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sound fixes for 4.1-rc1
Here are a few fixes that have been pending since the previous pull request: a regression fix for HD-audio multiple SPDIF / HDMI devices, several ALC256 codec fixes, a couple of i915 HDMI audio fixes, and various small fixes. Nothing exciting, just boring, but things good to have. -----BEGIN PGP SIGNATURE----- Version: GnuPG v2 iQIcBAABCAAGBQJVOliOAAoJEGwxgFQ9KSmkHZsQAIal+ho7D4Y+zPALKipfVOgG /LQmClH9FsdYTbISGv1pYjXmqZRtHAknX0G3eQNur6wEQRk5Cf6dxu38M39ec5kj /N/kRkwjrdnweTDrMy6pLQswpG0I4d2H7i3ooojraOHCDrQv/u2pyhniBZv6+LWi J7GH4Sr6yE8VHSGF9djQMc2u4Q4YKFYwOqrUYjrKkVA1IhdKATwCoNaUbHNf6yWV KKSW5NZc+vX1V0AoqTVeJ27xwXTIhVyhKYzxWCf78XdAY+Am3ctHbZanwvu5GwTs F2T5UbZdI/pVIdik5yuuuwIB84I+yIR6HAgevC0i49nNaV6NEtr9bB/4MTQJELcI 4asfcNmT7AhgfPxEvieqXCKqRpK2Cw0GxezYPYFfNj5jLv2m9lf5HOnpPHZgE6wv ppmtaCEVUxCXAKVZWW1LEkot5niv3dMzfFCb6kSuOO1kJv3AkyxMFsVYxkN96kNM gdh5NPLm1eO3pdUVv1ZW2Ur77rEMLqCnJ0g6EeRDwT/RwFzHFZsZq94ZX1LEuYhd xwKMWPUQYnDZKMWfZBlfl+Mvs3uWjnwvPw5h0a0gsD2E4F7SQql8t2oLGqmwqJsC YIGhbh+QM+su7jPyJ03+YMYGBylTv1g0Z19AQX5hiPBz4KjuP6ytdBBmaci8VlYM NSAYGBe8+vfaF8mTZ+6i =TICa -----END PGP SIGNATURE----- Merge tag 'sound-fix-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound Pull sound fixes from Takashi Iwai: "Here are a few fixes that have been pending since the previous pull request: a regression fix for HD-audio multiple SPDIF / HDMI devices, several ALC256 codec fixes, a couple of i915 HDMI audio fixes, and various small fixes. Nothing exciting, just boring, but things good to have" * tag 'sound-fix-4.1-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: ALSA: hda - fix headset mic detection problem for one more machine ALSA: hda/realtek - Fix Headphone Mic doesn't recording for ALC256 ALSA: hda - fix "num_steps = 0" error on ALC256 ALSA: usb-audio: Fix audio output on Roland SC-D70 sound module ALSA: hda - add AZX_DCAPS_I915_POWERWELL to Baytrail ALSA: hda - only sync BCLK to the display clock for Haswell & Broadwell ALSA: hda - Mute headphone pin on suspend on XPS13 9333 sound/oss: fix deadlock in sequencer_ioctl(SNDCTL_SEQ_OUTOFBAND) ALSA: asound.h - use SNDRV_CTL_ELEM_ID_NAME_MAXLEN ALSA: hda - potential (but unlikely) uninitialized variable ALSA: hda - Fix regression for slave SPDIF setups ALSA: intel8x0: Check pci_iomap() success for DEVICE_ALI ALSA: hda - simplify azx_has_pm_runtime
This commit is contained in:
commit
d19d133e43
@ -864,7 +864,7 @@ struct snd_ctl_elem_id {
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snd_ctl_elem_iface_t iface; /* interface identifier */
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unsigned int device; /* device/client number */
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unsigned int subdevice; /* subdevice (substream) number */
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unsigned char name[44]; /* ASCII name of item */
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unsigned char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; /* ASCII name of item */
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unsigned int index; /* index of item */
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};
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@ -681,13 +681,8 @@ static int seq_timing_event(unsigned char *event_rec)
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break;
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case TMR_ECHO:
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if (seq_mode == SEQ_2)
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seq_copy_to_input(event_rec, 8);
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else
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{
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parm = (parm << 8 | SEQ_ECHO);
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seq_copy_to_input((unsigned char *) &parm, 4);
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}
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parm = (parm << 8 | SEQ_ECHO);
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seq_copy_to_input((unsigned char *) &parm, 4);
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break;
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default:;
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@ -1324,7 +1319,6 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a
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int mode = translate_mode(file);
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struct synth_info inf;
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struct seq_event_rec event_rec;
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unsigned long flags;
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int __user *p = arg;
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orig_dev = dev = dev >> 4;
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@ -1479,9 +1473,7 @@ int sequencer_ioctl(int dev, struct file *file, unsigned int cmd, void __user *a
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case SNDCTL_SEQ_OUTOFBAND:
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if (copy_from_user(&event_rec, arg, sizeof(event_rec)))
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return -EFAULT;
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spin_lock_irqsave(&lock,flags);
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play_event(event_rec.arr);
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spin_unlock_irqrestore(&lock,flags);
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return 0;
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case SNDCTL_MIDI_INFO:
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@ -2529,7 +2529,7 @@ static void set_dig_out(struct hda_codec *codec, hda_nid_t nid,
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if (!d)
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return;
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for (; *d; d++)
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snd_hdac_regmap_update(&codec->core, nid,
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snd_hdac_regmap_update(&codec->core, *d,
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AC_VERB_SET_DIGI_CONVERT_1, mask, val);
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}
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@ -404,7 +404,7 @@ struct azx {
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((chip)->ops->reg_readb((dev)->sd_addr + AZX_REG_##reg))
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#define azx_has_pm_runtime(chip) \
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(!AZX_DCAPS_PM_RUNTIME || ((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME))
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((chip)->driver_caps & AZX_DCAPS_PM_RUNTIME)
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/* PCM setup */
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static inline struct azx_dev *get_azx_dev(struct snd_pcm_substream *substream)
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@ -55,6 +55,12 @@ void haswell_set_bclk(struct hda_intel *hda)
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int cdclk_freq;
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unsigned int bclk_m, bclk_n;
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struct i915_audio_component *acomp = &hda->audio_component;
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struct pci_dev *pci = hda->chip.pci;
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/* Only Haswell/Broadwell need set BCLK */
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if (pci->device != 0x0a0c && pci->device != 0x0c0c
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&& pci->device != 0x0d0c && pci->device != 0x160c)
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return;
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if (!acomp->ops)
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return;
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@ -297,6 +297,9 @@ enum {
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AZX_DCAPS_PM_RUNTIME | AZX_DCAPS_I915_POWERWELL |\
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AZX_DCAPS_SNOOP_TYPE(SCH))
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#define AZX_DCAPS_INTEL_BAYTRAIL \
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(AZX_DCAPS_INTEL_PCH_NOPM | AZX_DCAPS_I915_POWERWELL)
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#define AZX_DCAPS_INTEL_BRASWELL \
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(AZX_DCAPS_INTEL_PCH | AZX_DCAPS_I915_POWERWELL)
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@ -1992,7 +1995,7 @@ static const struct pci_device_id azx_ids[] = {
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.driver_data = AZX_DRIVER_SCH | AZX_DCAPS_INTEL_PCH_NOPM },
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/* BayTrail */
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{ PCI_DEVICE(0x8086, 0x0f04),
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.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_PCH_NOPM },
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.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BAYTRAIL },
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/* Braswell */
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{ PCI_DEVICE(0x8086, 0x2284),
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.driver_data = AZX_DRIVER_PCH | AZX_DCAPS_INTEL_BRASWELL },
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@ -582,8 +582,8 @@ static void print_conn_list(struct snd_info_buffer *buffer,
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/* Get Cache connections info */
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cache_len = snd_hda_get_conn_list(codec, nid, &list);
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if (cache_len != conn_len
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|| memcmp(list, conn, conn_len)) {
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if (cache_len >= 0 && (cache_len != conn_len ||
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memcmp(list, conn, conn_len) != 0)) {
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snd_iprintf(buffer, " In-driver Connection: %d\n", cache_len);
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if (cache_len > 0) {
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snd_iprintf(buffer, " ");
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@ -4176,17 +4176,15 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec,
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}
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}
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static unsigned int alc_power_filter_xps13(struct hda_codec *codec,
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hda_nid_t nid,
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unsigned int power_state)
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static void alc_shutup_dell_xps13(struct hda_codec *codec)
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{
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struct alc_spec *spec = codec->spec;
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int hp_pin = spec->gen.autocfg.hp_pins[0];
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/* Avoid pop noises when headphones are plugged in */
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if (spec->gen.hp_jack_present)
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if (nid == codec->core.afg || nid == 0x02 || nid == 0x15)
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return AC_PWRST_D0;
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return snd_hda_gen_path_power_filter(codec, nid, power_state);
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/* Prevent pop noises when headphones are plugged in */
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snd_hda_codec_write(codec, hp_pin, 0,
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AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE);
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msleep(20);
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}
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static void alc_fixup_dell_xps13(struct hda_codec *codec,
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@ -4197,8 +4195,7 @@ static void alc_fixup_dell_xps13(struct hda_codec *codec,
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struct hda_input_mux *imux = &spec->gen.input_mux;
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int i;
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spec->shutup = alc_no_shutup;
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codec->power_filter = alc_power_filter_xps13;
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spec->shutup = alc_shutup_dell_xps13;
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/* Make the internal mic the default input source. */
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for (i = 0; i < imux->num_items; i++) {
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@ -5231,6 +5228,16 @@ static const struct hda_model_fixup alc269_fixup_models[] = {
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{0x1b, 0x411111f0}, \
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{0x1e, 0x411111f0}
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#define ALC256_STANDARD_PINS \
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{0x12, 0x90a60140}, \
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{0x14, 0x90170110}, \
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{0x19, 0x411111f0}, \
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{0x1a, 0x411111f0}, \
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{0x1b, 0x411111f0}, \
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{0x1d, 0x40700001}, \
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{0x1e, 0x411111f0}, \
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{0x21, 0x02211020}
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#define ALC282_STANDARD_PINS \
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{0x14, 0x90170110}, \
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{0x18, 0x411111f0}, \
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@ -5331,15 +5338,11 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = {
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{0x1d, 0x40700001},
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{0x21, 0x02211050}),
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SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
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{0x12, 0x90a60140},
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{0x13, 0x40000000},
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{0x14, 0x90170110},
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{0x19, 0x411111f0},
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{0x1a, 0x411111f0},
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{0x1b, 0x411111f0},
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{0x1d, 0x40700001},
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{0x1e, 0x411111f0},
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{0x21, 0x02211020}),
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ALC256_STANDARD_PINS,
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{0x13, 0x40000000}),
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SND_HDA_PIN_QUIRK(0x10ec0256, 0x1028, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE,
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ALC256_STANDARD_PINS,
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{0x13, 0x411111f0}),
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SND_HDA_PIN_QUIRK(0x10ec0280, 0x103c, "HP", ALC280_FIXUP_HP_GPIO4,
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{0x12, 0x90a60130},
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{0x13, 0x40000000},
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@ -5667,6 +5670,8 @@ static int patch_alc269(struct hda_codec *codec)
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break;
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case 0x10ec0256:
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spec->codec_variant = ALC269_TYPE_ALC256;
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spec->gen.mixer_nid = 0; /* ALC256 does not have any loopback mixer path */
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alc_update_coef_idx(codec, 0x36, 1 << 13, 1 << 5); /* Switch pcbeep path to Line in path*/
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break;
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}
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@ -5680,8 +5685,8 @@ static int patch_alc269(struct hda_codec *codec)
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if (err < 0)
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goto error;
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if (!spec->gen.no_analog && spec->gen.beep_nid)
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set_beep_amp(spec, 0x0b, 0x04, HDA_INPUT);
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if (!spec->gen.no_analog && spec->gen.beep_nid && spec->gen.mixer_nid)
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set_beep_amp(spec, spec->gen.mixer_nid, 0x04, HDA_INPUT);
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codec->patch_ops = alc_patch_ops;
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codec->patch_ops.stream_pm = snd_hda_gen_stream_pm;
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@ -3101,13 +3101,13 @@ static int snd_intel8x0_create(struct snd_card *card,
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chip->bmaddr = pci_iomap(pci, 3, 0);
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else
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chip->bmaddr = pci_iomap(pci, 1, 0);
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port_inited:
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if (!chip->bmaddr) {
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dev_err(card->dev, "Controller space ioremap problem\n");
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snd_intel8x0_free(chip);
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return -EIO;
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}
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port_inited:
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chip->bdbars_count = bdbars[device_type];
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/* initialize offsets */
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@ -79,7 +79,10 @@ static u64 parse_audio_format_i_type(struct snd_usb_audio *chip,
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format = 1 << UAC_FORMAT_TYPE_I_PCM;
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}
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if (format & (1 << UAC_FORMAT_TYPE_I_PCM)) {
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if (chip->usb_id == USB_ID(0x0582, 0x0016) /* Edirol SD-90 */ &&
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if (((chip->usb_id == USB_ID(0x0582, 0x0016)) ||
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/* Edirol SD-90 */
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(chip->usb_id == USB_ID(0x0582, 0x000c))) &&
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/* Roland SC-D70 */
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sample_width == 24 && sample_bytes == 2)
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sample_bytes = 3;
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else if (sample_width > sample_bytes * 8) {
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@ -816,37 +816,11 @@ YAMAHA_DEVICE(0x7010, "UB99"),
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.data = (const struct snd_usb_audio_quirk[]) {
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{
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.ifnum = 0,
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.type = QUIRK_AUDIO_FIXED_ENDPOINT,
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.data = & (const struct audioformat) {
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.formats = SNDRV_PCM_FMTBIT_S24_3LE,
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.channels = 2,
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.iface = 0,
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.altsetting = 1,
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.altset_idx = 1,
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.attributes = 0,
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.endpoint = 0x01,
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.ep_attr = 0x01,
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.rates = SNDRV_PCM_RATE_CONTINUOUS,
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.rate_min = 44100,
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.rate_max = 44100,
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}
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.type = QUIRK_AUDIO_STANDARD_INTERFACE
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},
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{
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.ifnum = 1,
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.type = QUIRK_AUDIO_FIXED_ENDPOINT,
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.data = & (const struct audioformat) {
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.formats = SNDRV_PCM_FMTBIT_S24_3LE,
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.channels = 2,
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.iface = 1,
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.altsetting = 1,
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.altset_idx = 1,
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.attributes = 0,
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.endpoint = 0x81,
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.ep_attr = 0x01,
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.rates = SNDRV_PCM_RATE_CONTINUOUS,
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.rate_min = 44100,
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.rate_max = 44100,
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}
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.type = QUIRK_AUDIO_STANDARD_INTERFACE
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},
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{
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.ifnum = 2,
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