January 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] arecord: strange --duration behaviour with some sample rates
by Daniel Drake 20 Jan '15

20 Jan '15

Hi, With alsa-utils-1.0.28, we're seeing some strange behaviour when running: arecord -t wav -r 44100 -d 1 out.wav After recording 1 second of data to out.wav, a load of extra files are created like out-01.wav, out-02.wav, out-03.wav, etc. These extra files are 44 bytes in length. arecord seems to continue like this forever rather than exiting after the 1 second recording. The problem is somehow linked to the sample-rate. At 48000 or 8000, things work fine, just 44100 is bad. Tracing some values in the source: count is 44100, and rest is initialized to the same. In each iteration of the /* capture */ loop, f has value 5512, i.e. 5512 bytes are read. So rest decreases through values 38588, 33076, 27564, 22052, 16540, 11028, 5516. Now we read another 5512 bytes, rest is decreased to 4. So we now call pcm_read(audiobuf, f) where f has value 4. pcm_read adjusts that to 5512 and returns 5512. Now we hit: if (pcm_read(audiobuf, f) != f) break; We break out of that inner loop, count still has value 4 so we reach the end of the outer loop, and loop again, which causes a new file to be opened, etc. Sorry that I'm not quite up-to-scratch with pcm basics to suggest a quick fix, but hopefully this diagnosis is useful at least. Thanks, Daniel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] ALSA: hda - Enable docking station for an HP machine
by David Henningsson 20 Jan '15

20 Jan '15

From: TienFu Chen <tienfu.chen(a)canonical.com> On this machine we need to inform the driver where the docking station pins are, because it has not been set up by BIOS. Tested-by: TienFu Chen <tienfu.chen(a)canonical.com> BugLink: https://bugs.launchpad.net/bugs/1412800 Signed-off-by: TienFu Chen <tienfu.chen(a)canonical.com> Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1720f8a..0403061 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4417,6 +4417,7 @@ enum { ALC280_FIXUP_HP_GPIO4, ALC286_FIXUP_HP_GPIO_LED, ALC280_FIXUP_HP_GPIO2_MIC_HOTKEY, + ALC280_FIXUP_HP_DOCK_PINS, }; static const struct hda_fixup alc269_fixups[] = { @@ -4894,6 +4895,17 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc280_fixup_hp_gpio2_mic_hotkey, }, + [ALC280_FIXUP_HP_DOCK_PINS] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x1b, 0x21011020 }, /* line-out */ + { 0x1a, 0x01a1903c }, /* headset mic */ + { 0x18, 0x2181103f }, /* line-in */ + { }, + }, + .chained = true, + .chain_id = ALC280_FIXUP_HP_GPIO4 + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -4937,6 +4949,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x226b, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x226e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x2271, "HP", ALC286_FIXUP_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x2272, "HP", ALC280_FIXUP_HP_DOCK_PINS), SND_PCI_QUIRK(0x103c, 0x229e, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b2, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), SND_PCI_QUIRK(0x103c, 0x22b7, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC1), -- 1.9.1

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[alsa-devel] [PATCH] fm801: move to pcim_* and devm_* functions
by Andy Shevchenko 20 Jan '15

20 Jan '15

The managed functions allow to get ->probe() and ->remove() simplier. This patch converts code to use managed functions. While here remove the dead code and fix the value printed in error message. Signed-off-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> --- sound/pci/fm801.c | 29 +++++++---------------------- 1 file changed, 7 insertions(+), 22 deletions(-) diff --git a/sound/pci/fm801.c b/sound/pci/fm801.c index dcda3c1..2708500 100644 --- a/sound/pci/fm801.c +++ b/sound/pci/fm801.c @@ -1178,12 +1178,8 @@ static int snd_fm801_free(struct fm801 *chip) v4l2_device_unregister(&chip->v4l2_dev); } #endif - if (chip->irq >= 0) - free_irq(chip->irq, chip); pci_release_regions(chip->pci); - pci_disable_device(chip->pci); - kfree(chip); return 0; } @@ -1206,28 +1202,23 @@ static int snd_fm801_create(struct snd_card *card, }; *rchip = NULL; - if ((err = pci_enable_device(pci)) < 0) + if ((err = pcim_enable_device(pci)) < 0) return err; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) { - pci_disable_device(pci); + chip = devm_kzalloc(&pci->dev, sizeof(*chip), GFP_KERNEL); + if (chip == NULL) return -ENOMEM; - } spin_lock_init(&chip->reg_lock); chip->card = card; chip->pci = pci; chip->irq = -1; chip->tea575x_tuner = tea575x_tuner; - if ((err = pci_request_regions(pci, "FM801")) < 0) { - kfree(chip); - pci_disable_device(pci); + if ((err = pci_request_regions(pci, "FM801")) < 0) return err; - } chip->port = pci_resource_start(pci, 0); if ((tea575x_tuner & TUNER_ONLY) == 0) { - if (request_irq(pci->irq, snd_fm801_interrupt, IRQF_SHARED, - KBUILD_MODNAME, chip)) { - dev_err(card->dev, "unable to grab IRQ %d\n", chip->irq); + if (devm_request_irq(&pci->dev, pci->irq, snd_fm801_interrupt, + IRQF_SHARED, KBUILD_MODNAME, chip)) { + dev_err(card->dev, "unable to grab IRQ %d\n", pci->irq); snd_fm801_free(chip); return -EBUSY; } @@ -1242,12 +1233,6 @@ static int snd_fm801_create(struct snd_card *card, /* init might set tuner access method */ tea575x_tuner = chip->tea575x_tuner; - if (chip->irq >= 0 && (tea575x_tuner & TUNER_ONLY)) { - pci_clear_master(pci); - free_irq(chip->irq, chip); - chip->irq = -1; - } - if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops)) < 0) { snd_fm801_free(chip); return err; -- 1.8.3.101.g727a46b

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[alsa-devel] [RESEND PATCH] mfd: wm8994: set mfd id-base for regulator devs creation to avoid conflicts
by Inha Song 20 Jan '15

20 Jan '15

After commit: 6e3f62f0793e ("mfd: core: Fix platform-device id generation") We must set the id base when register a duplicate name of mfd_cell. If not, duplicate filename error was reported. - sysfs: cannot create duplicate filename '/devices/.../wm8994-ldo' Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/wm8994-core.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/drivers/mfd/wm8994-core.c b/drivers/mfd/wm8994-core.c index e6fab94..47cd3cc 100644 --- a/drivers/mfd/wm8994-core.c +++ b/drivers/mfd/wm8994-core.c @@ -36,12 +36,12 @@ static const struct mfd_cell wm8994_regulator_devs[] = { { .name = "wm8994-ldo", - .id = 1, + .id = 0, .pm_runtime_no_callbacks = true, }, { .name = "wm8994-ldo", - .id = 2, + .id = 1, .pm_runtime_no_callbacks = true, }, }; @@ -344,7 +344,7 @@ static int wm8994_device_init(struct wm8994 *wm8994, int irq) dev_set_drvdata(wm8994->dev, wm8994); /* Add the on-chip regulators first for bootstrapping */ - ret = mfd_add_devices(wm8994->dev, -1, + ret = mfd_add_devices(wm8994->dev, 0, wm8994_regulator_devs, ARRAY_SIZE(wm8994_regulator_devs), NULL, 0, NULL); -- 2.0.0.390.gcb682f8

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[alsa-devel] [PATCH 0/7] More line6 fixes / rewrites
by Takashi Iwai 20 Jan '15

20 Jan '15

Hi, this is yet another series of patches for line6 drivers. Most of patches are cleanups and trivial fixes while the last one is to reimplement the LED control via the standard LED class. As usual, patches are found in test/line6 branch for now. Takashi

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[alsa-devel] [PATCH 0/3] Minor line6 cleanup
by Chris Rorvick 20 Jan '15

20 Jan '15

These apply on top of test/line6. Chris Rorvick (3): ALSA: line6: Remove superfluous NULL checks ALSA: line6: Refer to manufacturer as "Line 6" ALSA: line6: Remove driver version from header comment sound/usb/line6/Kconfig | 4 ++-- sound/usb/line6/capture.c | 2 +- sound/usb/line6/capture.h | 2 +- sound/usb/line6/driver.c | 36 +++++++++++++----------------------- sound/usb/line6/driver.h | 14 +++++++------- sound/usb/line6/midi.c | 6 +++--- sound/usb/line6/midi.h | 4 ++-- sound/usb/line6/midibuf.c | 4 ++-- sound/usb/line6/midibuf.h | 2 +- sound/usb/line6/pcm.c | 2 +- sound/usb/line6/pcm.h | 10 +++++----- sound/usb/line6/playback.c | 2 +- sound/usb/line6/playback.h | 2 +- sound/usb/line6/pod.c | 6 +++--- sound/usb/line6/podhd.c | 6 +++--- sound/usb/line6/toneport.c | 4 ++-- sound/usb/line6/usbdefs.h | 2 +- sound/usb/line6/variax.c | 4 ++-- 18 files changed, 51 insertions(+), 61 deletions(-) -- 2.1.0

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[alsa-devel] [PATCH v2] ASoC: OMAP: mcbsp: ensure that CLKX and CLKR are not used as ouput pins when they are used as input clock for the SRG.
by Thomas Niederprüm 20 Jan '15

20 Jan '15

This patch fixes faulty behaviour in a setup where the input clock for the SRG is fed through the CLKR/CLKX pin but the McBSP is configured to be master (SND_SOC_DAIFMT_CBS_CFS). In that case of course CLKR/CLKX must not be configured as output pin. Otherwise the input clock is messed up horribly. This patch makes it possible to use the CLKR/CLKX pin rather than CLKS to inject a reference clock in setups where McBSP is master and not both rx and tx are used. However for this to work it has to be ensured that set_dai_sysclk() is called after set_dai_fmt(). This was tested on a beagleboard-xm using McBSP1 to drive a i2s DAC through the tx lines (CLKX,FSX,DX). Using this patch the CLKR pin is used to inject an external reference clock. changes since v1: - added comments explaining the bit masking to disable output on CLKR/CLKX Signed-off-by: Thomas Niederprüm <niederp(a)physik.uni-kl.de> --- sound/soc/omap/omap-mcbsp.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index bd3ef2a..c37f606 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -530,8 +530,19 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai, case OMAP_MCBSP_SYSCLK_CLKX_EXT: regs->srgr2 |= CLKSM; + regs->pcr0 |= SCLKME; + /* + * If McBSP is master but yet the CLKX/CLKR pin drives the SRG, + * disable output on those pins. This enables to inject the + * reference clock through CLKX/CLKR. For this to work + * set_dai_sysclk() _needs_ to be called after set_dai_fmt(). + */ + regs->pcr0 &= ~CLKXM; + break; case OMAP_MCBSP_SYSCLK_CLKR_EXT: regs->pcr0 |= SCLKME; + /* Disable ouput on CLKR pin in master mode */ + regs->pcr0 &= ~CLKRM; break; default: err = -ENODEV; -- 2.1.1

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[alsa-devel] [PATCH 00/17] More line6 driver code reorg and cleanups
by Takashi Iwai 20 Jan '15

20 Jan '15

Hi, here is a series of patches I hacked quickly after merging line6 stuff into sound git tree. There are two significant changes: - Split to each driver and renames: now there are 4 card modules, snd-usb-pod, snd-usb-podhd, snd-usb-toneport and snd-usb-variax, while snd-usb-line6 serves as a common helper module. - Many codes restrucruring, especially cleanup of the resource management; the destructor is solely called from card's private_free now The current merged code is found in topic/line6 branch of sound.git tree while this development code is found in test/line6 branch. The devel branch hasn't been merged to linux-next yet, I'm waiting for any good/bad test results. So, give me any feedback. Thanks! (BTW, the next step would be to straighten the init code in pod.c, etc instead of (ab)using the system timer. Instead, it should be a flat function with wait_event() of such.) Takashi

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[alsa-devel] [PATCH 1/2] ASoC: atmel_ssc_dai: fix start event for I2S mode
by Bo Shen 20 Jan '15

20 Jan '15

According to the I2S specification information as following: - WS = 0, channel 1 (left) - WS = 1, channel 2 (right) So, the start event should be TF/RF falling edge. Reported-by: Songjun Wu <songjun.wu(a)atmel.com> Signed-off-by: Bo Shen <voice.shen(a)atmel.com> --- sound/soc/atmel/atmel_ssc_dai.c | 18 ++++-------------- 1 file changed, 4 insertions(+), 14 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 99ff35e..e691aab 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -348,7 +348,6 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, struct atmel_pcm_dma_params *dma_params; int dir, channels, bits; u32 tfmr, rfmr, tcmr, rcmr; - int start_event; int ret; int fslen, fslen_ext; @@ -457,19 +456,10 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, * The SSC transmit clock is obtained from the BCLK signal on * on the TK line, and the SSC receive clock is * generated from the transmit clock. - * - * For single channel data, one sample is transferred - * on the falling edge of the LRC clock. - * For two channel data, one sample is - * transferred on both edges of the LRC clock. */ - start_event = ((channels == 1) - ? SSC_START_FALLING_RF - : SSC_START_EDGE_RF); - rcmr = SSC_BF(RCMR_PERIOD, 0) | SSC_BF(RCMR_STTDLY, START_DELAY) - | SSC_BF(RCMR_START, start_event) + | SSC_BF(RCMR_START, SSC_START_FALLING_RF) | SSC_BF(RCMR_CKI, SSC_CKI_RISING) | SSC_BF(RCMR_CKO, SSC_CKO_NONE) | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? @@ -478,14 +468,14 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, rfmr = SSC_BF(RFMR_FSEDGE, SSC_FSEDGE_POSITIVE) | SSC_BF(RFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(RFMR_FSLEN, 0) - | SSC_BF(RFMR_DATNB, 0) + | SSC_BF(RFMR_DATNB, (channels - 1)) | SSC_BIT(RFMR_MSBF) | SSC_BF(RFMR_LOOP, 0) | SSC_BF(RFMR_DATLEN, (bits - 1)); tcmr = SSC_BF(TCMR_PERIOD, 0) | SSC_BF(TCMR_STTDLY, START_DELAY) - | SSC_BF(TCMR_START, start_event) + | SSC_BF(TCMR_START, SSC_START_FALLING_RF) | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) | SSC_BF(TCMR_CKO, SSC_CKO_NONE) | SSC_BF(TCMR_CKS, ssc->clk_from_rk_pin ? @@ -495,7 +485,7 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, | SSC_BF(TFMR_FSDEN, 0) | SSC_BF(TFMR_FSOS, SSC_FSOS_NONE) | SSC_BF(TFMR_FSLEN, 0) - | SSC_BF(TFMR_DATNB, 0) + | SSC_BF(TFMR_DATNB, (channels - 1)) | SSC_BIT(TFMR_MSBF) | SSC_BF(TFMR_DATDEF, 0) | SSC_BF(TFMR_DATLEN, (bits - 1)); -- 2.3.0.rc0

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[alsa-devel] [PATCH 1/3] ARM: shmobile: armadillo800eva: Properly specify HDMI audio link format
by Lars-Peter Clausen 20 Jan '15

20 Jan '15

The DAI link format should be specified for the whole link rather than just one component on the link. So move the format specification for the HDMI audio link from the CPU component to the link itself. Since the sh-mobile-hdmi DAI driver doesn't implement the set_fmt() callback in this case there is no functional difference between only specifying the the format for the CPU side or for the whole link, but the later it will allow us to remove support for just specifying the format for one component. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- arch/arm/mach-shmobile/board-armadillo800eva.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/arch/arm/mach-shmobile/board-armadillo800eva.c b/arch/arm/mach-shmobile/board-armadillo800eva.c index 6d949f1..75de26c 100644 --- a/arch/arm/mach-shmobile/board-armadillo800eva.c +++ b/arch/arm/mach-shmobile/board-armadillo800eva.c @@ -1040,9 +1040,9 @@ static struct asoc_simple_card_info fsi2_hdmi_info = { .card = "FSI2B-HDMI", .codec = "sh-mobile-hdmi", .platform = "sh_fsi2", + .daifmt = SND_SOC_DAIFMT_CBS_CFS, .cpu_dai = { .name = "fsib-dai", - .fmt = SND_SOC_DAIFMT_CBS_CFS, }, .codec_dai = { .name = "sh_mobile_hdmi-hifi", -- 1.8.0

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