diff options
Diffstat (limited to 'sound')
35 files changed, 471 insertions, 176 deletions
diff --git a/sound/atmel/abdac.c b/sound/atmel/abdac.c index 872d59e35ee2..721d8fd45685 100644 --- a/sound/atmel/abdac.c +++ b/sound/atmel/abdac.c @@ -357,7 +357,8 @@ static int set_sample_rates(struct atmel_abdac *dac) if (new_rate < 0) break; /* make sure we are below the ABDAC clock */ - if (new_rate <= clk_get_rate(dac->pclk)) { + if (index < MAX_NUM_RATES && + new_rate <= clk_get_rate(dac->pclk)) { dac->rates[index] = new_rate / 256; index++; } diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 6e03b465e44e..a2104671f51d 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1937,6 +1937,8 @@ static int wait_for_avail(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DISCONNECTED: err = -EBADFD; goto _endloop; + case SNDRV_PCM_STATE_PAUSED: + continue; } if (!tout) { snd_printd("%s write error (DMA or IRQ trouble?)\n", diff --git a/sound/firewire/amdtp.c b/sound/firewire/amdtp.c index d3226892ad6b..9048777228e2 100644 --- a/sound/firewire/amdtp.c +++ b/sound/firewire/amdtp.c @@ -434,17 +434,14 @@ static void queue_out_packet(struct amdtp_out_stream *s, unsigned int cycle) return; index = s->packet_index; + /* this module generate empty packet for 'no data' */ syt = calculate_syt(s, cycle); - if (!(s->flags & CIP_BLOCKING)) { + if (!(s->flags & CIP_BLOCKING)) data_blocks = calculate_data_blocks(s); - } else { - if (syt != 0xffff) { - data_blocks = s->syt_interval; - } else { - data_blocks = 0; - syt = 0xffffff; - } - } + else if (syt != 0xffff) + data_blocks = s->syt_interval; + else + data_blocks = 0; buffer = s->buffer.packets[index].buffer; buffer[0] = cpu_to_be32(ACCESS_ONCE(s->source_node_id_field) | diff --git a/sound/firewire/dice.c b/sound/firewire/dice.c index 57bcd31fcc12..c0aa64941cee 100644 --- a/sound/firewire/dice.c +++ b/sound/firewire/dice.c @@ -1019,7 +1019,7 @@ static void dice_proc_read(struct snd_info_entry *entry, if (dice_proc_read_mem(dice, &tx_rx_header, sections[2], 2) < 0) return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx)); + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.tx) / 4); for (stream = 0; stream < tx_rx_header.number; ++stream) { if (dice_proc_read_mem(dice, &buf.tx, sections[2] + 2 + stream * tx_rx_header.size, @@ -1045,7 +1045,7 @@ static void dice_proc_read(struct snd_info_entry *entry, if (dice_proc_read_mem(dice, &tx_rx_header, sections[4], 2) < 0) return; - quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx)); + quadlets = min_t(u32, tx_rx_header.size, sizeof(buf.rx) / 4); for (stream = 0; stream < tx_rx_header.number; ++stream) { if (dice_proc_read_mem(dice, &buf.rx, sections[4] + 2 + stream * tx_rx_header.size, diff --git a/sound/pci/hda/hda_codec.h b/sound/pci/hda/hda_codec.h index 77db69480c19..7aa9870040c1 100644 --- a/sound/pci/hda/hda_codec.h +++ b/sound/pci/hda/hda_codec.h @@ -698,7 +698,6 @@ struct hda_bus { unsigned int in_reset:1; /* during reset operation */ unsigned int power_keep_link_on:1; /* don't power off HDA link */ unsigned int no_response_fallback:1; /* don't fallback at RIRB error */ - unsigned int avoid_link_reset:1; /* don't reset link at runtime PM */ int primary_dig_out_type; /* primary digital out PCM type */ }; diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 3067ed4fe3b2..c7f6d1cab606 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -474,6 +474,20 @@ static void invalidate_nid_path(struct hda_codec *codec, int idx) memset(path, 0, sizeof(*path)); } +/* return a DAC if paired to the given pin by codec driver */ +static hda_nid_t get_preferred_dac(struct hda_codec *codec, hda_nid_t pin) +{ + struct hda_gen_spec *spec = codec->spec; + const hda_nid_t *list = spec->preferred_dacs; + + if (!list) + return 0; + for (; *list; list += 2) + if (*list == pin) + return list[1]; + return 0; +} + /* look for an empty DAC slot */ static hda_nid_t look_for_dac(struct hda_codec *codec, hda_nid_t pin, bool is_digital) @@ -1192,7 +1206,14 @@ static int try_assign_dacs(struct hda_codec *codec, int num_outs, continue; } - dacs[i] = look_for_dac(codec, pin, false); + dacs[i] = get_preferred_dac(codec, pin); + if (dacs[i]) { + if (is_dac_already_used(codec, dacs[i])) + badness += bad->shared_primary; + } + + if (!dacs[i]) + dacs[i] = look_for_dac(codec, pin, false); if (!dacs[i] && !i) { /* try to steal the DAC of surrounds for the front */ for (j = 1; j < num_outs; j++) { @@ -2506,12 +2527,8 @@ static int create_out_jack_modes(struct hda_codec *codec, int num_pins, for (i = 0; i < num_pins; i++) { hda_nid_t pin = pins[i]; - if (pin == spec->hp_mic_pin) { - int ret = create_hp_mic_jack_mode(codec, pin); - if (ret < 0) - return ret; + if (pin == spec->hp_mic_pin) continue; - } if (get_out_jack_num_items(codec, pin) > 1) { struct snd_kcontrol_new *knew; char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; @@ -2764,7 +2781,7 @@ static int hp_mic_jack_mode_put(struct snd_kcontrol *kcontrol, val &= ~(AC_PINCTL_VREFEN | PIN_HP); val |= get_vref_idx(vref_caps, idx) | PIN_IN; } else - val = snd_hda_get_default_vref(codec, nid); + val = snd_hda_get_default_vref(codec, nid) | PIN_IN; } snd_hda_set_pin_ctl_cache(codec, nid, val); call_hp_automute(codec, NULL); @@ -2784,9 +2801,6 @@ static int create_hp_mic_jack_mode(struct hda_codec *codec, hda_nid_t pin) struct hda_gen_spec *spec = codec->spec; struct snd_kcontrol_new *knew; - if (get_out_jack_num_items(codec, pin) <= 1 && - get_in_jack_num_items(codec, pin) <= 1) - return 0; /* no need */ knew = snd_hda_gen_add_kctl(spec, "Headphone Mic Jack Mode", &hp_mic_jack_mode_enum); if (!knew) @@ -2815,6 +2829,42 @@ static int add_loopback_list(struct hda_gen_spec *spec, hda_nid_t mix, int idx) return 0; } +/* return true if either a volume or a mute amp is found for the given + * aamix path; the amp has to be either in the mixer node or its direct leaf + */ +static bool look_for_mix_leaf_ctls(struct hda_codec *codec, hda_nid_t mix_nid, + hda_nid_t pin, unsigned int *mix_val, + unsigned int *mute_val) +{ + int idx, num_conns; + const hda_nid_t *list; + hda_nid_t nid; + + idx = snd_hda_get_conn_index(codec, mix_nid, pin, true); + if (idx < 0) + return false; + + *mix_val = *mute_val = 0; + if (nid_has_volume(codec, mix_nid, HDA_INPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (nid_has_mute(codec, mix_nid, HDA_INPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); + if (*mix_val && *mute_val) + return true; + + /* check leaf node */ + num_conns = snd_hda_get_conn_list(codec, mix_nid, &list); + if (num_conns < idx) + return false; + nid = list[idx]; + if (!*mix_val && nid_has_volume(codec, nid, HDA_OUTPUT)) + *mix_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + if (!*mute_val && nid_has_mute(codec, nid, HDA_OUTPUT)) + *mute_val = HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT); + + return *mix_val || *mute_val; +} + /* create input playback/capture controls for the given pin */ static int new_analog_input(struct hda_codec *codec, int input_idx, hda_nid_t pin, const char *ctlname, int ctlidx, @@ -2822,12 +2872,11 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, { struct hda_gen_spec *spec = codec->spec; struct nid_path *path; - unsigned int val; + unsigned int mix_val, mute_val; int err, idx; - if (!nid_has_volume(codec, mix_nid, HDA_INPUT) && - !nid_has_mute(codec, mix_nid, HDA_INPUT)) - return 0; /* no need for analog loopback */ + if (!look_for_mix_leaf_ctls(codec, mix_nid, pin, &mix_val, &mute_val)) + return 0; path = snd_hda_add_new_path(codec, pin, mix_nid, 0); if (!path) @@ -2836,20 +2885,18 @@ static int new_analog_input(struct hda_codec *codec, int input_idx, spec->loopback_paths[input_idx] = snd_hda_get_path_idx(codec, path); idx = path->idx[path->depth - 1]; - if (nid_has_volume(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, val); + if (mix_val) { + err = __add_pb_vol_ctrl(spec, HDA_CTL_WIDGET_VOL, ctlname, ctlidx, mix_val); if (err < 0) return err; - path->ctls[NID_PATH_VOL_CTL] = val; + path->ctls[NID_PATH_VOL_CTL] = mix_val; } - if (nid_has_mute(codec, mix_nid, HDA_INPUT)) { - val = HDA_COMPOSE_AMP_VAL(mix_nid, 3, idx, HDA_INPUT); - err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, val); + if (mute_val) { + err = __add_pb_sw_ctrl(spec, HDA_CTL_WIDGET_MUTE, ctlname, ctlidx, mute_val); if (err < 0) return err; - path->ctls[NID_PATH_MUTE_CTL] = val; + path->ctls[NID_PATH_MUTE_CTL] = mute_val; } path->active = true; @@ -4271,6 +4318,26 @@ static unsigned int snd_hda_gen_path_power_filter(struct hda_codec *codec, return AC_PWRST_D3; } +/* mute all aamix inputs initially; parse up to the first leaves */ +static void mute_all_mixer_nid(struct hda_codec *codec, hda_nid_t mix) +{ + int i, nums; + const hda_nid_t *conn; + bool has_amp; + + nums = snd_hda_get_conn_list(codec, mix, &conn); + has_amp = nid_has_mute(codec, mix, HDA_INPUT); + for (i = 0; i < nums; i++) { + if (has_amp) + snd_hda_codec_amp_stereo(codec, mix, + HDA_INPUT, i, + 0xff, HDA_AMP_MUTE); + else if (nid_has_volume(codec, conn[i], HDA_OUTPUT)) + snd_hda_codec_amp_stereo(codec, conn[i], + HDA_OUTPUT, 0, + 0xff, HDA_AMP_MUTE); + } +} /* * Parse the given BIOS configuration and set up the hda_gen_spec @@ -4383,6 +4450,17 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, if (err < 0) return err; + /* create "Headphone Mic Jack Mode" if no input selection is + * available (or user specifies add_jack_modes hint) + */ + if (spec->hp_mic_pin && + (spec->auto_mic || spec->input_mux.num_items == 1 || + spec->add_jack_modes)) { + err = create_hp_mic_jack_mode(codec, spec->hp_mic_pin); + if (err < 0) + return err; + } + if (spec->add_jack_modes) { if (cfg->line_out_type != AUTO_PIN_SPEAKER_OUT) { err = create_out_jack_modes(codec, cfg->line_outs, @@ -4398,6 +4476,10 @@ int snd_hda_gen_parse_auto_config(struct hda_codec *codec, } } + /* mute all aamix input initially */ + if (spec->mixer_nid) + mute_all_mixer_nid(codec, spec->mixer_nid); + dig_only: parse_digital(codec); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 7e45cb44d151..0929a06df812 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -249,6 +249,9 @@ struct hda_gen_spec { const struct badness_table *main_out_badness; const struct badness_table *extra_out_badness; + /* preferred pin/DAC pairs; an array of paired NIDs */ + const hda_nid_t *preferred_dacs; + /* loopback mixing mode */ bool aamix_mode; diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 7a09404579a7..956871d8b3d2 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2994,8 +2994,7 @@ static int azx_runtime_suspend(struct device *dev) STATESTS_INT_MASK); azx_stop_chip(chip); - if (!chip->bus->avoid_link_reset) - azx_enter_link_reset(chip); + azx_enter_link_reset(chip); azx_clear_irq_pending(chip); if (chip->driver_caps & AZX_DCAPS_I915_POWERWELL) hda_display_power(false); @@ -3434,6 +3433,10 @@ static void check_probe_mask(struct azx *chip, int dev) * white/black-list for enable_msi */ static struct snd_pci_quirk msi_black_list[] = { + SND_PCI_QUIRK(0x103c, 0x2191, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x2192, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21f7, "HP", 0), /* AMD Hudson */ + SND_PCI_QUIRK(0x103c, 0x21fa, "HP", 0), /* AMD Hudson */ SND_PCI_QUIRK(0x1043, 0x81f2, "ASUS", 0), /* Athlon64 X2 + nvidia */ SND_PCI_QUIRK(0x1043, 0x81f6, "ASUS", 0), /* nvidia */ SND_PCI_QUIRK(0x1043, 0x822d, "ASUS", 0), /* Athlon64 X2 + nvidia MCP55 */ @@ -3877,7 +3880,8 @@ static int azx_probe(struct pci_dev *pci, } dev++; - complete_all(&chip->probe_wait); + if (chip->disabled) + complete_all(&chip->probe_wait); return 0; out_free: @@ -3954,10 +3958,10 @@ static int azx_probe_continue(struct azx *chip) if ((chip->driver_caps & AZX_DCAPS_PM_RUNTIME) || chip->use_vga_switcheroo) pm_runtime_put_noidle(&pci->dev); - return 0; - out_free: - chip->init_failed = 1; + if (err < 0) + chip->init_failed = 1; + complete_all(&chip->probe_wait); return err; } diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 1a83559f4cbd..699262a3e07a 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -147,6 +147,8 @@ static void ad_vmaster_eapd_hook(void *private_data, int enabled) if (!spec->eapd_nid) return; + if (codec->inv_eapd) + enabled = !enabled; snd_hda_codec_update_cache(codec, spec->eapd_nid, 0, AC_VERB_SET_EAPD_BTLENABLE, enabled ? 0x02 : 0x00); @@ -338,6 +340,14 @@ static int patch_ad1986a(struct hda_codec *codec) { int err; struct ad198x_spec *spec; + static hda_nid_t preferred_pairs[] = { + 0x1a, 0x03, + 0x1b, 0x03, + 0x1c, 0x04, + 0x1d, 0x05, + 0x1e, 0x03, + 0 + }; err = alloc_ad_spec(codec); if (err < 0) @@ -358,6 +368,11 @@ static int patch_ad1986a(struct hda_codec *codec) * So, let's disable the shared stream. */ spec->gen.multiout.no_share_stream = 1; + /* give fixed DAC/pin pairs */ + spec->gen.preferred_dacs = preferred_pairs; + + /* AD1986A can't manage the dynamic pin on/off smoothly */ + spec->gen.auto_mute_via_amp = 1; snd_hda_pick_fixup(codec, ad1986a_fixup_models, ad1986a_fixup_tbl, ad1986a_fixups); @@ -962,6 +977,7 @@ static void ad1884_fixup_hp_eapd(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: spec->gen.vmaster_mute.hook = ad1884_vmaster_hp_gpio_hook; + spec->gen.own_eapd_ctl = 1; snd_hda_sequence_write_cache(codec, gpio_init_verbs); break; case HDA_FIXUP_ACT_PROBE: diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index c205bb1747fd..3fbf2883e06e 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -2936,7 +2936,6 @@ static const struct snd_pci_quirk cxt5066_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0401, "Dell Vostro 1014", CXT5066_DELL_VOSTRO), SND_PCI_QUIRK(0x1028, 0x0408, "Dell Inspiron One 19T", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x1028, 0x050f, "Dell Inspiron", CXT5066_IDEAPAD), - SND_PCI_QUIRK(0x1028, 0x0510, "Dell Vostro", CXT5066_IDEAPAD), SND_PCI_QUIRK(0x103c, 0x360b, "HP G60", CXT5066_HP_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13f3, "Asus A52J", CXT5066_ASUS), SND_PCI_QUIRK(0x1043, 0x1643, "Asus K52JU", CXT5066_ASUS), @@ -3244,9 +3243,29 @@ enum { #if IS_ENABLED(CONFIG_THINKPAD_ACPI) #include <linux/thinkpad_acpi.h> +#include <acpi/acpi.h> static int (*led_set_func)(int, bool); +static acpi_status acpi_check_cb(acpi_handle handle, u32 lvl, void *context, + void **rv) +{ + bool *found = context; + *found = true; + return AE_OK; +} + +static bool is_thinkpad(struct hda_codec *codec) +{ + bool found = false; + if (codec->subsystem_id >> 16 != 0x17aa) + return false; + if (ACPI_SUCCESS(acpi_get_devices("LEN0068", acpi_check_cb, &found, NULL)) && found) + return true; + found = false; + return ACPI_SUCCESS(acpi_get_devices("IBM0068", acpi_check_cb, &found, NULL)) && found; +} + static void update_tpacpi_mute_led(void *private_data, int enabled) { struct hda_codec *codec = private_data; @@ -3279,6 +3298,8 @@ static void cxt_fixup_thinkpad_acpi(struct hda_codec *codec, bool removefunc = false; if (action == HDA_FIXUP_ACT_PROBE) { + if (!is_thinkpad(codec)) + return; if (!led_set_func) led_set_func = symbol_request(tpacpi_led_set); if (!led_set_func) { @@ -3494,6 +3515,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x17aa, 0x3975, "Lenovo U300s", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x3977, "Lenovo IdeaPad U310", CXT_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x17aa, 0x397b, "Lenovo S205", CXT_FIXUP_STEREO_DMIC), + SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", CXT_FIXUP_THINKPAD_ACPI), SND_PCI_QUIRK(0x1c06, 0x2011, "Lemote A1004", CXT_PINCFG_LEMOTE_A1004), SND_PCI_QUIRK(0x1c06, 0x2012, "Lemote A1205", CXT_PINCFG_LEMOTE_A1205), {} diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 08407bed093e..f281c8068557 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1142,32 +1142,34 @@ static void hdmi_setup_audio_infoframe(struct hda_codec *codec, static bool hdmi_present_sense(struct hdmi_spec_per_pin *per_pin, int repoll); -static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +static void jack_callback(struct hda_codec *codec, struct hda_jack_tbl *jack) { struct hdmi_spec *spec = codec->spec; + int pin_idx = pin_nid_to_pin_index(spec, jack->nid); + if (pin_idx < 0) + return; + + if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) + snd_hda_jack_report_sync(codec); +} + +static void hdmi_intrinsic_event(struct hda_codec *codec, unsigned int res) +{ int tag = res >> AC_UNSOL_RES_TAG_SHIFT; - int pin_nid; - int pin_idx; struct hda_jack_tbl *jack; int dev_entry = (res & AC_UNSOL_RES_DE) >> AC_UNSOL_RES_DE_SHIFT; jack = snd_hda_jack_tbl_get_from_tag(codec, tag); if (!jack) return; - pin_nid = jack->nid; jack->jack_dirty = 1; _snd_printd(SND_PR_VERBOSE, "HDMI hot plug event: Codec=%d Pin=%d Device=%d Inactive=%d Presence_Detect=%d ELD_Valid=%d\n", - codec->addr, pin_nid, dev_entry, !!(res & AC_UNSOL_RES_IA), + codec->addr, jack->nid, dev_entry, !!(res & AC_UNSOL_RES_IA), !!(res & AC_UNSOL_RES_PD), !!(res & AC_UNSOL_RES_ELDV)); - pin_idx = pin_nid_to_pin_index(spec, pin_nid); - if (pin_idx < 0) - return; - - if (hdmi_present_sense(get_pin(spec, pin_idx), 1)) - snd_hda_jack_report_sync(codec); + jack_callback(codec, jack); } static void hdmi_non_intrinsic_event(struct hda_codec *codec, unsigned int res) @@ -2095,7 +2097,8 @@ static int generic_hdmi_init(struct hda_codec *codec) hda_nid_t pin_nid = per_pin->pin_nid; hdmi_init_pin(codec, pin_nid); - snd_hda_jack_detect_enable(codec, pin_nid, pin_nid); + snd_hda_jack_detect_enable_callback(codec, pin_nid, pin_nid, + codec->jackpoll_interval > 0 ? jack_callback : NULL); } return 0; } @@ -2334,8 +2337,9 @@ static int simple_playback_build_controls(struct hda_codec *codec) int err; per_cvt = get_cvt(spec, 0); - err = snd_hda_create_spdif_out_ctls(codec, per_cvt->cvt_nid, - per_cvt->cvt_nid); + err = snd_hda_create_dig_out_ctls(codec, per_cvt->cvt_nid, + per_cvt->cvt_nid, + HDA_PCM_TYPE_HDMI); if (err < 0) return err; return simple_hdmi_build_jack(codec, 0); diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5e42059f10a1..c5646941539a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1780,8 +1780,11 @@ enum { ALC889_FIXUP_DAC_ROUTE, ALC889_FIXUP_MBP_VREF, ALC889_FIXUP_IMAC91_VREF, + ALC889_FIXUP_MBA21_VREF, ALC882_FIXUP_INV_DMIC, ALC882_FIXUP_NO_PRIMARY_HP, + ALC887_FIXUP_ASUS_BASS, + ALC887_FIXUP_BASS_CHMAP, }; static void alc889_fixup_coef(struct hda_codec *codec, @@ -1882,17 +1885,13 @@ static void alc889_fixup_mbp_vref(struct hda_codec *codec, } } -/* Set VREF on speaker pins on imac91 */ -static void alc889_fixup_imac91_vref(struct hda_codec *codec, - const struct hda_fixup *fix, int action) +static void alc889_fixup_mac_pins(struct hda_codec *codec, + const hda_nid_t *nids, int num_nids) { struct alc_spec *spec = codec->spec; - static hda_nid_t nids[2] = { 0x18, 0x1a }; int i; - if (action != HDA_FIXUP_ACT_INIT) - return; - for (i = 0; i < ARRAY_SIZE(nids); i++) { + for (i = 0; i < num_nids; i++) { unsigned int val; val = snd_hda_codec_get_pin_target(codec, nids[i]); val |= AC_PINCTL_VREF_50; @@ -1901,6 +1900,26 @@ static void alc889_fixup_imac91_vref(struct hda_codec *codec, spec->gen.keep_vref_in_automute = 1; } +/* Set VREF on speaker pins on imac91 */ +static void alc889_fixup_imac91_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[2] = { 0x18, 0x1a }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + +/* Set VREF on speaker pins on mba21 */ +static void alc889_fixup_mba21_vref(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static hda_nid_t nids[2] = { 0x18, 0x19 }; + + if (action == HDA_FIXUP_ACT_INIT) + alc889_fixup_mac_pins(codec, nids, ARRAY_SIZE(nids)); +} + /* Don't take HP output as primary * Strangely, the speaker output doesn't work on Vaio Z and some Vaio * all-in-one desktop PCs (for example VGC-LN51JGB) through DAC 0x05 @@ -1915,6 +1934,9 @@ static void alc882_fixup_no_primary_hp(struct hda_codec *codec, } } +static void alc_fixup_bass_chmap(struct hda_codec *codec, + const struct hda_fixup *fix, int action); + static const struct hda_fixup alc882_fixups[] = { [ALC882_FIXUP_ABIT_AW9D_MAX] = { .type = HDA_FIXUP_PINS, @@ -2097,6 +2119,12 @@ static const struct hda_fixup alc882_fixups[] = { .chained = true, .chain_id = ALC882_FIXUP_GPIO1, }, + [ALC889_FIXUP_MBA21_VREF] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc889_fixup_mba21_vref, + .chained = true, + .chain_id = ALC889_FIXUP_MBP_VREF, + }, [ALC882_FIXUP_INV_DMIC] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_inv_dmic_0x12, @@ -2105,6 +2133,19 @@ static const struct hda_fixup alc882_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc882_fixup_no_primary_hp, }, + [ALC887_FIXUP_ASUS_BASS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + {0x16, 0x99130130}, /* bass speaker */ + {} + }, + .chained = true, + .chain_id = ALC887_FIXUP_BASS_CHMAP, + }, + [ALC887_FIXUP_BASS_CHMAP] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_bass_chmap, + }, }; static const struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -2138,6 +2179,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1873, "ASUS W90V", ALC882_FIXUP_ASUS_W90V), SND_PCI_QUIRK(0x1043, 0x1971, "Asus W2JC", ALC882_FIXUP_ASUS_W2JC), SND_PCI_QUIRK(0x1043, 0x835f, "Asus Eee 1601", ALC888_FIXUP_EEE1601), + SND_PCI_QUIRK(0x1043, 0x84bc, "ASUS ET2700", ALC887_FIXUP_ASUS_BASS), SND_PCI_QUIRK(0x104d, 0x9047, "Sony Vaio TT", ALC889_FIXUP_VAIO_TT), SND_PCI_QUIRK(0x104d, 0x905a, "Sony Vaio Z", ALC882_FIXUP_NO_PRIMARY_HP), SND_PCI_QUIRK(0x104d, 0x9043, "Sony Vaio VGC-LN51JGB", ALC882_FIXUP_NO_PRIMARY_HP), @@ -2153,7 +2195,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x106b, 0x3000, "iMac", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3200, "iMac 7,1 Aluminum", ALC882_FIXUP_EAPD), SND_PCI_QUIRK(0x106b, 0x3400, "MacBookAir 1,1", ALC889_FIXUP_MBP_VREF), - SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBP_VREF), + SND_PCI_QUIRK(0x106b, 0x3500, "MacBookAir 2,1", ALC889_FIXUP_MBA21_VREF), SND_PCI_QUIRK(0x106b, 0x3600, "Macbook 3,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3800, "MacbookPro 4,1", ALC889_FIXUP_MBP_VREF), SND_PCI_QUIRK(0x106b, 0x3e00, "iMac 24 Aluminum", ALC885_FIXUP_MACPRO_GPIO), @@ -3268,6 +3310,7 @@ static void alc_headset_mode_ctia(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d60); alc_write_coef_idx(codec, 0xc3, 0x0000); break; @@ -3296,6 +3339,7 @@ static void alc_headset_mode_omtp(struct hda_codec *codec) alc_write_coef_idx(codec, 0x18, 0x7388); break; case 0x10ec0668: + alc_write_coef_idx(codec, 0x11, 0x0001); alc_write_coef_idx(codec, 0x15, 0x0d50); alc_write_coef_idx(codec, 0xc3, 0x0000); break; @@ -3581,11 +3625,6 @@ static void alc283_hp_automute_hook(struct hda_codec *codec, vref); } -static void alc283_chromebook_caps(struct hda_codec *codec) -{ - snd_hda_override_wcaps(codec, 0x03, 0); -} - static void alc283_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { @@ -3594,9 +3633,26 @@ static void alc283_fixup_chromebook(struct hda_codec *codec, switch (action) { case HDA_FIXUP_ACT_PRE_PROBE: - alc283_chromebook_caps(codec); + snd_hda_override_wcaps(codec, 0x03, 0); /* Disable AA-loopback as it causes white noise */ spec->gen.mixer_nid = 0; + break; + case HDA_FIXUP_ACT_INIT: + /* Enable Line1 input control by verb */ + val = alc_read_coef_idx(codec, 0x1a); + alc_write_coef_idx(codec, 0x1a, val | (1 << 4)); + break; + } +} + +static void alc283_fixup_sense_combo_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + int val; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: spec->gen.hp_automute_hook = alc283_hp_automute_hook; break; case HDA_FIXUP_ACT_INIT: @@ -3604,9 +3660,6 @@ 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; } } @@ -3796,11 +3849,14 @@ enum { ALC269_FIXUP_ASUS_X101, ALC271_FIXUP_AMIC_MIC2, ALC271_FIXUP_HP_GATE_MIC_JACK, + ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572, ALC269_FIXUP_ACER_AC700, ALC269_FIXUP_LIMIT_INT_MIC_BOOST, + ALC269VB_FIXUP_ASUS_ZENBOOK, ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED, ALC269VB_FIXUP_ORDISSIMO_EVE2, ALC283_FIXUP_CHROME_BOOK, + ALC283_FIXUP_SENSE_COMBO_JACK, ALC282_FIXUP_ASUS_TX300, ALC283_FIXUP_INT_MIC, ALC290_FIXUP_MONO_SPEAKERS, @@ -4056,6 +4112,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC271_FIXUP_AMIC_MIC2, }, + [ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC271_FIXUP_HP_GATE_MIC_JACK, + }, [ALC269_FIXUP_ACER_AC700] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -4075,6 +4137,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, + [ALC269VB_FIXUP_ASUS_ZENBOOK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc269_fixup_limit_int_mic_boost, + .chained = true, + .chain_id = ALC269VB_FIXUP_DMIC, + }, [ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED] = { .type = HDA_FIXUP_FUNC, .v.func = alc269_fixup_limit_int_mic_boost, @@ -4094,6 +4162,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc283_fixup_chromebook, }, + [ALC283_FIXUP_SENSE_COMBO_JACK] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc283_fixup_sense_combo_jack, + .chained = true, + .chain_id = ALC283_FIXUP_CHROME_BOOK, + }, [ALC282_FIXUP_ASUS_TX300] = { .type = HDA_FIXUP_FUNC, .v.func = alc282_fixup_asus_tx300, @@ -4141,6 +4215,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0740, "Acer AO725", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK(0x1025, 0x0742, "Acer AO756", ALC271_FIXUP_HP_GATE_MIC_JACK), SND_PCI_QUIRK_VENDOR(0x1025, "Acer Aspire", ALC271_FIXUP_DMIC), + SND_PCI_QUIRK(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), SND_PCI_QUIRK(0x1028, 0x0470, "Dell M101z", ALC269_FIXUP_DELL_M101Z), SND_PCI_QUIRK(0x1028, 0x05bd, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x05be, "Dell", ALC269_FIXUP_DELL2_MIC_NO_PRESENCE), @@ -4172,11 +4247,16 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0606, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), 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, 0x0610, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0613, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0614, "Dell Inspiron 3135", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0616, "Dell Vostro 5470", ALC290_FIXUP_MONO_SPEAKERS), SND_PCI_QUIRK(0x1028, 0x061f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0629, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0638, "Dell Inspiron 5439", ALC290_FIXUP_MONO_SPEAKERS), + SND_PCI_QUIRK(0x1028, 0x063e, "Dell", ALC269_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x063f, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0640, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), 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), @@ -4184,13 +4264,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x1973, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x1983, "HP Pavilion", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x218b, "HP", ALC269_FIXUP_LIMIT_INT_MIC_BOOST_MUTE_LED), - SND_PCI_QUIRK(0x103c, 0x21ed, "HP Falco Chromebook", ALC283_FIXUP_CHROME_BOOK), SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x1043, 0x103f, "ASUS TX300", ALC282_FIXUP_ASUS_TX300), SND_PCI_QUIRK(0x1043, 0x106d, "Asus K53BE", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x1043, 0x115d, "Asus 1015E", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), - 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, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), + SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), SND_PCI_QUIRK(0x1043, 0x1a13, "Asus G73Jw", ALC269_FIXUP_ASUS_G73JW), SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), @@ -4292,6 +4371,8 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC269_FIXUP_HP_GPIO_LED, .name = "hp-gpio-led"}, {.id = ALC269_FIXUP_DELL1_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, {.id = ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, .name = "dell-headset-dock"}, + {.id = ALC283_FIXUP_CHROME_BOOK, .name = "alc283-chrome"}, + {.id = ALC283_FIXUP_SENSE_COMBO_JACK, .name = "alc283-sense-combo"}, {} }; @@ -4467,6 +4548,7 @@ enum { ALC861_FIXUP_AMP_VREF_0F, ALC861_FIXUP_NO_JACK_DETECT, ALC861_FIXUP_ASUS_A6RP, + ALC660_FIXUP_ASUS_W7J, }; /* On some laptops, VREF of pin 0x0f is abused for controlling the main amp */ @@ -4516,10 +4598,22 @@ static const struct hda_fixup alc861_fixups[] = { .v.func = alc861_fixup_asus_amp_vref_0f, .chained = true, .chain_id = ALC861_FIXUP_NO_JACK_DETECT, + }, + [ALC660_FIXUP_ASUS_W7J] = { + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* ASUS W7J needs a magic pin setup on unused NID 0x10 + * for enabling outputs + */ + {0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, 0x24}, + { } + }, } }; static const struct snd_pci_quirk alc861_fixup_tbl[] = { + SND_PCI_QUIRK(0x1043, 0x1253, "ASUS W7J", ALC660_FIXUP_ASUS_W7J), + SND_PCI_QUIRK(0x1043, 0x1263, "ASUS Z35HL", ALC660_FIXUP_ASUS_W7J), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS A6Rp", ALC861_FIXUP_ASUS_A6RP), SND_PCI_QUIRK_VENDOR(0x1043, "ASUS laptop", ALC861_FIXUP_AMP_VREF_0F), SND_PCI_QUIRK(0x1462, 0x7254, "HP DX2200", ALC861_FIXUP_NO_JACK_DETECT), @@ -4715,7 +4809,7 @@ static const struct snd_pcm_chmap_elem asus_pcm_2_1_chmaps[] = { }; /* override the 2.1 chmap */ -static void alc662_fixup_bass_chmap(struct hda_codec *codec, +static void alc_fixup_bass_chmap(struct hda_codec *codec, const struct hda_fixup *fix, int action) { if (action == HDA_FIXUP_ACT_BUILD) { @@ -4923,7 +5017,7 @@ static const struct hda_fixup alc662_fixups[] = { }, [ALC662_FIXUP_BASS_CHMAP] = { .type = HDA_FIXUP_FUNC, - .v.func = alc662_fixup_bass_chmap, + .v.func = alc_fixup_bass_chmap, .chained = true, .chain_id = ALC662_FIXUP_ASUS_MODE4 }, @@ -4936,7 +5030,7 @@ static const struct hda_fixup alc662_fixups[] = { }, [ALC662_FIXUP_BASS_1A_CHMAP] = { .type = HDA_FIXUP_FUNC, - .v.func = alc662_fixup_bass_chmap, + .v.func = alc_fixup_bass_chmap, .chained = true, .chain_id = ALC662_FIXUP_BASS_1A, }, @@ -4952,8 +5046,11 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), 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(0x1028, 0x0623, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0624, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0625, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0626, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1028, 0x0628, "Dell", ALC668_FIXUP_DELL_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1632, "HP RP5800", ALC662_FIXUP_HP_RP5800), SND_PCI_QUIRK(0x1043, 0x11cd, "Asus N550", ALC662_FIXUP_BASS_1A_CHMAP), SND_PCI_QUIRK(0x1043, 0x1477, "ASUS N56VZ", ALC662_FIXUP_BASS_CHMAP), @@ -5118,6 +5215,7 @@ static int patch_alc662(struct hda_codec *codec) case 0x10ec0272: case 0x10ec0663: case 0x10ec0665: + case 0x10ec0668: set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT); break; case 0x10ec0273: @@ -5175,6 +5273,7 @@ static int patch_alc680(struct hda_codec *codec) */ static const struct hda_codec_preset snd_hda_preset_realtek[] = { { .id = 0x10ec0221, .name = "ALC221", .patch = patch_alc269 }, + { .id = 0x10ec0231, .name = "ALC231", .patch = patch_alc269 }, { .id = 0x10ec0233, .name = "ALC233", .patch = patch_alc269 }, { .id = 0x10ec0255, .name = "ALC255", .patch = patch_alc269 }, { .id = 0x10ec0260, .name = "ALC260", .patch = patch_alc260 }, diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index d2cc0041d9d3..088a5afbd1b9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2094,7 +2094,8 @@ static void stac92hd83xxx_fixup_hp_mic_led(struct hda_codec *codec, if (action == HDA_FIXUP_ACT_PRE_PROBE) { spec->mic_mute_led_gpio = 0x08; /* GPIO3 */ - codec->bus->avoid_link_reset = 1; + /* resetting controller clears GPIO, so we need to keep on */ + codec->bus->power_keep_link_on = 1; } } diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 8697cedccd21..1ead3c977a51 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -648,7 +648,7 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; - ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); ssc_writel(ssc_p->ssc->regs, IDR, dma_params->mask->ssc_error); pr_debug("%s enabled SSC_SR=0x%08x\n", @@ -657,6 +657,33 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, return 0; } +static int atmel_ssc_trigger(struct snd_pcm_substream *substream, + int cmd, struct snd_soc_dai *dai) +{ + struct atmel_ssc_info *ssc_p = &ssc_info[dai->id]; + struct atmel_pcm_dma_params *dma_params; + int dir; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + dir = 0; + else + dir = 1; + + dma_params = ssc_p->dma_params[dir]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + break; + default: + ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + break; + } + + return 0; +} #ifdef CONFIG_PM static int atmel_ssc_suspend(struct snd_soc_dai *cpu_dai) @@ -731,6 +758,7 @@ static const struct snd_soc_dai_ops atmel_ssc_dai_ops = { .startup = atmel_ssc_startup, .shutdown = atmel_ssc_shutdown, .prepare = atmel_ssc_prepare, + .trigger = atmel_ssc_trigger, .hw_params = atmel_ssc_hw_params, .set_fmt = atmel_ssc_set_dai_fmt, .set_clkdiv = atmel_ssc_set_dai_clkdiv, diff --git a/sound/soc/atmel/sam9x5_wm8731.c b/sound/soc/atmel/sam9x5_wm8731.c index 992ae38d5a15..7d6a9055874b 100644 --- a/sound/soc/atmel/sam9x5_wm8731.c +++ b/sound/soc/atmel/sam9x5_wm8731.c @@ -97,6 +97,8 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) goto out; } + snd_soc_card_set_drvdata(card, priv); + card->dev = &pdev->dev; card->owner = THIS_MODULE; card->dai_link = dai; @@ -107,7 +109,7 @@ static int sam9x5_wm8731_driver_probe(struct platform_device *pdev) dai->stream_name = "WM8731 PCM"; dai->codec_dai_name = "wm8731-hifi"; dai->init = sam9x5_wm8731_init; - dai->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + dai->dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM; ret = snd_soc_of_parse_card_name(card, "atmel,model"); diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index c3c7396a6181..0ab2dc296474 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -248,19 +248,6 @@ ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2L", ARIZONA_OUT6LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT2R", ARIZONA_OUT6RMIX_INPUT_1_SOURCE), -SOC_SINGLE("HPOUT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 1, 0), -SOC_SINGLE("HPOUT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUT2_OSR_SHIFT, 1, 0), -SOC_SINGLE("HPOUT3 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 1, 0), -SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, - ARIZONA_OUT4_OSR_SHIFT, 1, 0), -SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, - ARIZONA_OUT5_OSR_SHIFT, 1, 0), -SOC_SINGLE("SPKDAT2 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_6L, - ARIZONA_OUT6_OSR_SHIFT, 1, 0), - SOC_DOUBLE_R("HPOUT1 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_OUT1L_MUTE_SHIFT, 1, 1), SOC_DOUBLE_R("HPOUT2 Digital Switch", ARIZONA_DAC_DIGITAL_VOLUME_2L, @@ -293,18 +280,6 @@ SOC_DOUBLE_R_TLV("SPKDAT2 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_6L, ARIZONA_DAC_DIGITAL_VOLUME_6R, ARIZONA_OUT6L_VOL_SHIFT, 0xbf, 0, digital_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT1 Volume", ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUTPUT_PATH_CONFIG_1R, - ARIZONA_OUT1L_PGA_VOL_SHIFT, - 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT2 Volume", ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUTPUT_PATH_CONFIG_2R, - ARIZONA_OUT2L_PGA_VOL_SHIFT, - 0x34, 0x40, 0, ana_tlv), -SOC_DOUBLE_R_RANGE_TLV("HPOUT3 Volume", ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUTPUT_PATH_CONFIG_3R, - ARIZONA_OUT3L_PGA_VOL_SHIFT, 0x34, 0x40, 0, ana_tlv), - SOC_DOUBLE("SPKDAT1 Switch", ARIZONA_PDM_SPK1_CTRL_1, ARIZONA_SPK1L_MUTE_SHIFT, ARIZONA_SPK1R_MUTE_SHIFT, 1, 1), SOC_DOUBLE("SPKDAT2 Switch", ARIZONA_PDM_SPK2_CTRL_1, ARIZONA_SPK2L_MUTE_SHIFT, @@ -1037,7 +1012,7 @@ static const struct snd_soc_dapm_route wm5110_dapm_routes[] = { { "AEC Loopback", "HPOUT3L", "OUT3L" }, { "AEC Loopback", "HPOUT3R", "OUT3R" }, { "HPOUT3L", NULL, "OUT3L" }, - { "HPOUT3R", NULL, "OUT3L" }, + { "HPOUT3R", NULL, "OUT3R" }, { "AEC Loopback", "SPKOUTL", "OUT4L" }, { "SPKOUTLN", NULL, "OUT4L" }, diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 456bb8c6d759..bc7472c968e3 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -447,10 +447,10 @@ static int wm8731_set_dai_fmt(struct snd_soc_dai *codec_dai, iface |= 0x0001; break; case SND_SOC_DAIFMT_DSP_A: - iface |= 0x0003; + iface |= 0x0013; break; case SND_SOC_DAIFMT_DSP_B: - iface |= 0x0013; + iface |= 0x0003; break; default: return -EINVAL; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 3938fb1c203e..53bbfac6a83a 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1444,7 +1444,7 @@ static int wm8904_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: - aif1 |= WM8904_AIF_LRCLK_INV; + aif1 |= 0x3 | WM8904_AIF_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x3; break; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 543c5c2631b6..0f17ed3e29f4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2439,7 +2439,20 @@ static void wm8962_configure_bclk(struct snd_soc_codec *codec) snd_soc_update_bits(codec, WM8962_CLOCKING_4, WM8962_SYSCLK_RATE_MASK, clocking4); + /* DSPCLK_DIV can be only generated correctly after enabling SYSCLK. + * So we here provisionally enable it and then disable it afterward + * if current bias_level hasn't reached SND_SOC_BIAS_ON. + */ + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, WM8962_SYSCLK_ENA); + dspclk = snd_soc_read(codec, WM8962_CLOCKING1); + + if (codec->dapm.bias_level != SND_SOC_BIAS_ON) + snd_soc_update_bits(codec, WM8962_CLOCKING2, + WM8962_SYSCLK_ENA_MASK, 0); + if (dspclk < 0) { dev_err(codec->dev, "Failed to read DSPCLK: %d\n", dspclk); return; diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 253c88bb7a4c..4f05fb88bddf 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1259,6 +1259,8 @@ static int wm8990_set_bias_level(struct snd_soc_codec *codec, /* disable POBCTRL, SOFT_ST and BUFDCOPEN */ snd_soc_write(codec, WM8990_ANTIPOP2, 0x0); + + codec->cache_sync = 1; break; } diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 46ec0e9744d4..4fbcab63e61f 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1474,13 +1474,17 @@ static int wm_adsp2_ena(struct wm_adsp *dsp) return ret; /* Wait for the RAM to start, should be near instantaneous */ - count = 0; - do { + for (count = 0; count < 10; ++count) { ret = regmap_read(dsp->regmap, dsp->base + ADSP2_STATUS1, &val); if (ret != 0) return ret; - } while (!(val & ADSP2_RAM_RDY) && ++count < 10); + + if (val & ADSP2_RAM_RDY) + break; + + msleep(1); + } if (!(val & ADSP2_RAM_RDY)) { adsp_err(dsp, "Failed to start DSP RAM\n"); diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c index 61e48852b9e8..3fd76bc391de 100644 --- a/sound/soc/fsl/imx-wm8962.c +++ b/sound/soc/fsl/imx-wm8962.c @@ -130,8 +130,6 @@ static int imx_wm8962_set_bias_level(struct snd_soc_card *card, break; } - dapm->bias_level = level; - return 0; } diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index eb4373840bb6..3665f612819d 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -69,7 +69,6 @@ static int pcm030_fabric_probe(struct platform_device *op) return -ENOMEM; card->dev = &op->dev; - platform_set_drvdata(op, pdata); pdata->card = card; @@ -98,6 +97,8 @@ static int pcm030_fabric_probe(struct platform_device *op) if (ret) dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); + platform_set_drvdata(op, pdata); + return ret; } diff --git a/sound/soc/kirkwood/kirkwood-i2s.c b/sound/soc/kirkwood/kirkwood-i2s.c index d34d91743e3f..3920a5e8125f 100644 --- a/sound/soc/kirkwood/kirkwood-i2s.c +++ b/sound/soc/kirkwood/kirkwood-i2s.c @@ -33,6 +33,10 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) +#define KIRKWOOD_SPDIF_FORMATS \ + (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE) + static int kirkwood_i2s_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { @@ -244,15 +248,15 @@ static int kirkwood_i2s_play_trigger(struct snd_pcm_substream *substream, ctl); } - if (dai->id == 0) - ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ - else - ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ - switch (cmd) { case SNDRV_PCM_TRIGGER_START: /* configure */ ctl = priv->ctl_play; + if (dai->id == 0) + ctl &= ~KIRKWOOD_PLAYCTL_SPDIF_EN; /* i2s */ + else + ctl &= ~KIRKWOOD_PLAYCTL_I2S_EN; /* spdif */ + value = ctl & ~KIRKWOOD_PLAYCTL_ENABLE_MASK; writel(value, priv->io + KIRKWOOD_PLAYCTL); @@ -449,14 +453,14 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai[2] = { .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000, - .formats = KIRKWOOD_I2S_FORMATS, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, }, @@ -469,17 +473,17 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_I2S_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, .formats = KIRKWOOD_I2S_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, @@ -490,18 +494,18 @@ static struct snd_soc_dai_driver kirkwood_i2s_dai_extclk[2] = { .playback = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .capture = { .channels_min = 1, .channels_max = 2, - .rates = SNDRV_PCM_RATE_8000_192000 | - SNDRV_PCM_RATE_CONTINUOUS | - SNDRV_PCM_RATE_KNOT, - .formats = KIRKWOOD_I2S_FORMATS, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 5512, + .rate_max = 192000, + .formats = KIRKWOOD_SPDIF_FORMATS, }, .ops = &kirkwood_i2s_dai_ops, }, diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c index 6d216cb6c19b..3fde9e402710 100644 --- a/sound/soc/omap/n810.c +++ b/sound/soc/omap/n810.c @@ -100,12 +100,12 @@ static int n810_startup(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2); n810_ext_control(&codec->dapm); - return clk_enable(sys_clkout2); + return clk_prepare_enable(sys_clkout2); } static void n810_shutdown(struct snd_pcm_substream *substream) { - clk_disable(sys_clkout2); + clk_disable_unprepare(sys_clkout2); } static int n810_hw_params(struct snd_pcm_substream *substream, diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 14011d90d70a..ff60e11ecb56 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" select SND_SIMPLE_CARD + select REGMAP help This option enables R-Car SUR/SCU/SSIU/SSI sound support diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 4e53d87e881d..a66783e13a9c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3212,11 +3212,11 @@ int snd_soc_bytes_get(struct snd_kcontrol *kcontrol, break; case 2: ((u16 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be16(~params->mask); break; case 4: ((u32 *)(&ucontrol->value.bytes.data))[0] - &= ~params->mask; + &= cpu_to_be32(~params->mask); break; default: return -EINVAL; diff --git a/sound/soc/soc-devres.c b/sound/soc/soc-devres.c index b1d732255c02..3449c1e909ae 100644 --- a/sound/soc/soc-devres.c +++ b/sound/soc/soc-devres.c @@ -66,7 +66,7 @@ static void devm_card_release(struct device *dev, void *res) */ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) { - struct device **ptr; + struct snd_soc_card **ptr; int ret; ptr = devres_alloc(devm_card_release, sizeof(*ptr), GFP_KERNEL); @@ -75,7 +75,7 @@ int devm_snd_soc_register_card(struct device *dev, struct snd_soc_card *card) ret = snd_soc_register_card(card); if (ret == 0) { - *ptr = dev; + *ptr = card; devres_add(dev, ptr); } else { devres_free(ptr); diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index cbc9c96ce1f4..41949af3baae 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -305,6 +305,20 @@ static void dmaengine_pcm_request_chan_of(struct dmaengine_pcm *pcm, } } +static void dmaengine_pcm_release_chan(struct dmaengine_pcm *pcm) +{ + unsigned int i; + + for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; + i++) { + if (!pcm->chan[i]) + continue; + dma_release_channel(pcm->chan[i]); + if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) + break; + } +} + /** * snd_dmaengine_pcm_register - Register a dmaengine based PCM device * @dev: The parent device for the PCM device @@ -315,6 +329,7 @@ int snd_dmaengine_pcm_register(struct device *dev, const struct snd_dmaengine_pcm_config *config, unsigned int flags) { struct dmaengine_pcm *pcm; + int ret; pcm = kzalloc(sizeof(*pcm), GFP_KERNEL); if (!pcm) @@ -326,11 +341,20 @@ int snd_dmaengine_pcm_register(struct device *dev, dmaengine_pcm_request_chan_of(pcm, dev); if (flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) - return snd_soc_add_platform(dev, &pcm->platform, + ret = snd_soc_add_platform(dev, &pcm->platform, &dmaengine_no_residue_pcm_platform); else - return snd_soc_add_platform(dev, &pcm->platform, + ret = snd_soc_add_platform(dev, &pcm->platform, &dmaengine_pcm_platform); + if (ret) + goto err_free_dma; + + return 0; + +err_free_dma: + dmaengine_pcm_release_chan(pcm); + kfree(pcm); + return ret; } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_register); @@ -345,7 +369,6 @@ void snd_dmaengine_pcm_unregister(struct device *dev) { struct snd_soc_platform *platform; struct dmaengine_pcm *pcm; - unsigned int i; platform = snd_soc_lookup_platform(dev); if (!platform) @@ -353,15 +376,8 @@ void snd_dmaengine_pcm_unregister(struct device *dev) pcm = soc_platform_to_pcm(platform); - for (i = SNDRV_PCM_STREAM_PLAYBACK; i <= SNDRV_PCM_STREAM_CAPTURE; i++) { - if (pcm->chan[i]) { - dma_release_channel(pcm->chan[i]); - if (pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) - break; - } - } - snd_soc_remove_platform(platform); + dmaengine_pcm_release_chan(pcm); kfree(pcm); } EXPORT_SYMBOL_GPL(snd_dmaengine_pcm_unregister); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 42782c01e413..891b9a9bcbf8 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -148,12 +148,12 @@ static void soc_pcm_apply_msb(struct snd_pcm_substream *substream, } } -static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, +static void soc_pcm_init_runtime_hw(struct snd_pcm_runtime *runtime, struct snd_soc_pcm_stream *codec_stream, struct snd_soc_pcm_stream *cpu_stream) { - hw->rate_min = max(codec_stream->rate_min, cpu_stream->rate_min); - hw->rate_max = max(codec_stream->rate_max, cpu_stream->rate_max); + struct snd_pcm_hardware *hw = &runtime->hw; + hw->channels_min = max(codec_stream->channels_min, cpu_stream->channels_min); hw->channels_max = min(codec_stream->channels_max, @@ -166,6 +166,13 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_hardware *hw, if (cpu_stream->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS)) hw->rates |= codec_stream->rates; + + snd_pcm_limit_hw_rates(runtime); + + hw->rate_min = max(hw->rate_min, cpu_stream->rate_min); + hw->rate_min = max(hw->rate_min, codec_stream->rate_min); + hw->rate_max = min_not_zero(hw->rate_max, cpu_stream->rate_max); + hw->rate_max = min_not_zero(hw->rate_max, codec_stream->rate_max); } /* @@ -235,15 +242,14 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) /* Check that the codec and cpu DAIs are compatible */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->playback, + soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->playback, &cpu_dai_drv->playback); } else { - soc_pcm_init_runtime_hw(&runtime->hw, &codec_dai_drv->capture, + soc_pcm_init_runtime_hw(runtime, &codec_dai_drv->capture, &cpu_dai_drv->capture); } ret = -EINVAL; - snd_pcm_limit_hw_rates(runtime); if (!runtime->hw.rates) { printk(KERN_ERR "ASoC: %s <-> %s No matching rates\n", codec_dai->name, cpu_dai->name); @@ -594,12 +600,13 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) struct snd_soc_platform *platform = rtd->platform; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai *codec_dai = rtd->codec_dai; - struct snd_soc_codec *codec = rtd->codec; + bool playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* apply codec digital mute */ - if (!codec->active) + if ((playback && codec_dai->playback_active == 1) || + (!playback && codec_dai->capture_active == 1)) snd_soc_dai_digital_mute(codec_dai, 1, substream->stream); /* free any machine hw params */ diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 364bf6a907e1..8c819f811470 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -74,7 +74,7 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra20_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -83,10 +83,10 @@ static int tegra20_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA20_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA20_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA20_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA20_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 08bc6931c7c7..8c7c1028e579 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -67,15 +67,15 @@ static int tegra20_spdif_hw_params(struct snd_pcm_substream *substream, { struct device *dev = dai->dev; struct tegra20_spdif *spdif = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; int ret, spdifclock; - mask = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; + mask |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_MASK; switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: - val = TEGRA20_SPDIF_CTRL_PACK | - TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; + val |= TEGRA20_SPDIF_CTRL_PACK | + TEGRA20_SPDIF_CTRL_BIT_MODE_16BIT; break; default: return -EINVAL; diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index 231a785b3921..02247fee1cf7 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -118,7 +118,7 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) { struct tegra30_i2s *i2s = snd_soc_dai_get_drvdata(dai); - unsigned int mask, val; + unsigned int mask = 0, val = 0; switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: @@ -127,10 +127,10 @@ static int tegra30_i2s_set_fmt(struct snd_soc_dai *dai, return -EINVAL; } - mask = TEGRA30_I2S_CTRL_MASTER_ENABLE; + mask |= TEGRA30_I2S_CTRL_MASTER_ENABLE; switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBS_CFS: - val = TEGRA30_I2S_CTRL_MASTER_ENABLE; + val |= TEGRA30_I2S_CTRL_MASTER_ENABLE; break; case SND_SOC_DAIFMT_CBM_CFM: break; diff --git a/sound/usb/endpoint.c b/sound/usb/endpoint.c index b9ba0fcc45df..83aabea259d7 100644 --- a/sound/usb/endpoint.c +++ b/sound/usb/endpoint.c @@ -636,8 +636,22 @@ static int data_ep_set_params(struct snd_usb_endpoint *ep, if (usb_pipein(ep->pipe) || snd_usb_endpoint_implicit_feedback_sink(ep)) { + urb_packs = packs_per_ms; + /* + * Wireless devices can poll at a max rate of once per 4ms. + * For dataintervals less than 5, increase the packet count to + * allow the host controller to use bursting to fill in the + * gaps. + */ + if (snd_usb_get_speed(ep->chip->dev) == USB_SPEED_WIRELESS) { + int interval = ep->datainterval; + while (interval < 5) { + urb_packs <<= 1; + ++interval; + } + } /* make capture URBs <= 1 ms and smaller than a period */ - urb_packs = min(max_packs_per_urb, packs_per_ms); + urb_packs = min(max_packs_per_urb, urb_packs); while (urb_packs > 1 && urb_packs * maxsize >= period_bytes) urb_packs >>= 1; ep->nurbs = MAX_URBS; diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 3454262358b3..f4b12c216f1c 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1603,7 +1603,7 @@ static int snd_microii_controls_create(struct usb_mixer_interface *mixer) return err; } - return err; + return 0; } int snd_usb_mixer_apply_create_quirk(struct usb_mixer_interface *mixer) |