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

[alsa-devel] [PATCH 0/5] ASoC: stm32: sai: miscellaneous fixes
by Olivier Moysan 04 Mar '19

04 Mar '19

This patch-set is a collection of fixes for STM32 SAI driver. Olivier Moysan (5): ASoC: stm32: sai: fix iec958 controls indexation ASoC: stm32: sai: fix exposed capabilities in spdif mode ASoC: stm32: sai: fix race condition in irq handler ASoC: stm32: sai: fix oversampling mode ASoC: stm32: sai: fix set_sync service sound/soc/stm/stm32_sai.c | 7 ++++--- sound/soc/stm/stm32_sai_sub.c | 48 ++++++++++++++++++++++++++++++++++--------- 2 files changed, 42 insertions(+), 13 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ASoC: cs42xx8: Remove S32_LE in format list
by S.j. Wang 02 Mar '19

02 Mar '19

cs42xx8 is a 24-bit A/D and 24-bit D/A device, so the S32_LE should not be in the supported format list. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..271972337d23 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -32,8 +32,7 @@ #define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S20_3LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) + SNDRV_PCM_FMTBIT_S24_LE) /* codec private data */ struct cs42xx8_priv { -- 1.9.1

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[alsa-devel] [PATCH 3/3] arm64: dts: hi3660-hikey960: Add i2s & sound device
by Pengcheng Li 01 Mar '19

01 Mar '19

From: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Youlin Wang <wangyoulin1(a)hisilicon.com> Signed-off-by: Tanglei Han <hantanglei(a)huawei.com> Signed-off-by: Guangke Ji <jiguangke(a)huawei.com> Signed-off-by: Feng Chen <puck.chen(a)hisilicon.com> Signed-off-by: Kaihua Zhong <zhongkaihua(a)huawei.com> Signed-off-by: Jun Chen <chenjun14(a)huawei.com> Signed-off-by: John Stultz <john.stultz(a)linaro.org> Signed-off-by: Yiping Xu <xuyiping(a)hisilicon.com> Signed-off-by: Pengcheng Li <lipengcheng8(a)huawei.com> Cc: Wei Xu <xuwei5(a)hisilicon.com> Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Mark Rutland <mark.rutland(a)arm.com> --- arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts | 29 +++++++++++++++++++++++ 1 file changed, 29 insertions(+) diff --git a/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts b/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts index 186251f..457cebc 100644 --- a/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts +++ b/arch/arm64/boot/dts/hisilicon/hi3660-hikey960.dts @@ -27,6 +27,35 @@ /* rewrite this at bootloader */ reg = <0x0 0x0 0x0 0x0>; }; + + i2s2: hi3660_i2s { + compatible = "hisilicon,hi3660-i2s-1.0"; + reg = <0x0 0xe804f800 0x0 0x400>, + <0x0 0xe804e000 0x0 0x400>; + pinctrl-names = "default"; + pinctrl-0 = <&i2s2_pmx_func &i2s2_cfg_func>; + dmas = <&asp_dmac 18 &asp_dmac 19>; + dma-names = "rx", "tx"; + #sound-dai-cells = <0>; + }; + + sound { + compatible = "simple-audio-card"; + simple-audio-card,name = "hikey-hdmi"; + simple-audio-card,format = "i2s"; + + simple-audio-card,bitclock-master = <&sound_master>; + simple-audio-card,frame-master = <&sound_master>; + + sound_master: simple-audio-card,cpu { + sound-dai = <&i2s2>; + }; + + simple-audio-card,codec { + sound-dai = <&adv7533>; + }; + }; + }; &uart5 { -- 2.8.0

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[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MOTU MicroBook II
by Manuel Reinhardt 28 Feb '19

28 Feb '19

Add an entry to the quirks-table to for usb-audio to recognize the Microbook II (although it only exposes vendor interfaces). A simple boot quirk is also implemented to set up the sample rate and make sure that no audio urbs are sent before the device is ready. This patch only provides audio playback and capture at 96kHz sample rate. Notice the following shortcomings: - The sample rate is currently hardcoded to 96k although the device also supports 48k and 44.1k. - The various mixer controls of the MicroBook are not made available. - The keep-iface control should be on by default because the device shuts down whenever the altsetting is reset which is usually unwanted. (I don't know the best way to do this) - The communication format used by the MicroBook for sample rate setting and also other setup has been reverse engineered by looking at the usbmon output while running the windows driver through virtualbox. In this patch the first byte of every message is set to \0 while in the observed communications the first byte acts as a "message-counter" increasing its value with every message sent. Leaving it at \0 does not seem to affect the device. Signed-off-by: Manuel Reinhardt <manuel.rhdt(a)gmail.com> --- sound/usb/pcm.c | 4 ++ sound/usb/quirks-table.h | 65 +++++++++++++++++++++++++ sound/usb/quirks.c | 101 +++++++++++++++++++++++++++++++++++++++ 3 files changed, 170 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index db114f3977e0..056af0a57b22 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,6 +354,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + ep = 0x84; + ifnum = 0; + goto add_sync_ep_from_ifnum; } if (attr == USB_ENDPOINT_SYNC_ASYNC && diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b345beb447bd..86e80916a029 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3398,5 +3398,70 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* MOTU Microbook II */ +{ + USB_DEVICE(0x07fd, 0x0004), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MOTU", + .product_name = "MicroBookII", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x84, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x00d8, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x03, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x0120, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f372c624bbf4..e6ce1bbe6ca6 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1000,6 +1000,105 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } + +#define MICROBOOK_BUF_SIZE 128 + +static int snd_usb_motu_microbookii_communicate(struct usb_device *dev, u8 *buf, + int buf_size, int *length) +{ + int err, actual_length; + + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x01), buf, *length, + &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "MicroBookII snd: ", DUMP_PREFIX_NONE, 16, 1, + buf, actual_length, false); + + memset(buf, 0, buf_size); + + err = usb_interrupt_msg(dev, usb_rcvintpipe(dev, 0x82), buf, buf_size, + &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "MicroBookII rcv: ", DUMP_PREFIX_NONE, 16, 1, + buf, actual_length, false); + + *length = actual_length; + return 0; +} + +static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) +{ + int err, actual_length, poll_attempts = 0; + static const u8 set_samplerate_seq[] = { 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x14, + 0x00, 0x00, 0x00, 0x01 }; + static const u8 poll_ready_seq[] = { 0x00, 0x04, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x18 }; + u8 *buf = kzalloc(MICROBOOK_BUF_SIZE, GFP_KERNEL); + + if (!buf) + return -ENOMEM; + + dev_info(&dev->dev, "Waiting for MOTU Microbook II to boot up...\n"); + + /* First we tell the device which sample rate to use. */ + memcpy(buf, set_samplerate_seq, sizeof(set_samplerate_seq)); + actual_length = sizeof(set_samplerate_seq); + err = snd_usb_motu_microbookii_communicate(dev, buf, MICROBOOK_BUF_SIZE, + &actual_length); + + if (err < 0) { + dev_err(&dev->dev, + "failed setting the sample rate for Motu MicroBook II: %d\n", + err); + goto free_buf; + } + + /* Then we poll every 100 ms until the device informs of its readiness. */ + while (true) { + if (++poll_attempts > 100) { + dev_err(&dev->dev, + "failed booting Motu MicroBook II: timeout\n"); + err = -ENODEV; + goto free_buf; + } + + memset(buf, 0, MICROBOOK_BUF_SIZE); + memcpy(buf, poll_ready_seq, sizeof(poll_ready_seq)); + + actual_length = sizeof(poll_ready_seq); + err = snd_usb_motu_microbookii_communicate( + dev, buf, MICROBOOK_BUF_SIZE, &actual_length); + if (err < 0) { + dev_err(&dev->dev, + "failed booting Motu MicroBook II: communication error %d\n", + err); + goto free_buf; + } + + /* the device signals its readiness through a message of the + * form + * XX 06 00 00 00 00 0b 18 00 00 00 01 + * If the device is not yet ready to accept audio data, the + * last byte of that sequence is 00. + */ + if (actual_length == 12 && buf[actual_length - 1] == 1) + break; + + msleep(100); + } + + dev_info(&dev->dev, "MOTU MicroBook II ready\n"); + +free_buf: + kfree(buf); + return err; +} + /* * Setup quirks */ @@ -1177,6 +1276,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_gamecon780_boot_quirk(dev); case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + return snd_usb_motu_microbookii_boot_quirk(dev); } return 0; -- 2.20.1

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[alsa-devel] [PATCH] ALSA: usb-audio: Add quirk for MOTU MicroBook II
by Manuel Reinhardt 28 Feb '19

28 Feb '19

Add an entry to the quirks-table to for usb-audio to recognize the Microbook II (although it only exposes vendor interfaces). A simple boot quirk is also implemented to set up the sample rate and make sure that no audio urbs are sent before the device is ready. This patch only provides audio playback and capture at 96kHz sample rate. Notice the following shortcomings: - The sample rate is currently hardcoded to 96k although the device also supports 48k and 44.1k. - The various mixer controls of the MicroBook are not made available. - The keep-iface control should be on by default because the device shuts down whenever the altsetting is reset which is usually unwanted. (I don't know the best way to do this) - The communication format used by the MicroBook for sample rate setting and also other setup has been reverse engineered by looking at the usbmon output while running the windows driver through virtualbox. In this patch the first byte of every message is set to \0 while in the observed communications the first byte acts as a "message-counter" increasing its value with every message sent. Leaving it at \0 does not seem to affect the device. Signed-off-by: Manuel Reinhardt <manuel.rhdt(a)gmail.com> --- sound/usb/pcm.c | 4 ++ sound/usb/quirks-table.h | 65 +++++++++++++++++++++++++++ sound/usb/quirks.c | 97 ++++++++++++++++++++++++++++++++++++++++ 3 files changed, 166 insertions(+) diff --git a/sound/usb/pcm.c b/sound/usb/pcm.c index db114f3977e0..056af0a57b22 100644 --- a/sound/usb/pcm.c +++ b/sound/usb/pcm.c @@ -354,6 +354,10 @@ static int set_sync_ep_implicit_fb_quirk(struct snd_usb_substream *subs, ep = 0x81; ifnum = 1; goto add_sync_ep_from_ifnum; + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + ep = 0x84; + ifnum = 0; + goto add_sync_ep_from_ifnum; } if (attr == USB_ENDPOINT_SYNC_ASYNC && diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index b345beb447bd..86e80916a029 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3398,5 +3398,70 @@ AU0828_DEVICE(0x2040, 0x7270, "Hauppauge", "HVR-950Q"), .ifnum = QUIRK_NO_INTERFACE } }, +/* MOTU Microbook II */ +{ + USB_DEVICE(0x07fd, 0x0004), + .driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) { + .vendor_name = "MOTU", + .product_name = "MicroBookII", + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_COMPOSITE, + .data = (const struct snd_usb_audio_quirk[]) { + { + .ifnum = 0, + .type = QUIRK_AUDIO_STANDARD_MIXER, + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 6, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x84, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x00d8, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = 0, + .type = QUIRK_AUDIO_FIXED_ENDPOINT, + .data = &(const struct audioformat) { + .formats = SNDRV_PCM_FMTBIT_S24_3BE, + .channels = 8, + .iface = 0, + .altsetting = 1, + .altset_idx = 1, + .attributes = 0, + .endpoint = 0x03, + .rates = SNDRV_PCM_RATE_96000, + .ep_attr = USB_ENDPOINT_XFER_ISOC | + USB_ENDPOINT_SYNC_ASYNC, + .rate_min = 96000, + .rate_max = 96000, + .nr_rates = 1, + .maxpacksize = 0x0120, + .rate_table = (unsigned int[]) { + 96000 + } + } + }, + { + .ifnum = -1 + } + } + } +}, #undef USB_DEVICE_VENDOR_SPEC diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index f372c624bbf4..c86a06dd97dd 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1000,6 +1000,101 @@ static int snd_usb_axefx3_boot_quirk(struct usb_device *dev) return 0; } + +#define MICROBOOK_BUF_SIZE 128 + +static int snd_usb_motu_microbookii_communicate(struct usb_device *dev, u8 *buf, + int buf_size, int *length) +{ + int err, actual_length; + + err = usb_interrupt_msg(dev, usb_sndintpipe(dev, 0x01), buf, *length, + &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "snd: ", DUMP_PREFIX_NONE, 16, 1, buf, + actual_length, false); + + memset(buf, 0, buf_size); + + err = usb_interrupt_msg(dev, usb_rcvintpipe(dev, 0x82), buf, + buf_size, &actual_length, 1000); + if (err < 0) + return err; + + print_hex_dump(KERN_DEBUG, "rcv: ", DUMP_PREFIX_NONE, 16, 1, buf, + actual_length, false); + + *length = actual_length; + return 0; +} + +static int snd_usb_motu_microbookii_boot_quirk(struct usb_device *dev) +{ + int err, actual_length, poll_attempts = 0; + bool setup_finished = false; + static const u8 set_samplerate_seq[] = { 0x00, 0x00, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x14, + 0x00, 0x00, 0x00, 0x01 }; + static const u8 poll_ready_seq[] = { 0x00, 0x04, 0x00, 0x00, + 0x00, 0x00, 0x0b, 0x18 }; + u8 *buf = kzalloc(MICROBOOK_BUF_SIZE, GFP_KERNEL); + + if (!buf) + return -ENOMEM; + + dev_info(&dev->dev, "Waiting for MOTU Microbook II to boot up...\n"); + + /* First we tell the device which sample rate to use. */ + memcpy(buf, set_samplerate_seq, ARRAY_SIZE(set_samplerate_seq)); + actual_length = ARRAY_SIZE(set_samplerate_seq); + err = snd_usb_motu_microbookii_communicate(dev, buf, MICROBOOK_BUF_SIZE, + &actual_length); + + if (err < 0) { + kfree(buf); + dev_err(&dev->dev, + "failed setting the sample rate for Motu MicroBook II: %d\n", + err); + return err; + } + + /* Then we poll every 100 ms until the device informs of its readiness. */ + while (!setup_finished) { + if (++poll_attempts > 100) { + kfree(buf); + dev_err(&dev->dev, + "failed booting Motu MicroBook II: timeout\n"); + return -ENODEV; + } + + memset(buf, 0, MICROBOOK_BUF_SIZE); + memcpy(buf, poll_ready_seq, ARRAY_SIZE(poll_ready_seq)); + + actual_length = ARRAY_SIZE(poll_ready_seq); + err = snd_usb_motu_microbookii_communicate( + dev, buf, MICROBOOK_BUF_SIZE, &actual_length); + if (err < 0) { + kfree(buf); + dev_err(&dev->dev, + "failed booting Motu MicroBook II: communication error\n"); + return err; + } + + if (buf[actual_length - 1] == 1) + setup_finished = true; + + msleep(100); + } + + kfree(buf); + + dev_info(&dev->dev, "MOTU MicroBook II ready\n"); + + return 0; +} + /* * Setup quirks */ @@ -1177,6 +1272,8 @@ int snd_usb_apply_boot_quirk(struct usb_device *dev, return snd_usb_gamecon780_boot_quirk(dev); case USB_ID(0x2466, 0x8010): /* Fractal Audio Axe-Fx 3 */ return snd_usb_axefx3_boot_quirk(dev); + case USB_ID(0x07fd, 0x0004): /* MOTU MicroBook II */ + return snd_usb_motu_microbookii_boot_quirk(dev); } return 0; -- 2.20.1

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Re: [alsa-devel] [PATCH] dt-bindings: adau1977: Fix reset-gpios typo
by Rob Herring 28 Feb '19

28 Feb '19

On Fri, 15 Feb 2019 12:26:33 +0200, Bogdan Togorean wrote: > This change fixes a typo in the dt-binding examples (reset_gpio -> > reset-gpios). > Even though 'reset-gpio' is a valid construct for gpiolib the naming > 'reset-gpios' is more suited for dt-bindings documentation. > > Signed-off-by: Bogdan Togorean <bogdan.togorean(a)analog.com> > --- > Documentation/devicetree/bindings/sound/adi,adau1977.txt | 6 +++--- > 1 file changed, 3 insertions(+), 3 deletions(-) > Reviewed-by: Rob Herring <robh(a)kernel.org>

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[alsa-devel] WM8510 on i.MX6
by Schrempf Frieder 28 Feb '19

28 Feb '19

Hi, I have a board with i.MX6 and a WM8510 codec and I can play audio using an old vendor kernel (3.14) and my own asoc machine driver [1]. Now I want to run this with something more recent (currently trying on 4.14) and upstream drivers like simple-card or fsl-asoc-card. The problem is I'm only getting a little bit of noise from the speakers but nothing more. I would really appreciate it if someone more knowledgeable could have a quick look and maybe come up with some ideas how I can proceed debugging. Thanks, Frieder [1] https://paste.ee/p/VuZbg My devicetree setup for the simple-card approach looks like this: sound { compatible = "simple-audio-card"; simple-audio-card,name = "wm8510-audio"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&sound_codec>; simple-audio-card,frame-master = <&sound_codec>; simple-audio-card,widgets = "Speaker", "Ext Spk"; simple-audio-card,routing = "Ext Spk", "SPKOUTP", "Ext Spk", "SPKOUTN"; assigned-clocks = <&clks IMX6QDL_CLK_CKO2_SEL>, <&clks IMX6QDL_CLK_CKO2>; assigned-clock-parents = <&clks IMX6QDL_CLK_OSC>, <&clks IMX6QDL_CLK_CKO2_PODF>; assigned-clock-rates = <0 24000000>; sound_master: simple-audio-card,cpu { sound-dai = <&ssi2>; }; sound_codec: simple-audio-card,codec { sound-dai = <&codec>; clocks = <&clks IMX6QDL_CLK_CKO2>; }; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_audmux>; status = "okay"; ssi2 { fsl,audmux-port = <1>; fsl,port-config = < (IMX_AUDMUX_V2_PTCR_SYN | IMX_AUDMUX_V2_PTCR_TFSEL(4) | IMX_AUDMUX_V2_PTCR_TCSEL(4) | IMX_AUDMUX_V2_PTCR_TFSDIR | IMX_AUDMUX_V2_PTCR_TCLKDIR) IMX_AUDMUX_V2_PDCR_RXDSEL(4) >; }; aud5 { fsl,audmux-port = <4>; fsl,port-config = < IMX_AUDMUX_V2_PTCR_SYN IMX_AUDMUX_V2_PDCR_RXDSEL(1) >; }; }; &ssi2 { status = "okay"; }; &i2c2 { status = "okay"; clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c2>; codec: wm8510@1a { #sound-dai-cells = <0>; compatible = "wlf,wm8510"; reg = <0x1a>; }; }; &iomuxc { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_sound_mclk>; pinctrl_sound_mclk: sound { /* WM8510 MCLK */ fsl,pins = < MX6QDL_PAD_NANDF_CS2__CCM_CLKO2 0x130b0 >; }; pinctrl_audmux: audmux { fsl,pins = < MX6QDL_PAD_KEY_COL0__AUD5_TXC 0x130b0 MX6QDL_PAD_KEY_COL1__AUD5_TXFS 0x130b0 MX6QDL_PAD_KEY_ROW1__AUD5_RXD 0x130b0 MX6QDL_PAD_KEY_ROW0__AUD5_TXD 0x130b0 >; }; };

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[alsa-devel] Applied "ASoC: tlv320aic3x: fix reset gpio reference counting" to the asoc tree
by Mark Brown 28 Feb '19

28 Feb '19

The patch ASoC: tlv320aic3x: fix reset gpio reference counting has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 82ad759143ed77673db0d93d53c1cde7b99917ee Mon Sep 17 00:00:00 2001 From: Philipp Puschmann <philipp.puschmann(a)emlix.com> Date: Wed, 27 Feb 2019 16:17:33 +0100 Subject: [PATCH] ASoC: tlv320aic3x: fix reset gpio reference counting This patch fixes a bug that prevents freeing the reset gpio on unloading the module. aic3x_i2c_probe is called when loading the module and it calls list_add with a probably uninitialized list entry aic3x->list (next = prev = NULL)). So even if list_del is called it does nothing and in the end the gpio_reset is not freed. Then a repeated module probing fails silently because gpio_request fails. When moving INIT_LIST_HEAD to aic3x_i2c_probe we also have to move list_del to aic3x_i2c_remove because aic3x_remove may be called multiple times without aic3x_i2c_remove being called which leads to a NULL pointer dereference. Signed-off-by: Philipp Puschmann <philipp.puschmann(a)emlix.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic3x.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6aa0edf8c5ef..cea3ebecdb12 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1609,7 +1609,6 @@ static int aic3x_probe(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int ret, i; - INIT_LIST_HEAD(&aic3x->list); aic3x->component = component; for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { @@ -1692,7 +1691,6 @@ static void aic3x_remove(struct snd_soc_component *component) struct aic3x_priv *aic3x = snd_soc_component_get_drvdata(component); int i; - list_del(&aic3x->list); for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) regulator_unregister_notifier(aic3x->supplies[i].consumer, &aic3x->disable_nb[i].nb); @@ -1890,6 +1888,7 @@ static int aic3x_i2c_probe(struct i2c_client *i2c, if (ret != 0) goto err_gpio; + INIT_LIST_HEAD(&aic3x->list); list_add(&aic3x->list, &reset_list); return 0; @@ -1906,6 +1905,8 @@ static int aic3x_i2c_remove(struct i2c_client *client) { struct aic3x_priv *aic3x = i2c_get_clientdata(client); + list_del(&aic3x->list); + if (gpio_is_valid(aic3x->gpio_reset) && !aic3x_is_shared_reset(aic3x)) { gpio_set_value(aic3x->gpio_reset, 0); -- 2.20.1

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[alsa-devel] hdmi-codec: modifying params in hw_params() callback
by Russell King - ARM Linux admin 28 Feb '19

28 Feb '19

Hi all, While looking at hdmi-codec issues, I spotted this code: static int hdmi_codec_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { ... if (params_width(params) > 24) params->msbits = 24; params->msbits is a parameter that is negotiated and refined by libasound, and is part of the ALSA constraint system. The "Writing an ALSA driver" documentation says about the hw_params callback: "This is called when the hardware parameter (``hw_params``) is set up by the application, that is, once when the buffer size, the period size, the format, etc. are defined for the pcm substream." which suggests we should only be reading the parameters, not writing to them. What's more is that the msbits is a parameter that is refined with userspace, so surely the above should be a declared constraint? Digging a bit deeper, ASoC passes a private copy of the params to each codec - a copy is made, then fixups for TDM slots are applied, followed by any topology fixups by the DAI link (be_hw_params_fixup) before the codec driver's hw_params() callback is made. Afterwards, ASoC reads back the rate, channels and physical (memory) width and stores them in the codec's DAI structure. The msbits are not read. hdmi-codec also seems to do nothing with the msbits parameter other than the above code. So, all in all, it seems that the above code limiting msbits is redundant - nothing will read this modified value. Can we kill it? Thanks. -- RMK's Patch system: https://www.armlinux.org.uk/developer/patches/ FTTC broadband for 0.8mile line in suburbia: sync at 12.1Mbps down 622kbps up According to speedtest.net: 11.9Mbps down 500kbps up

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[alsa-devel] ftp.alsa-project.org connection problems
by Michael Niedermayer 28 Feb '19

28 Feb '19

Hi Iam not sure this is the correct place to report this to but i didnt see it being reported anywhere yet on the ML archive or google since a few days wget from ftp://ftp.alsa-project.org does not work anymore This was noticed as it broke the ffmpeg oss-fuzz build: https://oss-fuzz-build-logs.storage.googleapis.com/log-a252663b-b71c-49f3-8… and it is locally and on another server reproducable so it seems not a issue "on my side" intererstingly it works without pasv mode wget ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.1.0.tar.bz2 --2019-02-25 18:30:15-- ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.1.0.tar.bz2 => ‘alsa-lib-1.1.0.tar.bz2’ Resolving ftp.alsa-project.org (ftp.alsa-project.org)... 207.180.221.201 Connecting to ftp.alsa-project.org (ftp.alsa-project.org)|207.180.221.201|:21... connected. Logging in as anonymous ... Logged in! ==> SYST ... done. ==> PWD ... done. ==> TYPE I ... done. ==> CWD (1) /pub/lib ... done. ==> SIZE alsa-lib-1.1.0.tar.bz2 ... 929874 ==> PASV ... couldn't connect to 207.180.221.201 port 35333: No route to host wget --no-passive-ftp ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.1.0.tar.bz2 --2019-02-25 18:30:22-- ftp://ftp.alsa-project.org/pub/lib/alsa-lib-1.1.0.tar.bz2 => ‘alsa-lib-1.1.0.tar.bz2’ Resolving ftp.alsa-project.org (ftp.alsa-project.org)... 207.180.221.201 Connecting to ftp.alsa-project.org (ftp.alsa-project.org)|207.180.221.201|:21... connected. Logging in as anonymous ... Logged in! ==> SYST ... done. ==> PWD ... done. ==> TYPE I ... done. ==> CWD (1) /pub/lib ... done. ==> SIZE alsa-lib-1.1.0.tar.bz2 ... 929874 ==> PORT ... done. ==> RETR alsa-lib-1.1.0.tar.bz2 ... done. Length: 929874 (908K) (unauthoritative) 100%[===========================================================================================================================================================================================================================================================================>] 929,874 2.43MB/s in 0.4s 2019-02-25 18:30:22 (2.43 MB/s) - ‘alsa-lib-1.1.0.tar.bz2’ saved [929874] -- Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB I know you won't believe me, but the highest form of Human Excellence is to question oneself and others. -- Socrates

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