January 2017 - Alsa-devel - mailman.alsa-project.org

Re: [alsa-devel] Stream format conversion in SoC driver
by Clemens Ladisch 05 Jan '17

05 Jan '17

Giedrius Trainavičius wrote: >It uses I2S for sending / receiving audio. However, the master clock on our >sound card is generated by the ADC, which supports only 24 bit data, so >getting other sample formats is not possible. The I2S sample format need not be the same as what the DMA controller reads/writes from/to memory. >A question on 'plug' - it accepts 'nearest' as the argument for format, >rate, what happens in case the slave pcm stream supports the exact same >format as requested by user software? The plug plugin instantiates the actual conversion plugins only as needed; with no conversion, it vanishes entirely. Regards, Clemens

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[alsa-devel] [PATCH v6 0/3] Add platform clock for BayTrail platforms
by Irina Tirdea 05 Jan '17

05 Jan '17

These patches specifically enable the audio MCLK required by Baytrail CR devices. It is the remaining part of a bigger set of patches (already merged in Mark's tree) that enable sound for Baytrail CR devices (especially Asus T100TAF) [1]. They include the clock driver and enabling the clock in the pmc_atom code (along with moving of the non-architectural pmc_atom driver code into drivers/platform/x86). [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2016-August/111704.html Changes from v5: - fix build error reported by kbuild test robot - split the clk driver code from x86 platform changes Changes from v4: - move the pmc_atom driver from arch/x86/platform/atom to drivers/platform/x86 Changes from v3: - replace devm_kzalloc with devm_kcalloc - add x86 architecture maintainers Changes from v2: - move clk platform data structures to a separate include file - store clk_hw pointer for the fixed rate clocks Changes from v1: - register the clk device as child of pmc device - pass iomem pointer from pmc driver to clk driver to avoid using pmc_atom_read()/write() and use readl/writel API instead - use devm_clk_hw_register/clkdev_hw_create instead of clk_register/clkdev_create Irina Tirdea (3): clk: x86: Add Atom PMC platform clocks arch/x86/platform/atom: Move pmc_atom to drivers/platform/x86 platform/x86: Enable Atom PMC platform clocks arch/x86/Kconfig | 4 - arch/x86/platform/atom/Makefile | 1 - drivers/acpi/acpi_lpss.c | 2 +- drivers/clk/x86/Makefile | 3 + drivers/clk/x86/clk-byt-plt.c | 380 +++++++++++++++++++++ drivers/platform/x86/Kconfig | 5 + drivers/platform/x86/Makefile | 1 + .../atom => drivers/platform/x86}/pmc_atom.c | 81 ++++- include/linux/platform_data/x86/clk-byt-plt.h | 31 ++ .../linux/platform_data/x86}/pmc_atom.h | 3 + 10 files changed, 500 insertions(+), 11 deletions(-) create mode 100644 drivers/clk/x86/clk-byt-plt.c rename {arch/x86/platform/atom => drivers/platform/x86}/pmc_atom.c (88%) create mode 100644 include/linux/platform_data/x86/clk-byt-plt.h rename {arch/x86/include/asm => include/linux/platform_data/x86}/pmc_atom.h (98%) -- 1.9.1

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[alsa-devel] [PATCH 0/3] ALSA: firewire: fix minor bugs
by Takashi Sakamoto 05 Jan '17

05 Jan '17

Hi, This patchset fixes some minor regressions and mis-programming. These bugs has quite less influences, thus they're not worth for stable maintenance. I'd like to apply them into kernel 4.10 or later. Takashi Sakamoto (3): ALSA: fireworks: fix asymmetric API call at unit removal ALSA: firewire-tascam: Fix to handle error from initialization of stream data ALSA: firewire-lib: change structure member with proper type sound/firewire/amdtp-stream.c | 2 +- sound/firewire/amdtp-stream.h | 4 ++-- sound/firewire/fireworks/fireworks_stream.c | 2 +- sound/firewire/tascam/tascam-stream.c | 2 +- 4 files changed, 5 insertions(+), 5 deletions(-) -- 2.9.3

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[alsa-devel] [PATCH] ALSA: hda - Apply asus-mode8 fixup to ASUS X71SL
by Takashi Iwai 04 Jan '17

04 Jan '17

Although the old quirk table showed ASUS X71SL with ALC663 codec being compatible with asus-mode3 fixup, the bugzilla reporter explained that asus-model8 fits better for the dual headphone controls. So be it. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=191781 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index d43f396450f2..7d660ee1d5e8 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6984,6 +6984,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51), SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51), + SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8), SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16), SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP), SND_PCI_QUIRK(0x1043, 0x8469, "ASUS mobo", ALC662_FIXUP_NO_JACK_DETECT), -- 2.11.0

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[alsa-devel] Better support for Focusrite saffire pro 6 usb 1
by Juho Jokelainen 04 Jan '17

04 Jan '17

After getting said device, I was bummed it didn't work well in linux (or windows for that matter). Fortunately I fould an earlier post about it. http://alsa-devel.alsa-project.narkive.com/pr5aHEeM/rfc-patch-0-2-add-suppo… For whatever reason, that patch has not made it into kernel. I can vouch that this patch (with a few line change to support kernel changes in the last 3 years) does work and my device now supports both audio input and output. Unfortunately I have no experience in kernel development nor do I really understand the alsa system. Since the code has been kept out of kernel, I suspect there is something fishy about it. If in any way possible I would still like to see code that effectively does the same thing as this patch upstreamed. The slightly modified patch (for kernel 4.8): --- sound/usb/pcm.c | 44 ++++++++++++++++++++++++++++++++++++-------- sound/usb/quirks-table.h | 24 ++++++++++++++++++++++++ 2 files changed, 60 insertions(+), 8 deletions(-) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index 44d178e..19e319d 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -483,7 +483,7 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) struct usb_host_interface *alts; struct usb_interface_descriptor *altsd; struct usb_interface *iface; - int err; + int err, need_init, i; iface = usb_ifnum_to_if(dev, fmt->iface); if (WARN_ON(!iface)) @@ -498,12 +498,26 @@ static int set_format(struct snd_usb_substream *subs, struct audioformat *fmt) /* close the old interface */ if (subs->interface >= 0 && subs->interface != fmt->iface) { - err = usb_set_interface(subs->dev, subs->interface, 0); - if (err < 0) { - dev_err(&dev->dev, - "%d:%d: return to setting 0 failed (%d)\n", - fmt->iface, fmt->altsetting, err); - return -EIO; + + need_init = iface->cur_altsetting != alts; + + if (need_init) { + err = usb_set_interface(dev, fmt->iface, fmt->altsetting); + if (err < 0) { + dev_err(&dev->dev, + "%d:%d: usb_set_interface failed (%d)\n", + fmt->iface, fmt->altsetting, err); + return -EIO; + } + dev_dbg(&dev->dev, "setting usb interface %d:%d\n", + fmt->iface, fmt->altsetting); + } + + for (i = 0; i < alts->desc.bNumEndpoints; ++i) { + if (alts->endpoint[i].desc.bEndpointAddress == fmt->endpoint) + alts->endpoint[i].enabled = 1; + else if (need_init) + alts->endpoint[i].enabled = 0; } subs->interface = -1; subs->altset_idx = 0; @@ -1241,12 +1255,26 @@ static int snd_usb_pcm_close(struct snd_pcm_substream *substream, int direction) { struct snd_usb_stream *as = snd_pcm_substream_chip(substream); struct snd_usb_substream *subs = &as->substream[direction]; + struct usb_interface *iface; + struct usb_host_interface *cur_alts; + int i, ep_active = 0; stop_endpoints(subs, true); if (subs->interface >= 0 && !snd_usb_lock_shutdown(subs->stream->chip)) { - usb_set_interface(subs->dev, subs->interface, 0); + iface = usb_ifnum_to_if(subs->dev, subs->interface); + cur_alts = iface->cur_altsetting; + for (i = 0; i < cur_alts->desc.bNumEndpoints; ++i) { + if (cur_alts->endpoint[i].desc.bEndpointAddress == subs->ep_num) + cur_alts->endpoint[i].enabled = 0; + + ep_active += cur_alts->endpoint[i].enabled; + } + + if (!ep_active) + usb_set_interface(subs->dev, subs->interface, 0); + subs->interface = -1; snd_usb_unlock_shutdown(subs->stream->chip); } diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index c60a776..2670498 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2701,6 +2701,30 @@ YAMAHA_DEVICE(0x7010, "UB99"), .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, .endpoint = 0x01, .ep_attr = USB_ENDPOINT_XFER_ISOC, + .maxpacksize = 0x024c, + .rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .rate_min = 44100, + .rate_max = 48000, + .nr_rates = 2, + .rate_table = (unsigned int[]) { + 44100, 48000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3LE, + .channels = 2, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = UAC_EP_CS_ATTR_SAMPLE_RATE, + .endpoint = 0x82, + .ep_attr = USB_ENDPOINT_XFER_ISOC, + .maxpacksize = 0x0126, .rates = SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .rate_min = 44100, -- 2.10.2 my asoundrc for testing purposes: pcm.frontroute { type route slave.pcm "dmix:USB" ttable.0.0 1 ttable.1.1 1 } pcm.frontusb { type plug slave.pcm "frontroute" hint { show on description "USB front channels" } } pcm.rearroute { type route slave.pcm "dmix:USB" ttable.0.2 1 ttable.1.3 1 } pcm.rearusb { type plug slave.pcm "rearroute" hint { show on description "USB rear channels" } } pcm.micaroute { type route slave.pcm "dsnoop:USB" ttable.0.0 1 } pcm.mica { type plug slave.pcm "micaroute" hint { show on description "Focusrite Saffire Input 1" } } pcm.micbroute { type route slave.pcm "dsnoop:USB" ttable.0.1 1 } pcm.micb { type plug slave.pcm "micbroute" hint { show on description "Focusrite Saffire Input 2" } } pcm.!default { type plug slave { pcm "hw:1,0" } } ctl.!default { type hw card 1 } <https://gist.github.com/shiona/def87540b87466fff315c4ac805c6766>

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[alsa-devel] [PATCH] ASoC: rt5640: move DAC2 Power to rt5640_dapm_widgets
by Bard Liao 04 Jan '17

04 Jan '17

"DAC L2 Power" and "DAC R2 Power" are used by both rt5639 and rt5640. But it was defined in rt5640_specific_dapm_widgets[]. Move them to rt5640_dapm_widgets will let both rt5639 and rt5640 can use it. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5640.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index e29a6de..0f1b216 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1227,6 +1227,10 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, RT5640_PWR_DAC_L1_BIT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("DAC R1 Power", RT5640_PWR_DIG1, RT5640_PWR_DAC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC L2 Power", RT5640_PWR_DIG1, + RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("DAC R2 Power", RT5640_PWR_DIG1, + RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), /* SPK/OUT Mixer */ SND_SOC_DAPM_MIXER("SPK MIXL", RT5640_PWR_MIXER, RT5640_PWR_SM_L_BIT, 0, rt5640_spk_l_mix, ARRAY_SIZE(rt5640_spk_l_mix)), @@ -1322,10 +1326,6 @@ static int rt5640_hp_post_event(struct snd_soc_dapm_widget *w, rt5640_mono_mix, ARRAY_SIZE(rt5640_mono_mix)), SND_SOC_DAPM_SUPPLY("Improve MONO Amp Drv", RT5640_PWR_ANLG1, RT5640_PWR_MA_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC L2 Power", RT5640_PWR_DIG1, - RT5640_PWR_DAC_L2_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("DAC R2 Power", RT5640_PWR_DIG1, - RT5640_PWR_DAC_R2_BIT, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("MONOP"), SND_SOC_DAPM_OUTPUT("MONON"), -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] fsl_ssi: set fifo watermark to more reliable value
by Caleb Crome 04 Jan '17

04 Jan '17

From: Caleb Crome <caleb(a)crome.org> The fsl_ssi fifo watermark is by default set to 2 free spaces (i.e. activate DMA on FIFO when only 2 spaces are left.) This means the DMA must service the fifo within 2 audio samples, which is just not enough time for many use cases with high data rate. In many configurations the audio channel slips (causing l/r swap in stereo configurations, or channel slipping in multi-channel configurations). This patch gives more breathing room and allows the SSI to operate reliably by changing the fifio refill watermark to 8. There is no change in behavior for older chips (with an 8-deep fifo). Only the newer chips with a 15-deep fifo get the new behavior. I suspect a new fifo depth setting could be optimized on the older chips too, but I have not tested. Signed-off-by: Caleb Crome <caleb(a)crome.org> --- sound/soc/fsl/fsl_ssi.c | 74 +++++++++++++++++++++++++++++++++++-------------- 1 file changed, 53 insertions(+), 21 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index bedec4a..56245e6 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -224,6 +224,12 @@ struct fsl_ssi_soc_data { * @dbg_stats: Debugging statistics * * @soc: SoC specific data + * + * @fifo_watermark: the FIFO watermark setting. Notifies DMA when + * there are @fifo_watermark or fewer words in TX fifo or + * @fifo_watermark or more empty words in RX fifo. + * @dma_maxburst: max number of words to transfer in one go. So far, + * this is always the same as fifo_watermark. */ struct fsl_ssi_private { struct regmap *regs; @@ -263,6 +269,9 @@ struct fsl_ssi_private { const struct fsl_ssi_soc_data *soc; struct device *dev; + + u32 fifo_watermark; + u32 dma_maxburst; }; /* @@ -1051,21 +1060,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, regmap_write(regs, CCSR_SSI_SRCR, srcr); regmap_write(regs, CCSR_SSI_SCR, scr); - /* - * Set the watermark for transmit FIFI 0 and receive FIFO 0. We don't - * use FIFO 1. We program the transmit water to signal a DMA transfer - * if there are only two (or fewer) elements left in the FIFO. Two - * elements equals one frame (left channel, right channel). This value, - * however, depends on the depth of the transmit buffer. - * - * We set the watermark on the same level as the DMA burstsize. For - * fiq it is probably better to use the biggest possible watermark - * size. - */ - if (ssi_private->use_dma) - wm = ssi_private->fifo_depth - 2; - else - wm = ssi_private->fifo_depth; + wm = ssi_private->fifo_watermark; regmap_write(regs, CCSR_SSI_SFCSR, CCSR_SSI_SFCSR_TFWM0(wm) | CCSR_SSI_SFCSR_RFWM0(wm) | @@ -1373,12 +1368,8 @@ static int fsl_ssi_imx_probe(struct platform_device *pdev, dev_dbg(&pdev->dev, "could not get baud clock: %ld\n", PTR_ERR(ssi_private->baudclk)); - /* - * We have burstsize be "fifo_depth - 2" to match the SSI - * watermark setting in fsl_ssi_startup(). - */ - ssi_private->dma_params_tx.maxburst = ssi_private->fifo_depth - 2; - ssi_private->dma_params_rx.maxburst = ssi_private->fifo_depth - 2; + ssi_private->dma_params_tx.maxburst = ssi_private->dma_maxburst; + ssi_private->dma_params_rx.maxburst = ssi_private->dma_maxburst; ssi_private->dma_params_tx.addr = ssi_private->ssi_phys + CCSR_SSI_STX0; ssi_private->dma_params_rx.addr = ssi_private->ssi_phys + CCSR_SSI_SRX0; @@ -1543,6 +1534,47 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + /* + * Set the watermark for transmit FIFO 0 and receive FIFO 0. We don't + * use FIFO 1 but set the watermark appropriately nontheless. + * We program the transmit water to signal a DMA transfer + * if there are N elements left in the FIFO. For chips with 15-deep + * FIFOs, set watermark to 8. This allows the SSI to operate at a + * high data rate without channel slipping. Behavior is unchanged + * for the older chips with a fifo depth of only 8. A value of 4 + * might be appropriate for the older chips, but is left at + * fifo_depth-2 until sombody has a chance to test. + * + * We set the watermark on the same level as the DMA burstsize. For + * fiq it is probably better to use the biggest possible watermark + * size. + */ + switch (ssi_private->fifo_depth) { + case 15: + /* + * 2 samples is not enough when running at high data + * rates (like 48kHz @ 16 bits/channel, 16 channels) + * 8 seems to split things evenly and leave enough time + * for the DMA to fill the FIFO before it's over/under + * run. + */ + ssi_private->fifo_watermark = 8; + ssi_private->dma_maxburst = 8; + break; + case 8: + default: + /* + * maintain old behavior for older chips. + * Keeping it the same because I don't have an older + * board to test with. + * I suspect this could be changed to be something to + * leave some more space in the fifo. + */ + ssi_private->fifo_watermark = ssi_private->fifo_depth - 2; + ssi_private->dma_maxburst = ssi_private->fifo_depth - 2; + break; + } + dev_set_drvdata(&pdev->dev, ssi_private); if (ssi_private->soc->imx) { -- 2.7.4

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[alsa-devel] [PATCH 1/2] ALSA: vx: Fix possible transfer overflow
by Takashi Iwai 04 Jan '17

04 Jan '17

The pseudo DMA transfer codes in VX222 and VX-pocket driver have a slight bug where they check the buffer boundary wrongly, and may overflow. Also, the zero sample count might be handled badly for the playback (although it shouldn't happen in theory). This patch addresses these issues. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=141541 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/vx/vx_pcm.c | 6 ++++-- sound/pci/vx222/vx222_ops.c | 12 ++++++------ sound/pcmcia/vx/vxp_ops.c | 12 ++++++------ 3 files changed, 16 insertions(+), 14 deletions(-) diff --git a/sound/drivers/vx/vx_pcm.c b/sound/drivers/vx/vx_pcm.c index 11467272089e..41a617886cc7 100644 --- a/sound/drivers/vx/vx_pcm.c +++ b/sound/drivers/vx/vx_pcm.c @@ -1048,8 +1048,10 @@ static void vx_pcm_capture_update(struct vx_core *chip, struct snd_pcm_substream /* ok, let's accelerate! */ int align = pipe->align * 3; space = (count / align) * align; - vx_pseudo_dma_read(chip, runtime, pipe, space); - count -= space; + if (space > 0) { + vx_pseudo_dma_read(chip, runtime, pipe, space); + count -= space; + } } /* read the rest of bytes */ while (count > 0) { diff --git a/sound/pci/vx222/vx222_ops.c b/sound/pci/vx222/vx222_ops.c index af83b3b38052..8e457ea27f89 100644 --- a/sound/pci/vx222/vx222_ops.c +++ b/sound/pci/vx222/vx222_ops.c @@ -269,12 +269,12 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, /* Transfer using pseudo-dma. */ - if (offset + count > pipe->buffer_bytes) { + if (offset + count >= pipe->buffer_bytes) { int length = pipe->buffer_bytes - offset; count -= length; length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ - while (length-- > 0) { + for (; length > 0; length--) { outl(cpu_to_le32(*addr), port); addr++; } @@ -284,7 +284,7 @@ static void vx2_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, pipe->hw_ptr += count; count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ - while (count-- > 0) { + for (; count > 0; count--) { outl(cpu_to_le32(*addr), port); addr++; } @@ -307,12 +307,12 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, vx2_setup_pseudo_dma(chip, 0); /* Transfer using pseudo-dma. */ - if (offset + count > pipe->buffer_bytes) { + if (offset + count >= pipe->buffer_bytes) { int length = pipe->buffer_bytes - offset; count -= length; length >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ - while (length-- > 0) + for (; length > 0; length--) *addr++ = le32_to_cpu(inl(port)); addr = (u32 *)runtime->dma_area; pipe->hw_ptr = 0; @@ -320,7 +320,7 @@ static void vx2_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, pipe->hw_ptr += count; count >>= 2; /* in 32bit words */ /* Transfer using pseudo-dma. */ - while (count-- > 0) + for (; count > 0; count--) *addr++ = le32_to_cpu(inl(port)); vx2_release_pseudo_dma(chip); diff --git a/sound/pcmcia/vx/vxp_ops.c b/sound/pcmcia/vx/vxp_ops.c index 281972913c32..56aa1ba73ccc 100644 --- a/sound/pcmcia/vx/vxp_ops.c +++ b/sound/pcmcia/vx/vxp_ops.c @@ -369,12 +369,12 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, unsigned short *addr = (unsigned short *)(runtime->dma_area + offset); vx_setup_pseudo_dma(chip, 1); - if (offset + count > pipe->buffer_bytes) { + if (offset + count >= pipe->buffer_bytes) { int length = pipe->buffer_bytes - offset; count -= length; length >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ - while (length-- > 0) { + for (; length > 0; length--) { outw(cpu_to_le16(*addr), port); addr++; } @@ -384,7 +384,7 @@ static void vxp_dma_write(struct vx_core *chip, struct snd_pcm_runtime *runtime, pipe->hw_ptr += count; count >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ - while (count-- > 0) { + for (; count > 0; count--) { outw(cpu_to_le16(*addr), port); addr++; } @@ -411,12 +411,12 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, if (snd_BUG_ON(count % 2)) return; vx_setup_pseudo_dma(chip, 0); - if (offset + count > pipe->buffer_bytes) { + if (offset + count >= pipe->buffer_bytes) { int length = pipe->buffer_bytes - offset; count -= length; length >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ - while (length-- > 0) + for (; length > 0; length--) *addr++ = le16_to_cpu(inw(port)); addr = (unsigned short *)runtime->dma_area; pipe->hw_ptr = 0; @@ -424,7 +424,7 @@ static void vxp_dma_read(struct vx_core *chip, struct snd_pcm_runtime *runtime, pipe->hw_ptr += count; count >>= 1; /* in 16bit words */ /* Transfer using pseudo-dma. */ - while (count-- > 1) + for (; count > 1; count--) *addr++ = le16_to_cpu(inw(port)); /* Disable DMA */ pchip->regDIALOG &= ~VXP_DLG_DMAREAD_SEL_MASK; -- 2.11.0

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[alsa-devel] [PATCH] ALSA: hda - Fix up GPIO for ASUS ROG Ranger
by Takashi Iwai 04 Jan '17

04 Jan '17

ASUS ROG Ranger VIII with ALC1150 codec requires the extra GPIO pin to up for the front panel. Just use the existing fixup for setting up the GPIO pins. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=189411 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9448daff9d8b..d43f396450f2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -2230,6 +2230,7 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = { 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(0x1043, 0x8691, "ASUS ROG Ranger VIII", ALC882_FIXUP_GPIO3), 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), -- 2.11.0

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[alsa-devel] ASoC: fsl_sai: Independent TX and RX master slave configuration
by Vishal Thanki 04 Jan '17

04 Jan '17

Hi All, I am working on a custom board based on i.MX7d SoC from NXP. The hardware uses SAI (Synchronous Audio Interface) Port 1 of i.MX7d for playback and recording use cases. The required system configuration is: SoC should act as I2S master for TX part, and slave for RX part. However, I could not figure out a clean way to instruct CPU dai driver (sound/soc/fsl/fsl_sai.c) about different roles for TX and RX of a same DAI. The .set_fmt DAI op configures the TX and RX registers in the same manner. In order to achieve the required configuration (TX master, RX slave), I added a device tree property (i.e. "fsl,sai-is-rx-slave") and based on that make necessary changes in .set_fmt DAI operation. Attached is the patch for the same. Is this the right way to go? Any suggestions would be helpful. Thanks in advance, Vishal

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