diff options
author | Takashi Iwai <tiwai@suse.de> | 2011-05-02 08:41:40 (GMT) |
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committer | Takashi Iwai <tiwai@suse.de> | 2011-05-02 08:41:40 (GMT) |
commit | a3ea8e8f2474c35b4c3e22262991afddb93c4c0e (patch) | |
tree | 648bbeea304950d02e71863673be8154a5e491f6 /sound/pci | |
parent | c2de187e5b0f25b572ac1cb6cdf383f16123717d (diff) | |
parent | ebb47241ea0eac6a5a23404821a2d62f64c68496 (diff) | |
download | linux-fsl-qoriq-a3ea8e8f2474c35b4c3e22262991afddb93c4c0e.tar.xz |
Merge branch 'fix/hda' into topic/hda
Diffstat (limited to 'sound/pci')
36 files changed, 77 insertions, 78 deletions
diff --git a/sound/pci/ad1889.c b/sound/pci/ad1889.c index 4382d0f..d8f6fd6 100644 --- a/sound/pci/ad1889.c +++ b/sound/pci/ad1889.c @@ -29,7 +29,7 @@ * PM support * MIDI support * Game Port support - * SG DMA support (this will need *alot* of work) + * SG DMA support (this will need *a lot* of work) */ #include <linux/init.h> diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index f53a31e..f8ccc96 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -963,7 +963,7 @@ static int snd_card_asihpi_playback_open(struct snd_pcm_substream *substream) /*? also check ASI5000 samplerate source If external, only support external rate. - If internal and other stream playing, cant switch + If internal and other stream playing, can't switch */ init_timer(&dpcm->timer); diff --git a/sound/pci/asihpi/hpi.h b/sound/pci/asihpi/hpi.h index 6fc025c..255429c 100644 --- a/sound/pci/asihpi/hpi.h +++ b/sound/pci/asihpi/hpi.h @@ -725,7 +725,7 @@ enum HPI_AESEBU_ERRORS { #define HPI_PAD_TITLE_LEN 64 /** The text string containing the comment. */ #define HPI_PAD_COMMENT_LEN 256 -/** The PTY when the tuner has not recieved any PTY. */ +/** The PTY when the tuner has not received any PTY. */ #define HPI_PAD_PROGRAM_TYPE_INVALID 0xffff /** \} */ diff --git a/sound/pci/asihpi/hpi6000.c b/sound/pci/asihpi/hpi6000.c index 3e3c2ef..8c8aac4 100644 --- a/sound/pci/asihpi/hpi6000.c +++ b/sound/pci/asihpi/hpi6000.c @@ -423,7 +423,7 @@ static void subsys_create_adapter(struct hpi_message *phm, ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL); if (!ao.priv) { - HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n"); + HPI_DEBUG_LOG(ERROR, "can't get mem for adapter object\n"); phr->error = HPI_ERROR_MEMORY_ALLOC; return; } diff --git a/sound/pci/asihpi/hpi6205.c b/sound/pci/asihpi/hpi6205.c index 620525b..22e9f08 100644 --- a/sound/pci/asihpi/hpi6205.c +++ b/sound/pci/asihpi/hpi6205.c @@ -466,7 +466,7 @@ static void subsys_create_adapter(struct hpi_message *phm, ao.priv = kzalloc(sizeof(struct hpi_hw_obj), GFP_KERNEL); if (!ao.priv) { - HPI_DEBUG_LOG(ERROR, "cant get mem for adapter object\n"); + HPI_DEBUG_LOG(ERROR, "can't get mem for adapter object\n"); phr->error = HPI_ERROR_MEMORY_ALLOC; return; } diff --git a/sound/pci/asihpi/hpi_internal.h b/sound/pci/asihpi/hpi_internal.h index af678be..3b9fd11 100644 --- a/sound/pci/asihpi/hpi_internal.h +++ b/sound/pci/asihpi/hpi_internal.h @@ -607,7 +607,7 @@ struct hpi_data_compat32 { #endif struct hpi_buffer { - /** placehoder for backward compatability (see dwBufferSize) */ + /** placehoder for backward compatibility (see dwBufferSize) */ struct hpi_msg_format reserved; u32 command; /**< HPI_BUFFER_CMD_xxx*/ u32 pci_address; /**< PCI physical address of buffer for DSP DMA */ diff --git a/sound/pci/asihpi/hpimsgx.c b/sound/pci/asihpi/hpimsgx.c index bcbdf30..360028b 100644 --- a/sound/pci/asihpi/hpimsgx.c +++ b/sound/pci/asihpi/hpimsgx.c @@ -722,7 +722,7 @@ static u16 HPIMSGX__init(struct hpi_message *phm, return phr->error; } if (hr.error == 0) { - /* the adapter was created succesfully + /* the adapter was created successfully save the mapping for future use */ hpi_entry_points[hr.u.s.adapter_index] = entry_point_func; /* prepare adapter (pre-open streams etc.) */ diff --git a/sound/pci/au88x0/au88x0.h b/sound/pci/au88x0/au88x0.h index ecb8f4d..02f6e08 100644 --- a/sound/pci/au88x0/au88x0.h +++ b/sound/pci/au88x0/au88x0.h @@ -104,7 +104,7 @@ #define MIX_PLAYB(x) (vortex->mixplayb[x]) #define MIX_SPDIF(x) (vortex->mixspdif[x]) -#define NR_WTPB 0x20 /* WT channels per eahc bank. */ +#define NR_WTPB 0x20 /* WT channels per each bank. */ /* Structs */ typedef struct { diff --git a/sound/pci/au88x0/au88x0_a3d.c b/sound/pci/au88x0/au88x0_a3d.c index f4aa8ff..9ae8b3b 100644 --- a/sound/pci/au88x0/au88x0_a3d.c +++ b/sound/pci/au88x0/au88x0_a3d.c @@ -53,7 +53,7 @@ a3dsrc_GetTimeConsts(a3dsrc_t * a, short *HrtfTrack, short *ItdTrack, } #endif -/* Atmospheric absorbtion. */ +/* Atmospheric absorption. */ static void a3dsrc_SetAtmosTarget(a3dsrc_t * a, short aa, short b, short c, short d, @@ -835,7 +835,7 @@ snd_vortex_a3d_filter_put(struct snd_kcontrol *kcontrol, params[i] = ucontrol->value.integer.value[i]; /* Translate generic filter params to a3d filter params. */ vortex_a3d_translate_filter(a->filter, params); - /* Atmospheric absorbtion and filtering. */ + /* Atmospheric absorption and filtering. */ a3dsrc_SetAtmosTarget(a, a->filter[0], a->filter[1], a->filter[2], a->filter[3], a->filter[4]); diff --git a/sound/pci/au88x0/au88x0_pcm.c b/sound/pci/au88x0/au88x0_pcm.c index 5439d66..62e9591 100644 --- a/sound/pci/au88x0/au88x0_pcm.c +++ b/sound/pci/au88x0/au88x0_pcm.c @@ -44,10 +44,10 @@ static struct snd_pcm_hardware snd_vortex_playback_hw_adb = { .channels_min = 1, .channels_max = 2, .buffer_bytes_max = 0x10000, - .period_bytes_min = 0x1, + .period_bytes_min = 0x20, .period_bytes_max = 0x1000, .periods_min = 2, - .periods_max = 32, + .periods_max = 1024, }; #ifndef CHIP_AU8820 @@ -140,6 +140,9 @@ static int snd_vortex_pcm_open(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_PERIOD_BYTES)) < 0) return err; + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 64); + if (VORTEX_PCM_TYPE(substream->pcm) != VORTEX_PCM_WT) { #ifndef CHIP_AU8820 if (VORTEX_PCM_TYPE(substream->pcm) == VORTEX_PCM_A3D) { @@ -515,7 +518,7 @@ static int __devinit snd_vortex_new_pcm(vortex_t *chip, int idx, int nr) return -ENODEV; /* idx indicates which kind of PCM device. ADB, SPDIF, I2S and A3D share the - * same dma engine. WT uses it own separate dma engine whcih cant capture. */ + * same dma engine. WT uses it own separate dma engine which can't capture. */ if (idx == VORTEX_PCM_ADB) nr_capt = nr; else diff --git a/sound/pci/azt3328.c b/sound/pci/azt3328.c index 5715c4d0..9b7a634 100644 --- a/sound/pci/azt3328.c +++ b/sound/pci/azt3328.c @@ -140,7 +140,7 @@ * Possible remedies: * - use speaker (amplifier) output instead of headphone output * (in case crackling is due to overloaded output clipping) - * - plug card into a different PCI slot, preferrably one that isn't shared + * - plug card into a different PCI slot, preferably one that isn't shared * too much (this helps a lot, but not completely!) * - get rid of PCI VGA card, use AGP instead * - upgrade or downgrade BIOS diff --git a/sound/pci/ca0106/ca0106.h b/sound/pci/ca0106/ca0106.h index fc53b9b..e8e8ccc 100644 --- a/sound/pci/ca0106/ca0106.h +++ b/sound/pci/ca0106/ca0106.h @@ -51,7 +51,7 @@ * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify @@ -175,7 +175,7 @@ /* CA0106 pointer-offset register set, accessed through the PTR and DATA registers */ /********************************************************************************************************/ -/* Initally all registers from 0x00 to 0x3f have zero contents. */ +/* Initially all registers from 0x00 to 0x3f have zero contents. */ #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ /* One list entry: 4 bytes for DMA address, * 4 bytes for period_size << 16. @@ -223,7 +223,7 @@ * The jack has 4 poles. I will call 1 - Tip, 2 - Next to 1, 3 - Next to 2, 4 - Next to 3 * For Analogue: 1 -> Center Speaker, 2 -> Sub Woofer, 3 -> Ground, 4 -> Ground * For Digital: 1 -> Front SPDIF, 2 -> Rear SPDIF, 3 -> Center/Subwoofer SPDIF, 4 -> Ground. - * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Sheild on all three, 4 -> Red. + * Standard 4 pole Video A/V cable with RCA outputs: 1 -> White, 2 -> Yellow, 3 -> Shield on all three, 4 -> Red. * So, from this you can see that you cannot use a Standard 4 pole Video A/V cable with the SB Audigy LS card. */ /* The Front SPDIF PCM gets mixed with samples from the AC97 codec, so can only work for Stereo PCM and not AC3/DTS diff --git a/sound/pci/ca0106/ca0106_main.c b/sound/pci/ca0106/ca0106_main.c index 01b4938..4377592 100644 --- a/sound/pci/ca0106/ca0106_main.c +++ b/sound/pci/ca0106/ca0106_main.c @@ -117,7 +117,7 @@ * DAC: Unknown * Trying to handle it like the SB0410. * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/ca0106/ca0106_mixer.c b/sound/pci/ca0106/ca0106_mixer.c index 630aa49..84f3f92 100644 --- a/sound/pci/ca0106/ca0106_mixer.c +++ b/sound/pci/ca0106/ca0106_mixer.c @@ -42,7 +42,7 @@ * 0.0.18 * Add support for mute control on SB Live 24bit (cards w/ SPI DAC) * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/ca0106/ca0106_proc.c b/sound/pci/ca0106/ca0106_proc.c index ba96428..c694464 100644 --- a/sound/pci/ca0106/ca0106_proc.c +++ b/sound/pci/ca0106/ca0106_proc.c @@ -42,7 +42,7 @@ * 0.0.18 * Implement support for Line-in capture on SB Live 24bit. * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/cmipci.c b/sound/pci/cmipci.c index b5bb036..f4e5735 100644 --- a/sound/pci/cmipci.c +++ b/sound/pci/cmipci.c @@ -73,7 +73,7 @@ MODULE_PARM_DESC(mpu_port, "MPU-401 port."); module_param_array(fm_port, long, NULL, 0444); MODULE_PARM_DESC(fm_port, "FM port."); module_param_array(soft_ac3, bool, NULL, 0444); -MODULE_PARM_DESC(soft_ac3, "Sofware-conversion of raw SPDIF packets (model 033 only)."); +MODULE_PARM_DESC(soft_ac3, "Software-conversion of raw SPDIF packets (model 033 only)."); #ifdef SUPPORT_JOYSTICK module_param_array(joystick_port, int, NULL, 0444); MODULE_PARM_DESC(joystick_port, "Joystick port address."); @@ -656,8 +656,8 @@ out: } /* - * Program pll register bits, I assume that the 8 registers 0xf8 upto 0xff - * are mapped onto the 8 ADC/DAC sampling frequency which can be choosen + * Program pll register bits, I assume that the 8 registers 0xf8 up to 0xff + * are mapped onto the 8 ADC/DAC sampling frequency which can be chosen * at the register CM_REG_FUNCTRL1 (0x04). * Problem: other ways are also possible (any information about that?) */ @@ -666,7 +666,7 @@ static void snd_cmipci_set_pll(struct cmipci *cm, unsigned int rate, unsigned in unsigned int reg = CM_REG_PLL + slot; /* * Guess that this programs at reg. 0x04 the pos 15:13/12:10 - * for DSFC/ASFC (000 upto 111). + * for DSFC/ASFC (000 up to 111). */ /* FIXME: Init (Do we've to set an other register first before programming?) */ diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index b932154..13f33c0 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -1627,7 +1627,7 @@ static struct ct_atc atc_preset __devinitdata = { * Creates and initializes a hardware manager. * * Creates kmallocated ct_atc structure. Initializes hardware. - * Returns 0 if suceeds, or negative error code if fails. + * Returns 0 if succeeds, or negative error code if fails. */ int __devinit ct_atc_create(struct snd_card *card, struct pci_dev *pci, diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 0cf400f..a5c957d 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1285,7 +1285,7 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) hw_write_20kx(hw, PTPALX, ptp_phys_low); hw_write_20kx(hw, PTPAHX, ptp_phys_high); hw_write_20kx(hw, TRNCTL, trnctl); - hw_write_20kx(hw, TRNIS, 0x200c01); /* realy needed? */ + hw_write_20kx(hw, TRNIS, 0x200c01); /* really needed? */ return 0; } diff --git a/sound/pci/emu10k1/memory.c b/sound/pci/emu10k1/memory.c index 957a311..c250614 100644 --- a/sound/pci/emu10k1/memory.c +++ b/sound/pci/emu10k1/memory.c @@ -248,7 +248,7 @@ static int is_valid_page(struct snd_emu10k1 *emu, dma_addr_t addr) /* * map the given memory block on PTB. * if the block is already mapped, update the link order. - * if no empty pages are found, tries to release unsed memory blocks + * if no empty pages are found, tries to release unused memory blocks * and retry the mapping. */ int snd_emu10k1_memblk_map(struct snd_emu10k1 *emu, struct snd_emu10k1_memblk *blk) diff --git a/sound/pci/emu10k1/p16v.c b/sound/pci/emu10k1/p16v.c index 61b8ab3..a81dc44 100644 --- a/sound/pci/emu10k1/p16v.c +++ b/sound/pci/emu10k1/p16v.c @@ -69,7 +69,7 @@ * ADC: Philips 1361T (Stereo 24bit) * DAC: CS4382-K (8-channel, 24bit, 192Khz) * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify diff --git a/sound/pci/emu10k1/p16v.h b/sound/pci/emu10k1/p16v.h index 00f4817..4e0ee1a 100644 --- a/sound/pci/emu10k1/p16v.h +++ b/sound/pci/emu10k1/p16v.h @@ -59,7 +59,7 @@ * ADC: Philips 1361T (Stereo 24bit) * DAC: CS4382-K (8-channel, 24bit, 192Khz) * - * This code was initally based on code from ALSA's emu10k1x.c which is: + * This code was initially based on code from ALSA's emu10k1x.c which is: * Copyright (c) by Francisco Moraes <fmoraes@nc.rr.com> * * This program is free software; you can redistribute it and/or modify @@ -86,7 +86,7 @@ * The sample rate is also controlled by the same registers that control the rate of the EMU10K2 sample rate converters. */ -/* Initally all registers from 0x00 to 0x3f have zero contents. */ +/* Initially all registers from 0x00 to 0x3f have zero contents. */ #define PLAYBACK_LIST_ADDR 0x00 /* Base DMA address of a list of pointers to each period/size */ /* One list entry: 4 bytes for DMA address, * 4 bytes for period_size << 16. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 10f429f..2c40e41 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3730,7 +3730,7 @@ int snd_hda_codec_build_pcms(struct hda_codec *codec) * with the proper parameters for set up. * ops.cleanup should be called in hw_free for clean up of streams. * - * This function returns 0 if successfull, or a negative error code. + * This function returns 0 if successful, or a negative error code. */ int __devinit snd_hda_build_pcms(struct hda_bus *bus) { @@ -4923,7 +4923,7 @@ EXPORT_SYMBOL_HDA(snd_hda_suspend); * * Returns 0 if successful. * - * This fucntion is defined only when POWER_SAVE isn't set. + * This function is defined only when POWER_SAVE isn't set. * In the power-save mode, the codec is resumed dynamically. */ int snd_hda_resume(struct hda_bus *bus) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index faede0f..4695915 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -575,7 +575,7 @@ static int alc_ch_mode_put(struct snd_kcontrol *kcontrol, /* * Control the mode of pin widget settings via the mixer. "pc" is used - * instead of "%" to avoid consequences of accidently treating the % as + * instead of "%" to avoid consequences of accidentally treating the % as * being part of a format specifier. Maximum allowed length of a value is * 63 characters plus NULL terminator. * @@ -5896,6 +5896,7 @@ static void fillup_priv_adc_nids(struct hda_codec *codec, hda_nid_t *nids, static struct snd_pci_quirk beep_white_list[] = { SND_PCI_QUIRK(0x1043, 0x829f, "ASUS", 1), SND_PCI_QUIRK(0x1043, 0x83ce, "EeePC", 1), + SND_PCI_QUIRK(0x1043, 0x831a, "EeePC", 1), SND_PCI_QUIRK(0x8086, 0xd613, "Intel", 1), {} }; @@ -10066,7 +10067,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1028, 0x020d, "Dell Inspiron 530", ALC888_6ST_DELL), - SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavillion", ALC883_6ST_DIG), + SND_PCI_QUIRK(0x103c, 0x2a3d, "HP Pavilion", ALC883_6ST_DIG), SND_PCI_QUIRK(0x103c, 0x2a4f, "HP Samba", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a60, "HP Lucknow", ALC888_3ST_HP), SND_PCI_QUIRK(0x103c, 0x2a61, "HP Nettle", ALC883_6ST_DIG), @@ -10093,6 +10094,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x10f1, 0x2350, "TYAN-S2350", ALC888_6ST_DELL), SND_PCI_QUIRK(0x108e, 0x534d, NULL, ALC883_3ST_6ch), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte P35 DS3R", ALC882_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), @@ -10931,7 +10933,6 @@ enum { PINFIX_LENOVO_Y530, PINFIX_PB_M5210, PINFIX_ACER_ASPIRE_7736, - PINFIX_GIGABYTE_880GM, }; static const struct alc_fixup alc882_fixups[] = { @@ -10963,13 +10964,6 @@ static const struct alc_fixup alc882_fixups[] = { .type = ALC_FIXUP_SKU, .v.sku = ALC_FIXUP_SKU_IGNORE, }, - [PINFIX_GIGABYTE_880GM] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x1114410 }, /* set as speaker */ - { } - } - }, }; static struct snd_pci_quirk alc882_fixup_tbl[] = { @@ -10977,7 +10971,6 @@ static struct snd_pci_quirk alc882_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Y530", PINFIX_LENOVO_Y530), SND_PCI_QUIRK(0x147b, 0x107a, "Abit AW9D-MAX", PINFIX_ABIT_AW9D_MAX), SND_PCI_QUIRK(0x1025, 0x0296, "Acer Aspire 7736z", PINFIX_ACER_ASPIRE_7736), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", PINFIX_GIGABYTE_880GM), {} }; @@ -18489,6 +18482,8 @@ static struct snd_pci_quirk alc662_cfg_tbl[] = { ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB20x", ALC662_AUTO), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), + SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", + ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x152d, 0x2304, "Quanta WH1", ALC663_ASUS_H13), SND_PCI_QUIRK(0x1565, 0x820f, "Biostar TA780G M2+", ALC662_3ST_6ch_DIG), SND_PCI_QUIRK(0x1631, 0xc10c, "PB RS65", ALC663_ASUS_M51VA), @@ -19387,7 +19382,7 @@ enum { ALC662_FIXUP_IDEAPAD, ALC272_FIXUP_MARIO, ALC662_FIXUP_CZC_P10T, - ALC662_FIXUP_GIGABYTE, + ALC662_FIXUP_SKU_IGNORE, }; static const struct alc_fixup alc662_fixups[] = { @@ -19416,20 +19411,17 @@ static const struct alc_fixup alc662_fixups[] = { {} } }, - [ALC662_FIXUP_GIGABYTE] = { - .type = ALC_FIXUP_PINS, - .v.pins = (const struct alc_pincfg[]) { - { 0x14, 0x1114410 }, /* set as speaker */ - { } - } + [ALC662_FIXUP_SKU_IGNORE] = { + .type = ALC_FIXUP_SKU, + .v.sku = ALC_FIXUP_SKU_IGNORE, }, }; static struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1025, 0x0308, "Acer Aspire 8942G", ALC662_FIXUP_ASPIRE), + SND_PCI_QUIRK(0x1025, 0x031c, "Gateway NV79", ALC662_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x1025, 0x038b, "Acer Aspire 8943G", ALC662_FIXUP_ASPIRE), SND_PCI_QUIRK(0x144d, 0xc051, "Samsung R720", ALC662_FIXUP_IDEAPAD), - SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte", ALC662_FIXUP_GIGABYTE), SND_PCI_QUIRK(0x17aa, 0x38af, "Lenovo Ideapad Y550P", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x17aa, 0x3a0d, "Lenovo Ideapad Y550", ALC662_FIXUP_IDEAPAD), SND_PCI_QUIRK(0x1b35, 0x2206, "CZC P10T", ALC662_FIXUP_CZC_P10T), diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 6c5af3e..2b590d9 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -2475,7 +2475,7 @@ static int stac92xx_hp_switch_put(struct snd_kcontrol *kcontrol, spec->hp_switch = ucontrol->value.integer.value[0] ? nid : 0; - /* check to be sure that the ports are upto date with + /* check to be sure that the ports are up to date with * switch changes */ stac_issue_unsol_event(codec, nid); diff --git a/sound/pci/hda/patch_via.c b/sound/pci/hda/patch_via.c index 62fe64c..6a51ffb 100644 --- a/sound/pci/hda/patch_via.c +++ b/sound/pci/hda/patch_via.c @@ -844,14 +844,18 @@ static void notify_aa_path_ctls(struct hda_codec *codec) { int i; struct snd_ctl_elem_id id; - const char *labels[] = {"Mic", "Front Mic", "Line"}; + const char *labels[] = {"Mic", "Front Mic", "Line", "Rear Mic"}; + struct snd_kcontrol *ctl; memset(&id, 0, sizeof(id)); id.iface = SNDRV_CTL_ELEM_IFACE_MIXER; for (i = 0; i < ARRAY_SIZE(labels); i++) { sprintf(id.name, "%s Playback Volume", labels[i]); - snd_ctl_notify(codec->bus->card, SNDRV_CTL_EVENT_MASK_VALUE, - &id); + ctl = snd_hda_find_mixer_ctl(codec, id.name); + if (ctl) + snd_ctl_notify(codec->bus->card, + SNDRV_CTL_EVENT_MASK_VALUE, + &ctl->id); } } diff --git a/sound/pci/ice1712/aureon.c b/sound/pci/ice1712/aureon.c index 2f62522..3e4f8c1 100644 --- a/sound/pci/ice1712/aureon.c +++ b/sound/pci/ice1712/aureon.c @@ -148,7 +148,7 @@ static void aureon_pca9554_write(struct snd_ice1712 *ice, unsigned char reg, udelay(100); /* * send device address, command and value, - * skipping ack cycles inbetween + * skipping ack cycles in between */ for (j = 0; j < 3; j++) { switch (j) { @@ -2143,7 +2143,7 @@ static int __devinit aureon_init(struct snd_ice1712 *ice) ice->num_total_adcs = 2; } - /* to remeber the register values of CS8415 */ + /* to remember the register values of CS8415 */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (!ice->akm) return -ENOMEM; diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 4fc6d8b..f4594d7 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2755,7 +2755,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return err; } if (c->mpu401_1_name) - /* Prefered name available in card_info */ + /* Preferred name available in card_info */ snprintf(ice->rmidi[0]->name, sizeof(ice->rmidi[0]->name), "%s %d", c->mpu401_1_name, card->number); @@ -2772,7 +2772,7 @@ static int __devinit snd_ice1712_probe(struct pci_dev *pci, return err; } if (c->mpu401_2_name) - /* Prefered name available in card_info */ + /* Preferred name available in card_info */ snprintf(ice->rmidi[1]->name, sizeof(ice->rmidi[1]->name), "%s %d", c->mpu401_2_name, diff --git a/sound/pci/ice1712/pontis.c b/sound/pci/ice1712/pontis.c index cdb873f..92c1160 100644 --- a/sound/pci/ice1712/pontis.c +++ b/sound/pci/ice1712/pontis.c @@ -768,7 +768,7 @@ static int __devinit pontis_init(struct snd_ice1712 *ice) ice->num_total_dacs = 2; ice->num_total_adcs = 2; - /* to remeber the register values */ + /* to remember the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ice->akm) return -ENOMEM; diff --git a/sound/pci/ice1712/prodigy_hifi.c b/sound/pci/ice1712/prodigy_hifi.c index 6a9fee3..764cc93 100644 --- a/sound/pci/ice1712/prodigy_hifi.c +++ b/sound/pci/ice1712/prodigy_hifi.c @@ -1046,7 +1046,7 @@ static int __devinit prodigy_hifi_init(struct snd_ice1712 *ice) * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten */ ice->gpio.saved[0] = 0; - /* to remeber the register values */ + /* to remember the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ice->akm) @@ -1128,7 +1128,7 @@ static int __devinit prodigy_hd2_init(struct snd_ice1712 *ice) * don't call snd_ice1712_gpio_get/put(), otherwise it's overwritten */ ice->gpio.saved[0] = 0; - /* to remeber the register values */ + /* to remember the register values */ ice->akm = kzalloc(sizeof(struct snd_akm4xxx), GFP_KERNEL); if (! ice->akm) diff --git a/sound/pci/intel8x0.c b/sound/pci/intel8x0.c index 629a549..6c896db 100644 --- a/sound/pci/intel8x0.c +++ b/sound/pci/intel8x0.c @@ -534,7 +534,7 @@ static int snd_intel8x0_codec_semaphore(struct intel8x0 *chip, unsigned int code udelay(10); } while (time--); - /* access to some forbidden (non existant) ac97 registers will not + /* access to some forbidden (non existent) ac97 registers will not * reset the semaphore. So even if you don't get the semaphore, still * continue the access. We don't need the semaphore anyway. */ snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n", diff --git a/sound/pci/intel8x0m.c b/sound/pci/intel8x0m.c index 2ae8d29..27709f0 100644 --- a/sound/pci/intel8x0m.c +++ b/sound/pci/intel8x0m.c @@ -331,7 +331,7 @@ static int snd_intel8x0m_codec_semaphore(struct intel8x0m *chip, unsigned int co udelay(10); } while (time--); - /* access to some forbidden (non existant) ac97 registers will not + /* access to some forbidden (non existent) ac97 registers will not * reset the semaphore. So even if you don't get the semaphore, still * continue the access. We don't need the semaphore anyway. */ snd_printk(KERN_ERR "codec_semaphore: semaphore is not ready [0x%x][0x%x]\n", diff --git a/sound/pci/mixart/mixart_core.c b/sound/pci/mixart/mixart_core.c index d3350f3..3df0f53 100644 --- a/sound/pci/mixart/mixart_core.c +++ b/sound/pci/mixart/mixart_core.c @@ -265,7 +265,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int if (! timeout) { /* error - no ack */ mutex_unlock(&mgr->msg_mutex); - snd_printk(KERN_ERR "error: no reponse on msg %x\n", msg_frame); + snd_printk(KERN_ERR "error: no response on msg %x\n", msg_frame); return -EIO; } @@ -278,7 +278,7 @@ int snd_mixart_send_msg(struct mixart_mgr *mgr, struct mixart_msg *request, int err = get_msg(mgr, &resp, msg_frame); if( request->message_id != resp.message_id ) - snd_printk(KERN_ERR "REPONSE ERROR!\n"); + snd_printk(KERN_ERR "RESPONSE ERROR!\n"); mutex_unlock(&mgr->msg_mutex); return err; diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c index 833e718..304411c 100644 --- a/sound/pci/pcxhr/pcxhr_core.c +++ b/sound/pci/pcxhr/pcxhr_core.c @@ -1042,11 +1042,11 @@ void pcxhr_msg_tasklet(unsigned long arg) int i, j; if (mgr->src_it_dsp & PCXHR_IRQ_FREQ_CHANGE) - snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occured\n"); + snd_printdd("TASKLET : PCXHR_IRQ_FREQ_CHANGE event occurred\n"); if (mgr->src_it_dsp & PCXHR_IRQ_TIME_CODE) - snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occured\n"); + snd_printdd("TASKLET : PCXHR_IRQ_TIME_CODE event occurred\n"); if (mgr->src_it_dsp & PCXHR_IRQ_NOTIFY) - snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occured\n"); + snd_printdd("TASKLET : PCXHR_IRQ_NOTIFY event occurred\n"); if (mgr->src_it_dsp & (PCXHR_IRQ_FREQ_CHANGE | PCXHR_IRQ_TIME_CODE)) { /* clear events FREQ_CHANGE and TIME_CODE */ pcxhr_init_rmh(prmh, CMD_TEST_IT); @@ -1055,7 +1055,7 @@ void pcxhr_msg_tasklet(unsigned long arg) err, prmh->stat[0]); } if (mgr->src_it_dsp & PCXHR_IRQ_ASYNC) { - snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occured\n"); + snd_printdd("TASKLET : PCXHR_IRQ_ASYNC event occurred\n"); pcxhr_init_rmh(prmh, CMD_ASYNC); prmh->cmd[0] |= 1; /* add SEL_ASYNC_EVENTS */ @@ -1233,7 +1233,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) reg = PCXHR_INPL(mgr, PCXHR_PLX_L2PCIDB); PCXHR_OUTPL(mgr, PCXHR_PLX_L2PCIDB, reg); - /* timer irq occured */ + /* timer irq occurred */ if (reg & PCXHR_IRQ_TIMER) { int timer_toggle = reg & PCXHR_IRQ_TIMER; /* is a 24 bit counter */ @@ -1288,7 +1288,7 @@ irqreturn_t pcxhr_interrupt(int irq, void *dev_id) if (reg & PCXHR_IRQ_MASK) { if (reg & PCXHR_IRQ_ASYNC) { /* as we didn't request any async notifications, - * some kind of xrun error will probably occured + * some kind of xrun error will probably occurred */ /* better resynchronize all streams next interrupt : */ mgr->dsp_time_last = PCXHR_DSP_TIME_INVALID; diff --git a/sound/pci/rme96.c b/sound/pci/rme96.c index d5f5b44..9ff247f 100644 --- a/sound/pci/rme96.c +++ b/sound/pci/rme96.c @@ -150,7 +150,7 @@ MODULE_PARM_DESC(enable, "Enable RME Digi96 soundcard."); #define RME96_RCR_BITPOS_F1 28 #define RME96_RCR_BITPOS_F2 29 -/* Additonal register bits */ +/* Additional register bits */ #define RME96_AR_WSEL (1 << 0) #define RME96_AR_ANALOG (1 << 1) #define RME96_AR_FREQPAD_0 (1 << 2) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index a323eaf..949691a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -391,7 +391,7 @@ MODULE_SUPPORTED_DEVICE("{{RME HDSPM-MADI}}"); /* Status2 Register bits */ /* MADI ONLY */ -#define HDSPM_version0 (1<<0) /* not realy defined but I guess */ +#define HDSPM_version0 (1<<0) /* not really defined but I guess */ #define HDSPM_version1 (1<<1) /* in former cards it was ??? */ #define HDSPM_version2 (1<<2) @@ -936,7 +936,7 @@ struct hdspm { struct snd_kcontrol *playback_mixer_ctls[HDSPM_MAX_CHANNELS]; /* but input to much, so not used */ struct snd_kcontrol *input_mixer_ctls[HDSPM_MAX_CHANNELS]; - /* full mixer accessable over mixer ioctl or hwdep-device */ + /* full mixer accessible over mixer ioctl or hwdep-device */ struct hdspm_mixer *mixer; struct hdspm_tco *tco; /* NULL if no TCO detected */ diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c index 1b8f674..2b5c7a95 100644 --- a/sound/pci/sis7019.c +++ b/sound/pci/sis7019.c @@ -308,7 +308,7 @@ static irqreturn_t sis_interrupt(int irq, void *dev) u32 intr, status; /* We only use the DMA interrupts, and we don't enable any other - * source of interrupts. But, it is possible to see an interupt + * source of interrupts. But, it is possible to see an interrupt * status that didn't actually interrupt us, so eliminate anything * we're not expecting to avoid falsely claiming an IRQ, and an * ensuing endless loop. @@ -773,7 +773,7 @@ static void sis_prepare_timing_voice(struct voice *voice, vperiod = 0; } - /* The interrupt handler implements the timing syncronization, so + /* The interrupt handler implements the timing synchronization, so * setup its state. */ timing->flags |= VOICE_SYNC_TIMING; @@ -1139,7 +1139,7 @@ static int sis_chip_init(struct sis7019 *sis) */ outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); - /* Reset the syncronization groups for all of the channels + /* Reset the synchronization groups for all of the channels * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. * we'll need to change how we handle these. Until then, we just * assign sub-mixer 0 to all playback channels, and avoid any |