April 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v7 0/6] drm/bridge: sii902x: HDMI-audio support and some fixes
by Jyri Sarha 29 Apr '19

29 Apr '19

The already reviewed patches are in front. Not reviewed still patches 5-6. Changes since v6: - "dt-bindings: display: sii902x: Add HDMI audio bindings" - Replace references to ASoC (a Linux term) with more generic terms - Add "sil,"-prefix to "i2s-data-lanes"-property in the example node - "drm/bridge: sii902x: Implement HDMI audio support" - Do not use unsigned in sii902x_select_mclk_div()'s "distance" calculations to avoid bad type promotions to unsigned causing unexpected results - Always return the clock divisor by sii902x_select_mclk_div() - Disable regmap locking since the driver specific locking should be enough - Add missing mutex locking to the register access in the irq handler The previous round: https://patchwork.kernel.org/cover/10881169/ Jyri Sarha (5): drm/bridge: sii902x: Set output mode to HDMI or DVI according to EDID drm/bridge: sii902x: pixel clock unit is 10kHz instead of 1kHz dt-bindings: display: sii902x: Remove trailing white space dt-bindings: display: sii902x: Add HDMI audio bindings drm/bridge: sii902x: Implement HDMI audio support Tomi Valkeinen (1): drm/bridge: sii902x: add input_bus_flags .../bindings/display/bridge/sii902x.txt | 43 +- drivers/gpu/drm/bridge/sii902x.c | 488 +++++++++++++++++- 2 files changed, 523 insertions(+), 8 deletions(-) -- Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: Add support for high power analog output
by Saravanan Sekar 29 Apr '19

29 Apr '19

Add support to power and output level control for the analog high power output drivers HPOUT and HPCOM. Signed-off-by: Saravanan Sekar <sravanhome(a)gmail.com> --- sound/soc/codecs/tlv320aic3x.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 516d17cb2182..d4bafac802eb 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -419,6 +419,12 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Pop reduction */ SOC_ENUM("Output Driver Power-On time", aic3x_poweron_time_enum), SOC_ENUM("Output Driver Ramp-up step", aic3x_rampup_step_enum), + + /* Analog HPOUT, HPCOM power and output level controls */ + SOC_DOUBLE_R("Analog output power control", HPROUT_CTRL, + HPRCOM_CTRL, 0, 1, 0), + SOC_DOUBLE_R("Analog output level control", HPROUT_CTRL, + HPRCOM_CTRL, 4, 9, 0), }; /* For other than tlv320aic3104 */ -- 2.17.1

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[alsa-devel] [PATCH 0/2] ASoC: fix hw_params/hw_free validity checks
by Jerome Brunet 29 Apr '19

29 Apr '19

While working on a codec, I have found an odd situation. The hw_params callback which was provided by the codec dai was not called as expected. However, the hw_free callback was called, even if hw_params was skipped. After debugging, I found out that the dai was considered invalid because it did not specify any rates, only rate_min and rate_max. That's a first problem. If a codec dai is considered invalid, hw_free should probably not be called either, that's a 2nd issue. This patchset is meant to solve both issues. Jerome Brunet (2): ASoC: fix valid stream condition ASoC: skip hw_free on codec dai for which the stream is invalid sound/soc/soc-pcm.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH 1/2] ALSA: hda: fix unregister device twice on ASoC driver
by Bard liao 29 Apr '19

29 Apr '19

snd_hda_codec_device_new() is used by both legacy HDA and ASoC driver. However, we will call snd_hdac_device_unregister() in snd_hdac_ext_bus_device_remove() for ASoC device. This patch uses the type flag in hdac_device struct to determine is it a ASoC device or legacy HDA device and call snd_hdac_device_unregister() in snd_hda_codec_dev_free() only if it is a legacy HDA device. Signed-off-by: Bard liao <yung-chuan.liao(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 701a69d856f5..b20eb7fc83eb 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -832,7 +832,13 @@ static int snd_hda_codec_dev_free(struct snd_device *device) struct hda_codec *codec = device->device_data; codec->in_freeing = 1; - snd_hdac_device_unregister(&codec->core); + /* + * snd_hda_codec_device_new() is used by legacy HDA and ASoC driver. + * We can't unregister ASoC device since it will be unregistered in + * snd_hdac_ext_bus_device_remove(). + */ + if (codec->core.type == HDA_DEV_LEGACY) + snd_hdac_device_unregister(&codec->core); codec_display_power(codec, false); put_device(hda_codec_dev(codec)); return 0; -- 2.17.1

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[alsa-devel] [PATCH] ALSA: usx2y: fix a memory leak bug
by Wenwen Wang 29 Apr '19

29 Apr '19

In usX2Y_In04_init(), a new urb is firstly created through usb_alloc_urb() and saved to 'usX2Y->In04urb'. Then, a buffer is allocated through kmalloc() and saved to 'usX2Y->In04Buf'. After the urb is initialized, a sanity check is performed for the endpoint in the urb by invoking usb_urb_ep_type_check(). If the check fails, the error code EINVAL will be returned. In that case, however, the created urb and the allocated buffer are not freed, leading to memory leaks. To fix the above issue, free the urb and the buffer if the check fails. Signed-off-by: Wenwen Wang <wang6495(a)umn.edu> --- sound/usb/usx2y/usbusx2y.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/usx2y/usbusx2y.c b/sound/usb/usx2y/usbusx2y.c index da4a5a5..0817018 100644 --- a/sound/usb/usx2y/usbusx2y.c +++ b/sound/usb/usx2y/usbusx2y.c @@ -303,8 +303,11 @@ int usX2Y_In04_init(struct usX2Ydev *usX2Y) usX2Y->In04Buf, 21, i_usX2Y_In04Int, usX2Y, 10); - if (usb_urb_ep_type_check(usX2Y->In04urb)) + if (usb_urb_ep_type_check(usX2Y->In04urb)) { + kfree(usX2Y->In04Buf); + usb_put_urb(usX2Y->In04urb); return -EINVAL; + } return usb_submit_urb(usX2Y->In04urb, GFP_KERNEL); } -- 2.7.4

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[alsa-devel] [PATCH] sound: USB: line6: use dynamic buffers
by Greg Kroah-Hartman 28 Apr '19

28 Apr '19

The line6 driver uses a lot of USB buffers off of the stack, which is not allowed on many systems, causing the driver to crash on some of them. Fix this up by dynamically allocating the buffers with kmalloc() which allows for proper DMA-able memory. Reported-by: Christo Gouws <gouws.christo(a)gmail.com> Reported-by: Alan Stern <stern(a)rowland.harvard.edu> Tested-by: Christo Gouws <gouws.christo(a)gmail.com> Cc: stable <stable(a)vger.kernel.org> Signed-off-by: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> --- sound/usb/line6/driver.c | 60 ++++++++++++++++++++++++++------------------- sound/usb/line6/podhd.c | 21 +++++++++------ sound/usb/line6/toneport.c | 23 ++++++++++++----- 3 files changed, 64 insertions(+), 40 deletions(-) --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -351,12 +351,16 @@ int line6_read_data(struct usb_line6 *li { struct usb_device *usbdev = line6->usbdev; int ret; - unsigned char len; + unsigned char *len; unsigned count; if (address > 0xffff || datalen > 0xff) return -EINVAL; + len = kmalloc(sizeof(*len), GFP_KERNEL); + if (!len) + return -ENOMEM; + /* query the serial number: */ ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, @@ -365,7 +369,7 @@ int line6_read_data(struct usb_line6 *li if (ret < 0) { dev_err(line6->ifcdev, "read request failed (error %d)\n", ret); - return ret; + goto exit; } /* Wait for data length. We'll get 0xff until length arrives. */ @@ -375,28 +379,29 @@ int line6_read_data(struct usb_line6 *li ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, - 0x0012, 0x0000, &len, 1, + 0x0012, 0x0000, len, 1, LINE6_TIMEOUT * HZ); if (ret < 0) { dev_err(line6->ifcdev, "receive length failed (error %d)\n", ret); - return ret; + goto exit; } - if (len != 0xff) + if (*len != 0xff) break; } - if (len == 0xff) { + ret = -EIO; + if (*len == 0xff) { dev_err(line6->ifcdev, "read failed after %d retries\n", count); - return -EIO; - } else if (len != datalen) { + goto exit; + } else if (*len != datalen) { /* should be equal or something went wrong */ dev_err(line6->ifcdev, "length mismatch (expected %d, got %d)\n", - (int)datalen, (int)len); - return -EIO; + (int)datalen, (int)*len); + goto exit; } /* receive the result: */ @@ -405,12 +410,12 @@ int line6_read_data(struct usb_line6 *li 0x0013, 0x0000, data, datalen, LINE6_TIMEOUT * HZ); - if (ret < 0) { + if (ret < 0) dev_err(line6->ifcdev, "read failed (error %d)\n", ret); - return ret; - } - return 0; +exit: + kfree(len); + return ret; } EXPORT_SYMBOL_GPL(line6_read_data); @@ -422,12 +427,16 @@ int line6_write_data(struct usb_line6 *l { struct usb_device *usbdev = line6->usbdev; int ret; - unsigned char status; + unsigned char *status; int count; if (address > 0xffff || datalen > 0xffff) return -EINVAL; + status = kmalloc(sizeof(*status), GFP_KERNEL); + if (!status) + return -ENOMEM; + ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x0022, address, data, datalen, @@ -436,7 +445,7 @@ int line6_write_data(struct usb_line6 *l if (ret < 0) { dev_err(line6->ifcdev, "write request failed (error %d)\n", ret); - return ret; + goto exit; } for (count = 0; count < LINE6_READ_WRITE_MAX_RETRIES; count++) { @@ -447,28 +456,29 @@ int line6_write_data(struct usb_line6 *l USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x0012, 0x0000, - &status, 1, LINE6_TIMEOUT * HZ); + status, 1, LINE6_TIMEOUT * HZ); if (ret < 0) { dev_err(line6->ifcdev, "receiving status failed (error %d)\n", ret); - return ret; + goto exit; } - if (status != 0xff) + if (*status != 0xff) break; } - if (status == 0xff) { + if (*status == 0xff) { dev_err(line6->ifcdev, "write failed after %d retries\n", count); - return -EIO; - } else if (status != 0) { + ret = -EIO; + } else if (*status != 0) { dev_err(line6->ifcdev, "write failed (error %d)\n", ret); - return -EIO; + ret = -EIO; } - - return 0; +exit: + kfree(status); + return ret; } EXPORT_SYMBOL_GPL(line6_write_data); --- a/sound/usb/line6/podhd.c +++ b/sound/usb/line6/podhd.c @@ -225,28 +225,32 @@ static void podhd_startup_start_workqueu static int podhd_dev_start(struct usb_line6_podhd *pod) { int ret; - u8 init_bytes[8]; + u8 *init_bytes; int i; struct usb_device *usbdev = pod->line6.usbdev; + init_bytes = kmalloc(8, GFP_KERNEL); + if (!init_bytes) + return -ENOMEM; + ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_OUT, 0x11, 0, NULL, 0, LINE6_TIMEOUT * HZ); if (ret < 0) { dev_err(pod->line6.ifcdev, "read request failed (error %d)\n", ret); - return ret; + goto exit; } /* NOTE: looks like some kind of ping message */ ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | USB_DIR_IN, 0x11, 0x0, - &init_bytes, 3, LINE6_TIMEOUT * HZ); + init_bytes, 3, LINE6_TIMEOUT * HZ); if (ret < 0) { dev_err(pod->line6.ifcdev, "receive length failed (error %d)\n", ret); - return ret; + goto exit; } pod->firmware_version = @@ -255,7 +259,7 @@ static int podhd_dev_start(struct usb_li for (i = 0; i <= 16; i++) { ret = line6_read_data(&pod->line6, 0xf000 + 0x08 * i, init_bytes, 8); if (ret < 0) - return ret; + goto exit; } ret = usb_control_msg(usbdev, usb_sndctrlpipe(usbdev, 0), @@ -263,10 +267,9 @@ static int podhd_dev_start(struct usb_li USB_TYPE_STANDARD | USB_RECIP_DEVICE | USB_DIR_OUT, 1, 0, NULL, 0, LINE6_TIMEOUT * HZ); - if (ret < 0) - return ret; - - return 0; +exit: + kfree(init_bytes); + return ret; } static void podhd_startup_workqueue(struct work_struct *work) --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -365,16 +365,21 @@ static bool toneport_has_source_select(s /* Setup Toneport device. */ -static void toneport_setup(struct usb_line6_toneport *toneport) +static int toneport_setup(struct usb_line6_toneport *toneport) { - u32 ticks; + u32 *ticks; struct usb_line6 *line6 = &toneport->line6; struct usb_device *usbdev = line6->usbdev; + ticks = kmalloc(sizeof(*ticks), GFP_KERNEL); + if (!ticks) + return -ENOMEM; + /* sync time on device with host: */ /* note: 32-bit timestamps overflow in year 2106 */ - ticks = (u32)ktime_get_real_seconds(); - line6_write_data(line6, 0x80c6, &ticks, 4); + *ticks = (u32)ktime_get_real_seconds(); + line6_write_data(line6, 0x80c6, ticks, 4); + kfree(ticks); /* enable device: */ toneport_send_cmd(usbdev, 0x0301, 0x0000); @@ -451,7 +456,9 @@ static int toneport_init(struct usb_line return err; } - toneport_setup(toneport); + err = toneport_setup(toneport); + if (err) + return err; /* register audio system: */ return snd_card_register(line6->card); @@ -463,7 +470,11 @@ static int toneport_init(struct usb_line */ static int toneport_reset_resume(struct usb_interface *interface) { - toneport_setup(usb_get_intfdata(interface)); + int err; + + err = toneport_setup(usb_get_intfdata(interface)); + if (err) + return err; return line6_resume(interface); } #endif

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[alsa-devel] (No Subject)
by rodomar705＠protonmail.com 28 Apr '19

28 Apr '19

Already sent to alsa-user, however a developer asked to send this also on devel, so here it is: I have audio stutters with an ALC892 Realtek Chipset on a B450 GAMING ELITE board from Gigabyte during acquisition (at least with microphone input, both front and back. Line in not tested). No issues with Windows. After opening an issue and testing with a PulseAudio developer, he has determined that the issue is not within PulseAudio, but on ALSA. There are already two issues about that on the bug tracker on Linux, namely this two: ALC892: https://bugzilla.kernel.org/show_bug.cgi?id=203351 ALC1220: https://bugzilla.kernel.org/show_bug.cgi?id=195303 To me both appears to be identical, for this reason even if I have a ALC892 I’ve posted on the second one, just because it had more posts. The particular thing is that acquiring with PulseAudio on Audacity show the problem, however using ALSA directly the issue is not present. Trying using arecord still present the issue from PulseAudio. Tried to play with the parameters /sys/devices/audio to no avail. Compared the pinconfigs from Windows, they are identical. Can someone try to help me fix the issue? Sent with [ProtonMail](https://protonmail.com) Secure Email.

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[alsa-devel] [PATCH v2] ALSA: hda/realtek: Enable audio output of ASUS UX362FA with ALC294
by Ladislav Michl 28 Apr '19

28 Apr '19

The ASUS UX362FA with ALC294 cannot detect the outputs through the internal speaker and the headphone until ALC294_FIXUP_ASUS_SPK quirk applied. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=203443 Suggested-by: Petr Nosek <petr.nosek(a)gmail.com> Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- CHANGES: - v2: adjusted commit message 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 070749b50eff..490b42e3e3dc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6896,6 +6896,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x12f0, "ASUS X541UV", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x12e0, "ASUS X541SA", ALC256_FIXUP_ASUS_MIC), SND_PCI_QUIRK(0x1043, 0x13b0, "ASUS Z550SA", ALC256_FIXUP_ASUS_MIC), + SND_PCI_QUIRK(0x1043, 0x13e1, "ASUS UX362FA", ALC294_FIXUP_ASUS_SPK), SND_PCI_QUIRK(0x1043, 0x1427, "Asus Zenbook UX31E", ALC269VB_FIXUP_ASUS_ZENBOOK), SND_PCI_QUIRK(0x1043, 0x1517, "Asus Zenbook UX31A", ALC269VB_FIXUP_ASUS_ZENBOOK_UX31A), SND_PCI_QUIRK(0x1043, 0x16e3, "ASUS UX50", ALC269_FIXUP_STEREO_DMIC), -- 2.20.1

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[alsa-devel] HDAudio and pulse audio
by rander.wang 28 Apr '19

28 Apr '19

Hi all, I am working on sound open firmware in ASoC to enable HDAudio with pulse audio in ubuntu. There is a HDAudio codec driver hdac_hda.c in ASoC, which invokes HDAudio functions defined in sound/pci/hda, for ASoC to utilize the HDAudio codec. Now I get a issue: pulse audio can't detect headphone pluged or not. Could anyone help to tell me about Jack detection flow in HDAudio and how does HDAudio work with pulse audio ? Does HDAudio report a event to pulse audio when headphone detected ? As I know, ASoC codec driver would invoke snd_soc_jack_report to notify user mode, how about HDAudio driver ? Thanks! Rander

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Re: [alsa-devel] [PATCH] ALSA: hda/realtek: Enable internal speaker of ASUS UX362FA with ALC294
by Ladislav Michl 28 Apr '19

28 Apr '19

Hi Petr, please keep list Cced. On Sun, Apr 28, 2019 at 12:48:06AM +0100, Petr Nosek wrote: > Hi Ladislav, > > it repairs both. Inernal speaker and headphones. Ok, will adjust commit message. > I'm sending output of dmesg - I think it helps you to see information, > when is computer booting. Your patch looks like good. Assuming this part of your boot log comes from patched kernel, right? snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC294: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 snd_hda_codec_realtek hdaudioC0D0: inputs: snd_hda_codec_realtek hdaudioC0D0: Headset Mic=0x19 snd_hda_codec_realtek hdaudioC0D0: Internal Mic=0x12 > Can I do something else to help with repair? I'm not sure, what you mean > by the last sentence. If I have anything else to do. I'll send v2 shortly, Thank you, ladis

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