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Realtek analog audio output limited to 48 kHz, chips support up to 192kHz
by GitHub issues - edited 11 Jul '21

11 Jul '21

alsa-project/alsa-lib issue #163 was edited from markojurkovic: There seems to be an issue with ALSA and some onboard Realtek laptop audio chips, in my case Realtek ALC233 (but I've seen some reports online of people with other Realtek chips facing this issue). Namely, I've noticed that I cannot make use of all the sampling rates that the chip supports in GNU/Linux using ALSA. The chip supports up to 24-bit/192kHz analog stereo output, but in the three Linux distributions I've tried (Ubuntu, Fedora, Manjaro) I can only get either 44.1 or 48 kHz. The problem manifests itself by throwing the following error when attempting to play, for example, a 96kHz file in Audacious: "ALSA error: snd_pcm_hw_params_set_rate failed: Invalid argument." If I attempt to play such a file using PulseAudio, I am able to play the file, but it gets resampled to 48kHz (which I've verified by running "cat /proc/asound/card0/pcm0p/sub0/hw_params") In Windows, I can get up to 192kHz without problems using both WASAPI Exclusive and ASIO modes, meaning modes that directly communicate with hardware, just like ALSA. Following is the output of running "cat /proc/asound/card0/codec#0". This output indicates that the only supported sample rates for the sound card are 44100 and 48000 (for the analog outputs), whereas this Realtek chip supports also 96000 and 192000. Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x44 0x44] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC233 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Following is also the output of the alsa-info script: [alsa-debug.txt](https://github.com/alsa-project/alsa-lib/files/6797358/alsa… I tried different kernel versions, both generic and custom, on distros ranging from Ubuntu 19.04, Ubuntu 21.04, Pop OS 20.04 and 21.04, Fedora 32... Issue URL : https://github.com/alsa-project/alsa-lib/issues/163 Repository URL: https://github.com/alsa-project/alsa-lib

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Realtek analog audio output limited to 48 kHz, chips support up to 192kHz
by GitHub issues - edited 11 Jul '21

11 Jul '21

alsa-project/alsa-lib issue #163 was edited from markojurkovic: There seems to be an issue with ALSA and some onboard Realtek laptop audio chips, in my case Realtek ALC233 (but I've seen some reports online of people with other Realtek chips facing this issue). Namely, I've noticed that I cannot make use of all the sampling rates that the chip supports in GNU/Linux using ALSA. The chip supports up to 24-bit/192kHz analog stereo output, but in the three Linux distributions I've tried (Ubuntu, Fedora, Manjaro) I can only get either 44.1 or 48 kHz. The problem manifests itself by throwing the following error when attempting to play, for example, a 96kHz file in Audacious: "ALSA error: snd_pcm_hw_params_set_rate failed: Invalid argument." If I attempt to play such a file using PulseAudio, I am able to play the file, but it gets resampled to 48kHz (which I've verified by running "cat /proc/asound/card0/pcm0p/sub0/hw_params") In Windows, I can get up to 192kHz without problems using both WASAPI Exclusive and ASIO modes, meaning modes that directly communicate with hardware, just like ALSA. Following is the output of running "cat /proc/asound/card0/codec#0". This output indicates that the only supported sample rates for the sound card are 44100 and 48000 (for the analog outputs), whereas this Realtek chip supports also 96000 and 192000. Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x44 0x44] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC233 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Attached to this post is also the output of the alsa-info script. [alsa-debug.txt](https://github.com/alsa-project/alsa-lib/files/6797358/alsa… I tried different kernel versions, both generic and custom, on distros ranging from Ubuntu 19.04, Ubuntu 21.04, Pop OS 20.04 and 21.04, Fedora 32... Issue URL : https://github.com/alsa-project/alsa-lib/issues/163 Repository URL: https://github.com/alsa-project/alsa-lib

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Realtek analog audio output limited to 48 kHz, chips support up to 192kHz
by GitHub issues - opened 11 Jul '21

11 Jul '21

alsa-project/alsa-lib issue #163 was opened from markojurkovic: There seems to be an issue with ALSA and some onboard Realtek laptop audio chips, in my case Realtek ALC233 (but I've seen some reports online of people with other Realtek chips facing this issue). Namely, I've noticed that I cannot make use of all the sampling rates that the chip supports in GNU/Linux using ALSA. The chip supports up to 24-bit/192kHz analog stereo output, but in the three Linux distributions I've tried (Ubuntu, Fedora, Manjaro) I can only get either 44.1 or 48 kHz. The problem manifests itself by throwing the following error when attempting to play, for example, a 96kHz file in Audacious: "ALSA error: snd_pcm_hw_params_set_rate failed: Invalid argument." If I attempt to play such a file using PulseAudio, I am able to play the file, but it gets resampled to 48kHz (which I've verified by running "cat /proc/asound/card0/pcm0p/sub0/hw_params") In Windows, I can get up to 192kHz without problems using both WASAPI Exclusive and ASIO modes, meaning modes that directly communicate with hardware, just like ALSA. Following is the output of running "cat /proc/asound/card0/codec#0". This output indicates that the only supported sample rates for the sound card are 44100 and 48000, whereas this Realtek chip supports also 96000 and 192000. Node 0x02 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x44 0x44] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Node 0x03 [Audio Output] wcaps 0x41d: Stereo Amp-Out Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="ALC233 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x57, nsteps=0x57, stepsize=0x02, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=1, channel=0 PCM: rates [0x60]: 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 EPSS Power: setting=D0, actual=D0 Attached to this post is also the output of the alsa-info script. [alsa-debug.txt](https://github.com/alsa-project/alsa-lib/files/6797358/alsa… I tried different kernel versions, both generic and custom, on distros ranging from Ubuntu 19.04, Ubuntu 21.04, Pop OS 20.04 and 21.04, Fedora 32... Issue URL : https://github.com/alsa-project/alsa-lib/issues/163 Repository URL: https://github.com/alsa-project/alsa-lib

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Re: "Chipmunk audio" on Logitech USB webcams
by Tristan Miller 11 Jul '21

11 Jul '21

Greetings. On Sun, 11 Jul 2021 12:38:12 +0200, Greg KH <gregkh(a)linuxfoundation.org> wrote: > On Sun, Jul 11, 2021 at 10:25:03AM +0200, Tristan Miller wrote: > > Many models of Logitech USB webcams malfunction under Linux by > > producing audio that sounds distorted, garbled, or high-pitched. > > For most users, the problem happens randomly, and can usually be > > worked around by restarting the audio application or by unplugging > > the webcam and then plugging it back in again. > > Please report the information directly here, to this list, and to the > sound developers at alsa-devel(a)alsa-project.org and we can all work to > resolve this. The problem as I'm currently experiencing it is as follows: I have a Logitech C270 USB webcam running on openSUSE Tumbleweed for x86_64 with the 5.13.0 kernel. Sometimes audio recorded by the camera's built-in microphone sounds artificially high-pitched and garbled. I've experienced this with lots of different applications, including arecord, Skype, and various browser-based videoconferencing software running in Firefox or Chromium. The problem occurs rarely (maybe once every 30 times I use the webcam microphone) and seemingly at random; restarting the audio application or unplugging the webcam and plugging it back in again usually works around the issue. So unfortunately I can't reliably reproduce the problem. I haven't noticed any errors in the system logs but then again am not sure exactly what to look for and where. I've been experiencing this problem at least as far back as kernel 4.13.12. Some further details (more available on request): # uname -a Linux butterfield.nothingisreal.com 5.13.0-1-default #1 SMP Fri Jul 2 05:54:32 UTC 2021 (aa40472) x86_64 x86_64 x86_64 GNU/Linux # lsusb -s 001:004 -v |head -17 Bus 001 Device 004: ID 046d:0825 Logitech, Inc. Webcam C270 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x046d Logitech, Inc. idProduct 0x0825 Webcam C270 bcdDevice 0.10 iManufacturer 0 iProduct 0 iSerial 2 6457F4D0 bNumConfigurations Previous reports which apparently cover the same problem: https://bugzilla.kernel.org/show_bug.cgi?id=35922 https://bugzilla.kernel.org/show_bug.cgi?id=44281 https://bugzilla.kernel.org/show_bug.cgi?id=105081 https://bugzilla.kernel.org/show_bug.cgi?id=203763 https://bugs.launchpad.net/ubuntu/+source/linux/+bug/459445 https://bugs.launchpad.net/ubuntu/+source/linux/+bug/843431 https://bugs.launchpad.net/ubuntu/+source/linux/+bug/884210 https://bugs.launchpad.net/ubuntu/+source/linux/+bug/1018020 https://bugzilla.suse.com/show_bug.cgi?id=736093 https://bugzilla.redhat.com/show_bug.cgi?id=729269 https://bugzilla.redhat.com/show_bug.cgi?id=742010 https://bugs.archlinux.org/task/26528 https://bbs.archlinux.org/viewtopic.php?id=121607 https://bbs.archlinux.org/viewtopic.php?id=236056 https://forums.linuxmint.com/viewtopic.php?t=237677 Regards, Tristan -- =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Tristan Miller Free Software developer, ferret herder, logologist https://logological.org/ =-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-

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[PATCH v5 1/2] ASoC: sunxi: Add Allwinner H6 Digital MIC driver
by fengzheng923＠gmail.com 10 Jul '21

10 Jul '21

From: Ban Tao <fengzheng923(a)gmail.com> The Allwinner H6 and later SoCs have an DMIC block which is capable of capture. Signed-off-by: Ban Tao <fengzheng923(a)gmail.com> --- v1->v2: 1.Fix some compilation errors. 2.Modify some code styles. --- v2->v3: None. --- v3->v4: 1.add sig_bits. --- v4->v5: None. --- MAINTAINERS | 7 + sound/soc/sunxi/Kconfig | 8 + sound/soc/sunxi/Makefile | 1 + sound/soc/sunxi/sun50i-dmic.c | 405 ++++++++++++++++++++++++++++++++++ 4 files changed, 421 insertions(+) create mode 100644 sound/soc/sunxi/sun50i-dmic.c diff --git a/MAINTAINERS b/MAINTAINERS index e924f9e5df97..8d700baaa3ca 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -760,6 +760,13 @@ L: linux-media(a)vger.kernel.org S: Maintained F: drivers/staging/media/sunxi/cedrus/ +ALLWINNER DMIC DRIVERS +M: Ban Tao <fengzheng923(a)gmail.com> +L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) +S: Maintained +F: Documentation/devicetree/bindings/sound/allwinner,sun50i-h6-dmic.yaml +F: sound/soc/sunxi/sun50i-dmic.c + ALPHA PORT M: Richard Henderson <rth(a)twiddle.net> M: Ivan Kokshaysky <ink(a)jurassic.park.msu.ru> diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig index ddcaaa98d3cb..2a3bf7722e11 100644 --- a/sound/soc/sunxi/Kconfig +++ b/sound/soc/sunxi/Kconfig @@ -56,6 +56,14 @@ config SND_SUN4I_SPDIF Say Y or M to add support for the S/PDIF audio block in the Allwinner A10 and affiliated SoCs. +config SND_SUN50I_DMIC + tristate "Allwinner H6 DMIC Support" + depends on (OF && ARCH_SUNXI) || COMPILE_TEST + select SND_SOC_GENERIC_DMAENGINE_PCM + help + Say Y or M to add support for the DMIC audio block in the Allwinner + H6 and affiliated SoCs. + config SND_SUN8I_ADDA_PR_REGMAP tristate select REGMAP diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile index a86be340a076..4483fe9c94ef 100644 --- a/sound/soc/sunxi/Makefile +++ b/sound/soc/sunxi/Makefile @@ -6,3 +6,4 @@ obj-$(CONFIG_SND_SUN8I_CODEC_ANALOG) += sun8i-codec-analog.o obj-$(CONFIG_SND_SUN50I_CODEC_ANALOG) += sun50i-codec-analog.o obj-$(CONFIG_SND_SUN8I_CODEC) += sun8i-codec.o obj-$(CONFIG_SND_SUN8I_ADDA_PR_REGMAP) += sun8i-adda-pr-regmap.o +obj-$(CONFIG_SND_SUN50I_DMIC) += sun50i-dmic.o diff --git a/sound/soc/sunxi/sun50i-dmic.c b/sound/soc/sunxi/sun50i-dmic.c new file mode 100644 index 000000000000..720d15e4b8ba --- /dev/null +++ b/sound/soc/sunxi/sun50i-dmic.c @@ -0,0 +1,405 @@ +// SPDX-License-Identifier: GPL-2.0-or-later +// +// This driver supports the DMIC in Allwinner's H6 SoCs. +// +// Copyright 2021 Ban Tao <fengzheng923(a)gmail.com> + +#include <linux/clk.h> +#include <linux/device.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/pm_runtime.h> +#include <linux/reset.h> +#include <sound/dmaengine_pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#define SUN50I_DMIC_EN_CTL (0x00) + #define SUN50I_DMIC_EN_CTL_GLOBE BIT(8) + #define SUN50I_DMIC_EN_CTL_CHAN(v) ((v) << 0) + #define SUN50I_DMIC_EN_CTL_CHAN_MASK GENMASK(7, 0) +#define SUN50I_DMIC_SR (0x04) + #define SUN50I_DMIC_SR_SAMOLE_RATE(v) ((v) << 0) + #define SUN50I_DMIC_SR_SAMOLE_RATE_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CTL (0x08) + #define SUN50I_DMIC_CTL_OVERSAMPLE_RATE BIT(0) +#define SUN50I_DMIC_DATA (0x10) +#define SUN50I_DMIC_INTC (0x14) + #define SUN50I_DMIC_FIFO_DRQ_EN BIT(2) +#define SUN50I_DMIC_INT_STA (0x18) + #define SUN50I_DMIC_INT_STA_OVERRUN_IRQ_PENDING BIT(1) + #define SUN50I_DMIC_INT_STA_DATA_IRQ_PENDING BIT(0) +#define SUN50I_DMIC_RXFIFO_CTL (0x1c) + #define SUN50I_DMIC_RXFIFO_CTL_FLUSH BIT(31) + #define SUN50I_DMIC_RXFIFO_CTL_MODE BIT(9) + #define SUN50I_DMIC_RXFIFO_CTL_RESOLUTION BIT(8) +#define SUN50I_DMIC_CH_NUM (0x24) + #define SUN50I_DMIC_CH_NUM_N(v) ((v) << 0) + #define SUN50I_DMIC_CH_NUM_N_MASK GENMASK(2, 0) +#define SUN50I_DMIC_CNT (0x2c) + #define SUN50I_DMIC_CNT_N BIT(0) +#define SUN50I_DMIC_HPF_CTRL (0x38) +#define SUN50I_DMIC_VERSION (0x50) + + +struct sun50i_dmic_dev { + struct clk *dmic_clk; + struct clk *bus_clk; + struct reset_control *rst; + struct regmap *regmap; + struct snd_dmaengine_dai_dma_data dma_params_rx; + unsigned int chan_en; +}; + +struct dmic_rate { + unsigned int samplerate; + unsigned int rate_bit; +}; + +static int sun50i_dmic_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0)); + + /* only support capture */ + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_FLUSH, + SUN50I_DMIC_RXFIFO_CTL_FLUSH); + regmap_write(host->regmap, SUN50I_DMIC_CNT, SUN50I_DMIC_CNT_N); + + return 0; +} + +static int sun50i_dmic_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + int i = 0; + unsigned long rate = params_rate(params); + unsigned int mclk = 0; + unsigned int channels = params_channels(params); + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(cpu_dai); + struct dmic_rate dmic_rate_s[] = { + {44100, 0x0}, + {48000, 0x0}, + {22050, 0x2}, + {24000, 0x2}, + {11025, 0x4}, + {12000, 0x4}, + {32000, 0x1}, + {16000, 0x3}, + {8000, 0x5}, + }; + + /* DMIC num is N+1 */ + regmap_update_bits(host->regmap, SUN50I_DMIC_CH_NUM, + SUN50I_DMIC_CH_NUM_N_MASK, + SUN50I_DMIC_CH_NUM_N(channels - 1)); + host->chan_en = (1 << channels) - 1; + regmap_write(host->regmap, SUN50I_DMIC_HPF_CTRL, host->chan_en); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, + SUN50I_DMIC_RXFIFO_CTL_MODE); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, 0); + break; + case SNDRV_PCM_FORMAT_S24_LE: + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_MODE, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_RXFIFO_CTL, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION, + SUN50I_DMIC_RXFIFO_CTL_RESOLUTION); + break; + default: + dev_err(cpu_dai->dev, "Invalid format!\n"); + return -EINVAL; + } + + switch (rate) { + case 11025: + case 22050: + case 44100: + mclk = 22579200; + break; + case 8000: + case 12000: + case 16000: + case 24000: + case 32000: + case 48000: + mclk = 24576000; + break; + default: + dev_err(cpu_dai->dev, "Invalid rate!\n"); + return -EINVAL; + } + + if (clk_set_rate(host->dmic_clk, mclk)) { + dev_err(cpu_dai->dev, "mclk : %u not support\n", mclk); + return -EINVAL; + } + + for (i = 0; i < ARRAY_SIZE(dmic_rate_s); i++) { + if (dmic_rate_s[i].samplerate == rate) { + regmap_update_bits(host->regmap, SUN50I_DMIC_SR, + SUN50I_DMIC_SR_SAMOLE_RATE_MASK, + SUN50I_DMIC_SR_SAMOLE_RATE(dmic_rate_s[i].rate_bit)); + break; + } + } + + switch (params_physical_width(params)) { + case 16: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + break; + case 32: + host->dma_params_rx.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + break; + default: + dev_err(cpu_dai->dev, "Unsupported physical sample width: %d\n", + params_physical_width(params)); + return -EINVAL; + } + + /* oversamplerate adjust */ + if (params_rate(params) >= 24000) + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE); + else + regmap_update_bits(host->regmap, SUN50I_DMIC_CTL, + SUN50I_DMIC_CTL_OVERSAMPLE_RATE, 0); + + return 0; +} + +static int sun50i_dmic_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + int ret = 0; + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + if (substream->stream != SNDRV_PCM_STREAM_CAPTURE) + return -EINVAL; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + /* DRQ ENABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, + SUN50I_DMIC_FIFO_DRQ_EN); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(host->chan_en)); + /* Global enable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, + SUN50I_DMIC_EN_CTL_GLOBE); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + /* DRQ DISABLE */ + regmap_update_bits(host->regmap, SUN50I_DMIC_INTC, + SUN50I_DMIC_FIFO_DRQ_EN, 0); + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_CHAN_MASK, + SUN50I_DMIC_EN_CTL_CHAN(0)); + /* Global disable */ + regmap_update_bits(host->regmap, SUN50I_DMIC_EN_CTL, + SUN50I_DMIC_EN_CTL_GLOBE, 0); + break; + + default: + ret = -EINVAL; + break; + } + return ret; +} + +static int sun50i_dmic_soc_dai_probe(struct snd_soc_dai *dai) +{ + struct sun50i_dmic_dev *host = snd_soc_dai_get_drvdata(dai); + + snd_soc_dai_init_dma_data(dai, NULL, &host->dma_params_rx); + + return 0; +} + +static const struct snd_soc_dai_ops sun50i_dmic_dai_ops = { + .startup = sun50i_dmic_startup, + .trigger = sun50i_dmic_trigger, + .hw_params = sun50i_dmic_hw_params, +}; + +static const struct regmap_config sun50i_dmic_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN50I_DMIC_VERSION, + .cache_type = REGCACHE_NONE, +}; + +#define SUN50I_DMIC_RATES (SNDRV_PCM_RATE_8000_48000) +#define SUN50I_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver sun50i_dmic_dai = { + .capture = { + .channels_min = 1, + .channels_max = 8, + .rates = SUN50I_DMIC_RATES, + .formats = SUN50I_FORMATS, + .sig_bits = 21, + }, + .probe = sun50i_dmic_soc_dai_probe, + .ops = &sun50i_dmic_dai_ops, + .name = "dmic", +}; + +static const struct of_device_id sun50i_dmic_of_match[] = { + { + .compatible = "allwinner,sun50i-h6-dmic", + }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, sun50i_dmic_of_match); + +static const struct snd_soc_component_driver sun50i_dmic_component = { + .name = "sun50i-dmic", +}; + +static int sun50i_dmic_runtime_suspend(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + + clk_disable_unprepare(host->dmic_clk); + clk_disable_unprepare(host->bus_clk); + + return 0; +} + +static int sun50i_dmic_runtime_resume(struct device *dev) +{ + struct sun50i_dmic_dev *host = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(host->dmic_clk); + if (ret) + return ret; + ret = clk_prepare_enable(host->bus_clk); + if (ret) + clk_disable_unprepare(host->dmic_clk); + + return ret; +} + +static int sun50i_dmic_probe(struct platform_device *pdev) +{ + struct sun50i_dmic_dev *host; + struct resource *res; + int ret; + void __iomem *base; + + host = devm_kzalloc(&pdev->dev, sizeof(*host), GFP_KERNEL); + if (!host) + return -ENOMEM; + + /* Get the addresses */ + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) + return dev_err_probe(&pdev->dev, PTR_ERR(base), + "get resource failed.\n"); + + host->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun50i_dmic_regmap_config); + + /* Clocks */ + host->bus_clk = devm_clk_get(&pdev->dev, "bus"); + if (IS_ERR(host->bus_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->bus_clk), + "failed to get bus clock.\n"); + + host->dmic_clk = devm_clk_get(&pdev->dev, "mod"); + if (IS_ERR(host->dmic_clk)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->dmic_clk), + "failed to get dmic clock.\n"); + + host->dma_params_rx.addr = res->start + SUN50I_DMIC_DATA; + host->dma_params_rx.maxburst = 8; + + platform_set_drvdata(pdev, host); + + host->rst = devm_reset_control_get_optional_exclusive(&pdev->dev, NULL); + if (IS_ERR(host->rst)) + return dev_err_probe(&pdev->dev, PTR_ERR(host->rst), + "Failed to get reset.\n"); + reset_control_deassert(host->rst); + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun50i_dmic_component, &sun50i_dmic_dai, 1); + if (ret) + return dev_err_probe(&pdev->dev, ret, + "failed to register component.\n"); + + pm_runtime_enable(&pdev->dev); + if (!pm_runtime_enabled(&pdev->dev)) { + ret = sun50i_dmic_runtime_resume(&pdev->dev); + if (ret) + goto err_unregister; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) + goto err_suspend; + + return 0; +err_suspend: + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); +err_unregister: + pm_runtime_disable(&pdev->dev); + return ret; +} + +static int sun50i_dmic_remove(struct platform_device *pdev) +{ + pm_runtime_disable(&pdev->dev); + if (!pm_runtime_status_suspended(&pdev->dev)) + sun50i_dmic_runtime_suspend(&pdev->dev); + + return 0; +} + +static const struct dev_pm_ops sun50i_dmic_pm = { + SET_RUNTIME_PM_OPS(sun50i_dmic_runtime_suspend, + sun50i_dmic_runtime_resume, NULL) +}; + +static struct platform_driver sun50i_dmic_driver = { + .driver = { + .name = "sun50i-dmic", + .of_match_table = of_match_ptr(sun50i_dmic_of_match), + .pm = &sun50i_dmic_pm, + }, + .probe = sun50i_dmic_probe, + .remove = sun50i_dmic_remove, +}; + +module_platform_driver(sun50i_dmic_driver); + +MODULE_DESCRIPTION("Allwinner sun50i DMIC SoC Interface"); +MODULE_AUTHOR("Ban Tao <fengzheng923(a)gmail.com>"); +MODULE_LICENSE("GPL"); +MODULE_ALIAS("platform:sun50i-dmic"); -- 2.17.1

3 2

[PATCH AUTOSEL 4.4 21/23] ALSA: hda: Add IRQ check for platform_get_irq()
by Sasha Levin 10 Jul '21

10 Jul '21

From: Jiajun Cao <jjcao20(a)fudan.edu.cn> [ Upstream commit 8c13212443230d03ff25014514ec0d53498c0912 ] The function hda_tegra_first_init() neglects to check the return value after executing platform_get_irq(). hda_tegra_first_init() should check the return value (if negative error number) for errors so as to not pass a negative value to the devm_request_irq(). Fix it by adding a check for the return value irq_id. Signed-off-by: Jiajun Cao <jjcao20(a)fudan.edu.cn> Signed-off-by: Xin Tan <tanxin.ctf(a)gmail.com> Reviewed-by: Thierry Reding <treding(a)nvidia.com> Link: https://lore.kernel.org/r/20210622131947.94346-1-jjcao20@fudan.edu.cn Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/pci/hda/hda_tegra.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 039fbbb1e53c..89359a962e47 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -363,6 +363,9 @@ static int hda_tegra_first_init(struct azx *chip, struct platform_device *pdev) unsigned short gcap; int irq_id = platform_get_irq(pdev, 0); + if (irq_id < 0) + return irq_id; + err = hda_tegra_init_chip(chip, pdev); if (err) return err; -- 2.30.2

1 0

[PATCH AUTOSEL 4.4 16/23] ALSA: bebob: add support for ToneWeal FW66
by Sasha Levin 10 Jul '21

10 Jul '21

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 50ebe56222bfa0911a932930f9229ee5995508d9 ] A user of FFADO project reported the issue of ToneWeal FW66. As a result, the device is identified as one of applications of BeBoB solution. I note that in the report the device returns contradictory result in plug discovery process for audio subunit. Fortunately ALSA BeBoB driver doesn't perform it thus it's likely to handle the device without issues. I receive no reaction to test request for this patch yet, however it would be worth to add support for it. daniel@gibbonmoon:/sys/bus/firewire/devices/fw1$ grep -r . * Binary file config_rom matches dev:244:1 guid:0x0023270002000000 hardware_version:0x000002 is_local:0 model:0x020002 model_name:FW66 power/runtime_active_time:0 power/runtime_active_kids:0 power/runtime_usage:0 power/runtime_status:unsupported power/async:disabled power/runtime_suspended_time:0 power/runtime_enabled:disabled power/control:auto subsystem/drivers_autoprobe:1 uevent:MAJOR=244 uevent:MINOR=1 uevent:DEVNAME=fw1 units:0x00a02d:0x010001 vendor:0x002327 vendor_name:ToneWeal fw1.0/uevent:MODALIAS=ieee1394:ven00002327mo00020002sp0000A02Dver00010001 fw1.0/power/runtime_active_time:0 fw1.0/power/runtime_active_kids:0 fw1.0/power/runtime_usage:0 fw1.0/power/runtime_status:unsupported fw1.0/power/async:disabled fw1.0/power/runtime_suspended_time:0 fw1.0/power/runtime_enabled:disabled fw1.0/power/control:auto fw1.0/model:0x020002 fw1.0/rom_index:15 fw1.0/specifier_id:0x00a02d fw1.0/model_name:FW66 fw1.0/version:0x010001 fw1.0/modalias:ieee1394:ven00002327mo00020002sp0000A02Dver00010001 Cc: Daniel Jozsef <daniel.jozsef(a)gmail.com> Reference: https://lore.kernel.org/alsa-devel/20200119164335.GA11974@workstation/ Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20210619083922.16060-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/Kconfig | 1 + sound/firewire/bebob/bebob.c | 3 +++ 2 files changed, 4 insertions(+) diff --git a/sound/firewire/Kconfig b/sound/firewire/Kconfig index 4199cfc4a96a..850315d1abca 100644 --- a/sound/firewire/Kconfig +++ b/sound/firewire/Kconfig @@ -117,6 +117,7 @@ config SND_BEBOB * M-Audio Ozonic/NRV10/ProfireLightBridge * M-Audio FireWire 1814/ProjectMix IO * Digidesign Mbox 2 Pro + * ToneWeal FW66 To compile this driver as a module, choose M here: the module will be called snd-bebob. diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index 64dca7931272..c3c14e383e73 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -60,6 +60,7 @@ static DECLARE_BITMAP(devices_used, SNDRV_CARDS); #define VEN_MAUDIO1 0x00000d6c #define VEN_MAUDIO2 0x000007f5 #define VEN_DIGIDESIGN 0x00a07e +#define OUI_SHOUYO 0x002327 #define MODEL_FOCUSRITE_SAFFIRE_BOTH 0x00000000 #define MODEL_MAUDIO_AUDIOPHILE_BOTH 0x00010060 @@ -461,6 +462,8 @@ static const struct ieee1394_device_id bebob_id_table[] = { &maudio_special_spec), /* Digidesign Mbox 2 Pro */ SND_BEBOB_DEV_ENTRY(VEN_DIGIDESIGN, 0x0000a9, &spec_normal), + // Toneweal FW66. + SND_BEBOB_DEV_ENTRY(OUI_SHOUYO, 0x020002, &spec_normal), /* IDs are unknown but able to be supported */ /* Apogee, Mini-ME Firewire */ /* Apogee, Mini-DAC Firewire */ -- 2.30.2

1 0

[PATCH AUTOSEL 4.4 15/23] ASoC: soc-core: Fix the error return code in snd_soc_of_parse_audio_routing()
by Sasha Levin 10 Jul '21

10 Jul '21

From: Zhen Lei <thunder.leizhen(a)huawei.com> [ Upstream commit 7d3865a10b9ff2669c531d5ddd60bf46b3d48f1e ] When devm_kcalloc() fails, the error code -ENOMEM should be returned instead of -EINVAL. Signed-off-by: Zhen Lei <thunder.leizhen(a)huawei.com> Link: https://lore.kernel.org/r/20210617103729.1918-1-thunder.leizhen@huawei.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b927f9c81d92..e69a7f8b6163 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3394,7 +3394,7 @@ int snd_soc_of_parse_audio_routing(struct snd_soc_card *card, if (!routes) { dev_err(card->dev, "ASoC: Could not allocate DAPM route table\n"); - return -EINVAL; + return -ENOMEM; } for (i = 0; i < num_routes; i++) { -- 2.30.2

1 0

[PATCH AUTOSEL 4.4 13/23] ALSA: ppc: fix error return code in snd_pmac_probe()
by Sasha Levin 10 Jul '21

10 Jul '21

From: Yang Yingliang <yangyingliang(a)huawei.com> [ Upstream commit 80b9c1be567c3c6bbe0d4b290af578e630485b5d ] If snd_pmac_tumbler_init() or snd_pmac_tumbler_post_init() fails, snd_pmac_probe() need return error code. Reported-by: Hulk Robot <hulkci(a)huawei.com> Signed-off-by: Yang Yingliang <yangyingliang(a)huawei.com> Link: https://lore.kernel.org/r/20210616021121.1991502-1-yangyingliang@huawei.com Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/ppc/powermac.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/ppc/powermac.c b/sound/ppc/powermac.c index 33c6be9fb388..7c70ba5e2540 100644 --- a/sound/ppc/powermac.c +++ b/sound/ppc/powermac.c @@ -90,7 +90,11 @@ static int snd_pmac_probe(struct platform_device *devptr) sprintf(card->shortname, "PowerMac %s", name_ext); sprintf(card->longname, "%s (Dev %d) Sub-frame %d", card->shortname, chip->device_id, chip->subframe); - if ( snd_pmac_tumbler_init(chip) < 0 || snd_pmac_tumbler_post_init() < 0) + err = snd_pmac_tumbler_init(chip); + if (err < 0) + goto __error; + err = snd_pmac_tumbler_post_init(); + if (err < 0) goto __error; break; case PMAC_AWACS: -- 2.30.2

1 0

[PATCH AUTOSEL 4.4 10/23] ALSA: sb: Fix potential double-free of CSP mixer elements
by Sasha Levin 10 Jul '21

10 Jul '21

From: Takashi Iwai <tiwai(a)suse.de> [ Upstream commit c305366a37441c2ac90b08711cb6f032b43672f2 ] snd_sb_qsound_destroy() contains the calls of removing the previously created mixer controls, but it doesn't clear the pointers. As snd_sb_qsound_destroy() itself may be repeatedly called via ioctl, this could lead to double-free potentially. Fix it by clearing the struct fields properly afterwards. Link: https://lore.kernel.org/r/20210608140540.17885-4-tiwai@suse.de Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/isa/sb/sb16_csp.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/isa/sb/sb16_csp.c b/sound/isa/sb/sb16_csp.c index 2cc068be7d3b..90fa57ad14c0 100644 --- a/sound/isa/sb/sb16_csp.c +++ b/sound/isa/sb/sb16_csp.c @@ -1086,10 +1086,14 @@ static void snd_sb_qsound_destroy(struct snd_sb_csp * p) card = p->chip->card; down_write(&card->controls_rwsem); - if (p->qsound_switch) + if (p->qsound_switch) { snd_ctl_remove(card, p->qsound_switch); - if (p->qsound_space) + p->qsound_switch = NULL; + } + if (p->qsound_space) { snd_ctl_remove(card, p->qsound_space); + p->qsound_space = NULL; + } up_write(&card->controls_rwsem); /* cancel pending transfer of QSound parameters */ -- 2.30.2

1 0