->start_frame & 0xFFFF) == (usX2Y->wait_iso_frame & 0xFFFF))) - subs->completed_urb = urb; - else { - usX2Y_error_sequence(usX2Y, subs, urb); - return; - } + subs->completed_urb = urb; capsubs = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE]; capsubs2 = usX2Y->subs[SNDRV_PCM_STREAM_CAPTURE + 2]; playbacksubs = usX2Y->subs[SNDRV_PCM_STREAM_PLAYBACK]; |