August 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Allow SND_SOC_WM8978 to be selected manually
by Geert Uytterhoeven 28 Aug '14

28 Aug '14

When using a DT-based multi-platform kernel, there's not always Kconfig logic that selects the right codec driver. Allow the user to manually select WM8978. This is needed for Armadillo 800 EVA using a generic r8a7740 multi-platform kernel. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7678122f8fe0..8ab154756934 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -731,7 +731,8 @@ config SND_SOC_WM8974 tristate config SND_SOC_WM8978 - tristate + tristate "Wolfson Microelectronics WM8978 codec" + depends on I2C config SND_SOC_WM8983 tristate -- 1.9.1

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Re: [alsa-devel] [question about DPCM] dpcm_be_disconnect race condition
by Mark Brown 28 Aug '14

28 Aug '14

On Wed, Aug 27, 2014 at 04:15:00PM +0800, Qiao Zhou wrote: > Hi Liam, Mark > > I'm developing audio with DPCM on one phone platform, and met one kernel > panic issue related with dpcm_be_connect. Could you please help to suggest? Adding Liam. > Taking audio playback for example: > 1. User space writes all the data into audio buffer. When the blocking write > returns, it closes audio path(which updates dapm and triggers > soc_dpcm_runtime_update). > 2. When user space closes audio path, kernel audio driver(DMA) is still > transferring data from audio buffer to audio peripheral. So DMA interrupt > will come, and the interrupt handler runs on different cpu comparing to > which the task to close audio path runs on. > 3. Due to system schedule, below case may happen. > > cpu0 cpu1 > > soc_dapm_mixer_update_power -> | DMA interrupt -> > soc_dpcm_runtime_update -> | snd_pcm_period_elapsed -> > dpcm_be_disconnect(old) -> | snd_pcm_update_hw_ptr0 -> > __________________________ | xrun -> > | | | snd_pcm_stop -> > |list_del(&dpcm->list_be) | | dpcm_fe_dai_trigger -> > |list_del(&dpcm->list_fe) | | dpcm_be_dai_trigger -> > |kfree(dpcm); | | list_for_each_entry(dpcm.. > |_________________________| | > > On CPU1, list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) > will get dpcm and corresponding be, but there is no protection against the > dpcm_be_disconnect on cpu0. Related dpcm for the BE may be invalid/freed > when cpu1 tries to access it. > Should we require user space to make sure audio transfer ends before closing > audio path to avoid such issue? I'm not sure if I missed anything or used it > wrongly. Well, it should do that in general if it wants to make sure that the audio actually gets played but the kernel should protect against this. We should be holding a lock whenever we update the lists. > Thanks in advance! > -- > > Best Regards > Qiao >

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[alsa-devel] [PATCHv3 0/4] ASoC: Convert to use regmap framework's endianness method.
by Xiubo Li 27 Aug '14

27 Aug '14

Changes in V3: - Delete 'ASoC: fsl-sai: rename big_endian_data to is_msb_first.' patch. Changes in V2: - Modified the regmap config to const type. - Added ASRC patch. - Followed Rutland's advice. Xiubo Li (4): ASoC: fsl-asrc: Convert to use regmap framework's endianness method. ASoC: fsl-esai: Convert to use regmap framework's endianness method. ASoC: fsl-spdif: Convert to use regmap framework's endianness method. ASoC: fsl-sai: Convert to use regmap framework's endianness method. Documentation/devicetree/bindings/sound/fsl-sai.txt | 7 +++---- sound/soc/fsl/fsl_asrc.c | 6 +----- sound/soc/fsl/fsl_esai.c | 5 +---- sound/soc/fsl/fsl_sai.c | 6 +----- sound/soc/fsl/fsl_sai.h | 1 - sound/soc/fsl/fsl_spdif.c | 5 +---- 6 files changed, 7 insertions(+), 23 deletions(-) -- 1.8.5

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Re: [alsa-devel] [PATCH 2/2] ASoC: rt5677: Add DMIC2 clock selection
by Mark Brown 27 Aug '14

27 Aug '14

On Wed, Aug 27, 2014 at 07:50:34PM +0800, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> > > There are two pins can be used for rt5677's DMIC2 clock. This patch > add the select options for it. Applied, thanks.

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Re: [alsa-devel] [PATCH 1/2] ASoC: rt5677: correct mismatch widget name
by Mark Brown 27 Aug '14

27 Aug '14

On Wed, Aug 27, 2014 at 07:50:33PM +0800, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> > > We name MICBIAS1 in dapm widget, but micbias1 in route table. Applied, thanks.

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[alsa-devel] [PATCH] ASoC: fsl-sai: using 'lsb-first' property instead of 'big-endian-data'.
by Xiubo Li 27 Aug '14

27 Aug '14

This property used for configuring whether the LSB or the MSB is transmitted first for the fifo data. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Documentation/devicetree/bindings/sound/fsl-sai.txt | 8 ++++---- sound/soc/fsl/fsl_sai.c | 6 +++--- sound/soc/fsl/fsl_sai.h | 2 +- 3 files changed, 8 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-sai.txt b/Documentation/devicetree/bindings/sound/fsl-sai.txt index 06a405e..4956b14 100644 --- a/Documentation/devicetree/bindings/sound/fsl-sai.txt +++ b/Documentation/devicetree/bindings/sound/fsl-sai.txt @@ -20,9 +20,9 @@ Required properties: See ../pinctrl/pinctrl-bindings.txt for details of the property values. - big-endian: Boolean property, required if all the FTM_PWM registers are big-endian rather than little-endian. -- big-endian-data: If this property is absent, the little endian mode will - be in use as default, or the big endian mode will be in use for all the - fifo data. +- lsb-first: Configures whether the LSB or the MSB is transmitted first for + the fifo data. If this property is absent, the MSB is transmitted first as + default, or the LSB is transmitted first. - fsl,sai-synchronous-rx: This is a boolean property. If present, indicating that SAI will work in the synchronous mode (sync Tx with Rx) which means both the transimitter and receiver will send and receive data by following @@ -53,5 +53,5 @@ sai2: sai@40031000 { dmas = <&edma0 0 VF610_EDMA_MUXID0_SAI2_TX>, <&edma0 0 VF610_EDMA_MUXID0_SAI2_RX>; big-endian; - big-endian-data; + lsb-first; }; diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a6eb784..7eeb1dd 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -175,7 +175,7 @@ static int fsl_sai_set_dai_fmt_tr(struct snd_soc_dai *cpu_dai, bool tx = fsl_dir == FSL_FMT_TRANSMITTER; u32 val_cr2 = 0, val_cr4 = 0; - if (!sai->big_endian_data) + if (!sai->is_lsb_first) val_cr4 |= FSL_SAI_CR4_MF; /* DAI mode */ @@ -304,7 +304,7 @@ static int fsl_sai_hw_params(struct snd_pcm_substream *substream, val_cr5 |= FSL_SAI_CR5_WNW(word_width); val_cr5 |= FSL_SAI_CR5_W0W(word_width); - if (sai->big_endian_data) + if (sai->is_lsb_first) val_cr5 |= FSL_SAI_CR5_FBT(0); else val_cr5 |= FSL_SAI_CR5_FBT(word_width - 1); @@ -573,7 +573,7 @@ static int fsl_sai_probe(struct platform_device *pdev) if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx6sx-sai")) sai->sai_on_imx = true; - sai->big_endian_data = of_property_read_bool(np, "big-endian-data"); + sai->is_lsb_first = of_property_read_bool(np, "lsb-first"); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); base = devm_ioremap_resource(&pdev->dev, res); diff --git a/sound/soc/fsl/fsl_sai.h b/sound/soc/fsl/fsl_sai.h index 2cded44..3466720 100644 --- a/sound/soc/fsl/fsl_sai.h +++ b/sound/soc/fsl/fsl_sai.h @@ -132,7 +132,7 @@ struct fsl_sai { struct clk *bus_clk; struct clk *mclk_clk[FSL_SAI_MCLK_MAX]; - bool big_endian_data; + bool is_lsb_first; bool is_dsp_mode; bool sai_on_imx; bool synchronous[2]; -- 1.8.5

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[alsa-devel] [PATCH] ALSA: hda - Set up initial pins for Acer Aspire V5
by Takashi Iwai 27 Aug '14

27 Aug '14

Acer Aspire V5 doesn't set up the pins correctly at the cold boot while the pins are corrected after the warm reboot. This patch gives the proper pin configs statically in the driver as a workaround. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=81561 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d71270a3f73f..d446ac3137b3 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4408,6 +4408,7 @@ enum { ALC292_FIXUP_TPT440_DOCK, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, + ALC282_FIXUP_ASPIRE_V5_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -4855,6 +4856,22 @@ static const struct hda_fixup alc269_fixups[] = { .chained_before = true, .chain_id = ALC255_FIXUP_DELL1_MIC_NO_PRESENCE }, + [ALC282_FIXUP_ASPIRE_V5_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x12, 0x90a60130 }, + { 0x14, 0x90170110 }, + { 0x17, 0x40000008 }, + { 0x18, 0x411111f0 }, + { 0x19, 0x411111f0 }, + { 0x1a, 0x411111f0 }, + { 0x1b, 0x411111f0 }, + { 0x1d, 0x40f89b2d }, + { 0x1e, 0x411111f0 }, + { 0x21, 0x0321101f }, + { }, + }, + }, }; @@ -4866,6 +4883,7 @@ 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(0x1025, 0x0775, "Acer Aspire E1-572", ALC271_FIXUP_HP_GATE_MIC_JACK_E1_572), + SND_PCI_QUIRK(0x1025, 0x079b, "Acer Aspire V5-573G", ALC282_FIXUP_ASPIRE_V5_PINS), 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), -- 2.1.0

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[alsa-devel] [PATCH 1/1] ice1712: Replacing hex with #defines
by Konstantinos Tsimpoukas 27 Aug '14

27 Aug '14

Adds to te readability of the ice1712 driver. Signed-off-by: Konstantinos Tsimpoukas <kostaslinuxxx(a)gmail.com> --- sound/pci/ice1712/ice1712.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/ice1712/ice1712.c b/sound/pci/ice1712/ice1712.c index 87f7fc4..206ed2c 100644 --- a/sound/pci/ice1712/ice1712.c +++ b/sound/pci/ice1712/ice1712.c @@ -2528,7 +2528,7 @@ static int snd_ice1712_free(struct snd_ice1712 *ice) if (!ice->port) goto __hw_end; /* mask all interrupts */ - outb(0xc0, ICEMT(ice, IRQ)); + outb(ICE1712_MULTI_CAPTURE | ICE1712_MULTI_PLAYBACK, ICEMT(ice, IRQ)); outb(0xff, ICEREG(ice, IRQMASK)); /* --- */ __hw_end: -- 1.9.1

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[alsa-devel] [PATCH 0/2] ASoC/regmap: rt5640: Fix resume
by Jarkko Nikula 27 Aug '14

27 Aug '14

Realtek RT5642 audio codec fails to resume because regmap/regcache tries to sync consecutive registers using block writes but RT5642 doesn't support autoincrementing I2C writes according to debugging and oscilloscope measurments. I'm not sure is there regression after 75a5f89f635c ("regmap: cache: Write consecutive registers in a single block write") or was RT564x resume ever working in mainline so I didn't mark stable yet. Both patches are independent from each other and can be applied into their own subsystems separately. Jarkko Nikula (2): regmap: cache: Fix regcache_sync_block for non-autoincrementing devices ASoC: rt5640: Do not allow regmap to use bulk read-write operations drivers/base/regmap/regcache.c | 14 +++++++++++++- sound/soc/codecs/rt5640.c | 1 + 2 files changed, 14 insertions(+), 1 deletion(-) -- 2.1.0

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[alsa-devel] [PATCH] sound: pci: ctxfi: fix broken user-visible string
by Sudip Mukherjee 26 Aug '14

26 Aug '14

fixed broken user visible strings Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- I am not exactly sure about the patch as some lines after fixing has crossed the limit of 80 characters. But the coding style says : "never break user-visible strings such as printk messages, because that breaks the ability to grep for them" sound/pci/ctxfi/ctatc.c | 18 +++++++----------- sound/pci/ctxfi/cthw20k1.c | 3 +-- sound/pci/ctxfi/cthw20k2.c | 11 +++++------ sound/pci/ctxfi/ctmixer.c | 6 ++---- sound/pci/ctxfi/ctresource.c | 20 ++++++++------------ sound/pci/ctxfi/ctvmem.c | 6 ++---- sound/pci/ctxfi/xfi.c | 6 ++---- 7 files changed, 27 insertions(+), 43 deletions(-) diff --git a/sound/pci/ctxfi/ctatc.c b/sound/pci/ctxfi/ctatc.c index ce9061a..d92a08c 100644 --- a/sound/pci/ctxfi/ctatc.c +++ b/sound/pci/ctxfi/ctatc.c @@ -185,7 +185,7 @@ static unsigned int convert_format(snd_pcm_format_t snd_format) case SNDRV_PCM_FORMAT_FLOAT_LE: return SRC_SF_F32; default: - pr_err("ctxfi: not recognized snd format is %d \n", + pr_err("ctxfi: not recognized snd format is %d\n", snd_format); return SRC_SF_S16; } @@ -1282,8 +1282,7 @@ static int atc_identify_card(struct ct_atc *atc, unsigned int ssid) p = snd_pci_quirk_lookup_id(vendor_id, device_id, list); if (p) { if (p->value < 0) { - pr_err("ctxfi: " - "Device %04x:%04x is black-listed\n", + pr_err("ctxfi: Device %04x:%04x is black-listed\n", vendor_id, device_id); return -ENOENT; } @@ -1315,8 +1314,7 @@ int ct_atc_create_alsa_devs(struct ct_atc *atc) err = alsa_dev_funcs[i].create(atc, i, alsa_dev_funcs[i].public_name); if (err) { - pr_err("ctxfi: " - "Creating alsa device %d failed!\n", i); + pr_err("ctxfi: Creating alsa device %d failed!\n", i); return err; } } @@ -1351,8 +1349,7 @@ static int atc_create_hw_devs(struct ct_atc *atc) err = rsc_mgr_funcs[i].create(atc->hw, &atc->rsc_mgrs[i]); if (err) { - pr_err("ctxfi: " - "Failed to create rsc_mgr %d!!!\n", i); + pr_err("ctxfi: Failed to create rsc_mgr %d!!!\n", i); return err; } } @@ -1399,8 +1396,8 @@ static int atc_get_resources(struct ct_atc *atc) err = daio_mgr->get_daio(daio_mgr, &da_desc, (struct daio **)&atc->daios[i]); if (err) { - pr_err("ctxfi: Failed to get DAIO " - "resource %d!!!\n", i); + pr_err("ctxfi: Failed to get DAIO resource %d!!!\n", + i); return err; } atc->n_daio++; @@ -1603,8 +1600,7 @@ static int atc_resume(struct ct_atc *atc) /* Do hardware resume. */ err = atc_hw_resume(atc); if (err < 0) { - pr_err("ctxfi: pci_enable_device failed, " - "disabling device\n"); + pr_err("ctxfi: pci_enable_device failed, disabling device\n"); snd_card_disconnect(atc->card); return err; } diff --git a/sound/pci/ctxfi/cthw20k1.c b/sound/pci/ctxfi/cthw20k1.c index 782641e..71d496f 100644 --- a/sound/pci/ctxfi/cthw20k1.c +++ b/sound/pci/ctxfi/cthw20k1.c @@ -1911,8 +1911,7 @@ static int hw_card_start(struct hw *hw) /* Set DMA transfer mask */ if (pci_set_dma_mask(pci, CT_XFI_DMA_MASK) < 0 || pci_set_consistent_dma_mask(pci, CT_XFI_DMA_MASK) < 0) { - pr_err("architecture does not support PCI " - "busmaster DMA with mask 0x%llx\n", + pr_err("architecture does not support PCI busmaster DMA with mask 0x%llx\n", CT_XFI_DMA_MASK); err = -ENXIO; goto error1; diff --git a/sound/pci/ctxfi/cthw20k2.c b/sound/pci/ctxfi/cthw20k2.c index 8a72fac..df2d8c5 100644 --- a/sound/pci/ctxfi/cthw20k2.c +++ b/sound/pci/ctxfi/cthw20k2.c @@ -1246,8 +1246,7 @@ static int hw_trn_init(struct hw *hw, const struct trn_conf *info) /* Set up device page table */ if ((~0UL) == info->vm_pgt_phys) { - pr_alert("ctxfi: " - "Wrong device page table page address!!!\n"); + pr_alert("ctxfi: Wrong device page table page address!!!\n"); return -1; } @@ -1890,8 +1889,8 @@ static int hw_adc_init(struct hw *hw, const struct adc_conf *info) hw20k2_i2c_write(hw, MAKE_WM8775_ADDR(WM8775_MMC, 0x0A), MAKE_WM8775_DATA(0x0A)); } else { - pr_alert("ctxfi: Invalid master sampling " - "rate (msr %d)!!!\n", info->msr); + pr_alert("ctxfi: Invalid master sampling rate (msr %d)!!!\n", + info->msr); err = -EINVAL; goto error; } @@ -2034,8 +2033,8 @@ static int hw_card_start(struct hw *hw) /* Set DMA transfer mask */ if (pci_set_dma_mask(pci, CT_XFI_DMA_MASK) < 0 || pci_set_consistent_dma_mask(pci, CT_XFI_DMA_MASK) < 0) { - pr_err("ctxfi: architecture does not support PCI " - "busmaster DMA with mask 0x%llx\n", CT_XFI_DMA_MASK); + pr_err("ctxfi: architecture does not support PCI busmaster DMA with mask 0x%llx\n", + CT_XFI_DMA_MASK); err = -ENXIO; goto error1; } diff --git a/sound/pci/ctxfi/ctmixer.c b/sound/pci/ctxfi/ctmixer.c index 8d986e4..017fa91 100644 --- a/sound/pci/ctxfi/ctmixer.c +++ b/sound/pci/ctxfi/ctmixer.c @@ -854,8 +854,7 @@ static int ct_mixer_get_resources(struct ct_mixer *mixer) for (i = 0; i < (NUM_CT_SUMS * CHN_NUM); i++) { err = sum_mgr->get_sum(sum_mgr, &sum_desc, &sum); if (err) { - pr_err("ctxfi:Failed to get sum resources for " - "front output!\n"); + pr_err("ctxfi:Failed to get sum resources for front output!\n"); break; } mixer->sums[i] = sum; @@ -869,8 +868,7 @@ static int ct_mixer_get_resources(struct ct_mixer *mixer) for (i = 0; i < (NUM_CT_AMIXERS * CHN_NUM); i++) { err = amixer_mgr->get_amixer(amixer_mgr, &am_desc, &amixer); if (err) { - pr_err("ctxfi:Failed to get amixer resources " - "for mixer obj!\n"); + pr_err("ctxfi:Failed to get amixer resources for mixer obj!\n"); break; } mixer->amixers[i] = amixer; diff --git a/sound/pci/ctxfi/ctresource.c b/sound/pci/ctxfi/ctresource.c index 11ac934..e49d2be 100644 --- a/sound/pci/ctxfi/ctresource.c +++ b/sound/pci/ctxfi/ctresource.c @@ -162,14 +162,12 @@ int rsc_init(struct rsc *rsc, u32 idx, enum RSCTYP type, u32 msr, void *hw) case DAIO: break; default: - pr_err( - "ctxfi: Invalid resource type value %d!\n", type); + pr_err("ctxfi: Invalid resource type value %d!\n", type); return -EINVAL; } if (err) { - pr_err( - "ctxfi: Failed to get resource control block!\n"); + pr_err("ctxfi: Failed to get resource control block!\n"); return err; } @@ -192,8 +190,8 @@ int rsc_uninit(struct rsc *rsc) case DAIO: break; default: - pr_err("ctxfi: " - "Invalid resource type value %d!\n", rsc->type); + pr_err("ctxfi: Invalid resource type value %d!\n", + rsc->type); break; } @@ -235,15 +233,13 @@ int rsc_mgr_init(struct rsc_mgr *mgr, enum RSCTYP type, case SUM: break; default: - pr_err( - "ctxfi: Invalid resource type value %d!\n", type); + pr_err("ctxfi: Invalid resource type value %d!\n", type); err = -EINVAL; goto error; } if (err) { - pr_err( - "ctxfi: Failed to get manager control block!\n"); + pr_err("ctxfi: Failed to get manager control block!\n"); goto error; } @@ -286,8 +282,8 @@ int rsc_mgr_uninit(struct rsc_mgr *mgr) case SUM: break; default: - pr_err("ctxfi: " - "Invalid resource type value %d!\n", mgr->type); + pr_err("ctxfi: Invalid resource type value %d!\n", + mgr->type); break; } diff --git a/sound/pci/ctxfi/ctvmem.c b/sound/pci/ctxfi/ctvmem.c index 5ea015b..38163f5 100644 --- a/sound/pci/ctxfi/ctvmem.c +++ b/sound/pci/ctxfi/ctvmem.c @@ -36,8 +36,7 @@ get_vm_block(struct ct_vm *vm, unsigned int size) size = CT_PAGE_ALIGN(size); if (size > vm->size) { - pr_err("ctxfi: Fail! No sufficient device virtual " - "memory space available!\n"); + pr_err("ctxfi: Fail! No sufficient device virtual memory space available!\n"); return NULL; } @@ -132,8 +131,7 @@ ct_vm_map(struct ct_vm *vm, struct snd_pcm_substream *substream, int size) block = get_vm_block(vm, size); if (block == NULL) { - pr_err("ctxfi: No virtual memory block that is big " - "enough to allocate!\n"); + pr_err("ctxfi: No virtual memory block that is big enough to allocate!\n"); return NULL; } diff --git a/sound/pci/ctxfi/xfi.c b/sound/pci/ctxfi/xfi.c index af8c498..35e85ba 100644 --- a/sound/pci/ctxfi/xfi.c +++ b/sound/pci/ctxfi/xfi.c @@ -78,15 +78,13 @@ ct_card_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) if ((reference_rate != 48000) && (reference_rate != 44100)) { pr_err("ctxfi: Invalid reference_rate value %u!!!\n", reference_rate); - pr_err("ctxfi: The valid values for reference_rate " - "are 48000 and 44100, Value 48000 is assumed.\n"); + pr_err("ctxfi: The valid values for reference_rate are 48000 and 44100, Value 48000 is assumed.\n"); reference_rate = 48000; } if ((multiple != 1) && (multiple != 2) && (multiple != 4)) { pr_err("ctxfi: Invalid multiple value %u!!!\n", multiple); - pr_err("ctxfi: The valid values for multiple are " - "1, 2 and 4, Value 2 is assumed.\n"); + pr_err("ctxfi: The valid values for multiple are 1, 2 and 4, Value 2 is assumed.\n"); multiple = 2; } err = ct_atc_create(card, pci, reference_rate, multiple, -- 1.8.1.2

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