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

[alsa-devel] [PATCH] ASoC: SOF: Intel: hda: Fix COMPILE_TEST build error
by YueHaibing 04 Jun '19

04 Jun '19

while building without PCI: sound/soc/sof/intel/hda.o: In function `hda_dsp_probe': hda.c:(.text+0x79c): undefined reference to `pci_ioremap_bar' hda.c:(.text+0x79c): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `pci_ioremap_bar' hda.c:(.text+0x7c4): undefined reference to `pci_ioremap_bar' hda.c:(.text+0x7c4): relocation truncated to fit: R_AARCH64_CALL26 against undefined symbol `pci_ioremap_bar' Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: e13ef82a9ab8 ("ASoC: SOF: add COMPILE_TEST for PCI options") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/sof/intel/hda.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 68db2ac..c1703c4 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -231,7 +231,9 @@ static int hda_init(struct snd_sof_dev *sdev) /* initialise hdac bus */ bus->addr = pci_resource_start(pci, 0); +#if IS_ENABLED(CONFIG_PCI) bus->remap_addr = pci_ioremap_bar(pci, 0); +#endif if (!bus->remap_addr) { dev_err(bus->dev, "error: ioremap error\n"); return -ENXIO; @@ -458,7 +460,9 @@ int hda_dsp_probe(struct snd_sof_dev *sdev) goto hdac_bus_unmap; /* DSP base */ +#if IS_ENABLED(CONFIG_PCI) sdev->bar[HDA_DSP_BAR] = pci_ioremap_bar(pci, HDA_DSP_BAR); +#endif if (!sdev->bar[HDA_DSP_BAR]) { dev_err(sdev->dev, "error: ioremap error\n"); ret = -ENXIO; -- 2.7.4

4 4

[alsa-devel] [PATCH] ASoC: pcm3168a: Implement set_tdm_slot callback
by Peter Ujfalusi 04 Jun '19

04 Jun '19

Initially we only going to care about the slot_width as for example DSP_A/B needs 32 bclk per slots and to be able to use TDM mode the codec (and CPU) needs to use DSP_A/B modes. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/pcm3168a.c | 60 ++++++++++++++++++++++++++++++++----- 1 file changed, 52 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/pcm3168a.c b/sound/soc/codecs/pcm3168a.c index e0d5839fe1a7..9eb24ca09793 100644 --- a/sound/soc/codecs/pcm3168a.c +++ b/sound/soc/codecs/pcm3168a.c @@ -56,6 +56,9 @@ struct pcm3168a_priv { unsigned long sysclk; unsigned int adc_fmt; unsigned int dac_fmt; + int tdm_slots; + u32 tdm_mask[2]; + int slot_width; }; static const char *const pcm3168a_roll_off[] = { "Sharp", "Slow" }; @@ -387,6 +390,41 @@ static int pcm3168a_set_dai_fmt_adc(struct snd_soc_dai *dai, return pcm3168a_set_dai_fmt(dai, format, false); } +static int pcm3168a_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, + int slot_width) +{ + struct snd_soc_component *component = dai->component; + struct pcm3168a_priv *pcm3168a = snd_soc_component_get_drvdata(component); + + if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) { + dev_err(component->dev, + "Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n", + tx_mask, rx_mask, slots); + return -EINVAL; + } + + if (slot_width && + (slot_width != 16 && slot_width != 24 && slot_width != 32 )) { + dev_err(component->dev, "Unsupported slot_width %d\n", + slot_width); + return -EINVAL; + } + + pcm3168a->tdm_slots = slots; + pcm3168a->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask; + pcm3168a->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask; + + if (pcm3168a->slot_width && pcm3168a->slot_width != slot_width) { + dev_err(component->dev, "Not matching slot_width %d vs %d\n", + pcm3168a->slot_width, slot_width); + return -EINVAL; + } + + pcm3168a->slot_width = slot_width; + return 0; +} + static int pcm3168a_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -431,22 +469,26 @@ static int pcm3168a_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - min_frame_size = params_width(params) * 2; + if (pcm3168a->slot_width) + min_frame_size = pcm3168a->slot_width; + else + min_frame_size = params_width(params); + switch (min_frame_size) { - case 32: + case 16: if (master_mode || (fmt != PCM3168A_FMT_RIGHT_J)) { - dev_err(component->dev, "32-bit frames are supported only for slave mode using right justified\n"); + dev_err(component->dev, "16-bit slots are supported only for slave mode using right justified\n"); return -EINVAL; } fmt = PCM3168A_FMT_RIGHT_J_16; break; - case 48: + case 24: if (master_mode || (fmt & PCM3168A_FMT_DSP_MASK)) { - dev_err(component->dev, "48-bit frames not supported in master mode, or slave mode using DSP\n"); + dev_err(component->dev, "24-bit slots not supported in master mode, or slave mode using DSP\n"); return -EINVAL; } break; - case 64: + case 32: break; default: dev_err(component->dev, "unsupported frame size: %d\n", min_frame_size); @@ -554,14 +596,16 @@ static const struct snd_soc_dai_ops pcm3168a_dac_dai_ops = { .set_fmt = pcm3168a_set_dai_fmt_dac, .set_sysclk = pcm3168a_set_dai_sysclk, .hw_params = pcm3168a_hw_params, - .digital_mute = pcm3168a_digital_mute + .digital_mute = pcm3168a_digital_mute, + .set_tdm_slot = pcm3168a_set_tdm_slot, }; static const struct snd_soc_dai_ops pcm3168a_adc_dai_ops = { .startup = pcm3168a_startup, .set_fmt = pcm3168a_set_dai_fmt_adc, .set_sysclk = pcm3168a_set_dai_sysclk, - .hw_params = pcm3168a_hw_params + .hw_params = pcm3168a_hw_params, + .set_tdm_slot = pcm3168a_set_tdm_slot, }; static struct snd_soc_dai_driver pcm3168a_dais[] = { -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 1

[alsa-devel] [PATCH] ASoC: da7219: Fix build error without CONFIG_I2C
by YueHaibing 04 Jun '19

04 Jun '19

Fix gcc build error while CONFIG_I2C is not set sound/soc/codecs/da7219.c:2640:1: warning: data definition has no type or storage class module_i2c_driver(da7219_i2c_driver); ^~~~~~~~~~~~~~~~~ sound/soc/codecs/da7219.c:2640:1: error: type defaults to int in declaration of module_i2c_driver [-Werror=implicit-int] sound/soc/codecs/da7219.c:2640:1: warning: parameter names (without types) in function declaration sound/soc/codecs/da7219.c:2629:26: warning: da7219_i2c_driver defined but not used [-Wunused-variable] Reported-by: Hulk Robot <hulkci(a)huawei.com> Fixes: 6d817c0e9fd7 ("ASoC: codecs: Add da7219 codec driver") Signed-off-by: YueHaibing <yuehaibing(a)huawei.com> --- sound/soc/mediatek/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig index 933ab51..111e44b 100644 --- a/sound/soc/mediatek/Kconfig +++ b/sound/soc/mediatek/Kconfig @@ -133,7 +133,7 @@ config SND_SOC_MT8183_MT6358_TS3A227E_MAX98357A config SND_SOC_MT8183_DA7219_MAX98357A tristate "ASoC Audio driver for MT8183 with DA7219 MAX98357A codec" - depends on SND_SOC_MT8183 + depends on SND_SOC_MT8183 && I2C select SND_SOC_MT6358 select SND_SOC_MAX98357A select SND_SOC_DA7219 -- 2.7.4

2 1

[alsa-devel] [PATCH v2 20/22] dt: bindings: fix some broken links from txt->yaml conversion
by Mauro Carvalho Chehab 04 Jun '19

04 Jun '19

Some new files got converted to yaml, but references weren't updated accordingly. Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> --- Documentation/devicetree/bindings/media/st,stm32-dcmi.txt | 2 +- Documentation/devicetree/bindings/sound/st,stm32-i2s.txt | 2 +- Documentation/devicetree/bindings/sound/st,stm32-sai.txt | 2 +- MAINTAINERS | 2 +- 4 files changed, 4 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/media/st,stm32-dcmi.txt b/Documentation/devicetree/bindings/media/st,stm32-dcmi.txt index 249790a93017..3122ded82eb4 100644 --- a/Documentation/devicetree/bindings/media/st,stm32-dcmi.txt +++ b/Documentation/devicetree/bindings/media/st,stm32-dcmi.txt @@ -11,7 +11,7 @@ Required properties: - clock-names: must contain "mclk", which is the DCMI peripherial clock - pinctrl: the pincontrol settings to configure muxing properly for pins that connect to DCMI device. - See Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt. + See Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml. - dmas: phandle to DMA controller node, see Documentation/devicetree/bindings/dma/stm32-dma.txt - dma-names: must contain "tx", which is the transmit channel from DCMI to DMA diff --git a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt index 58c341300552..cbf24bcd1b8d 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-i2s.txt @@ -18,7 +18,7 @@ Required properties: See Documentation/devicetree/bindings/dma/stm32-dma.txt. - dma-names: Identifier for each DMA request line. Must be "tx" and "rx". - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml Optional properties: - resets: Reference to a reset controller asserting the reset controller diff --git a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt index 3f4467ff0aa2..944743dd9212 100644 --- a/Documentation/devicetree/bindings/sound/st,stm32-sai.txt +++ b/Documentation/devicetree/bindings/sound/st,stm32-sai.txt @@ -41,7 +41,7 @@ SAI subnodes required properties: "tx": if sai sub-block is configured as playback DAI "rx": if sai sub-block is configured as capture DAI - pinctrl-names: should contain only value "default" - - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.txt + - pinctrl-0: see Documentation/devicetree/bindings/pinctrl/st,stm32-pinctrl.yaml SAI subnodes Optional properties: - st,sync: specify synchronization mode. diff --git a/MAINTAINERS b/MAINTAINERS index 0dc7c3c5ddb0..2ab2337a029c 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -1298,7 +1298,7 @@ ARM PRIMECELL SSP PL022 SPI DRIVER M: Linus Walleij <linus.walleij(a)linaro.org> L: linux-arm-kernel(a)lists.infradead.org (moderated for non-subscribers) S: Maintained -F: Documentation/devicetree/bindings/spi/spi_pl022.txt +F: Documentation/devicetree/bindings/spi/spi-pl022.yaml F: drivers/spi/spi-pl022.c ARM PRIMECELL UART PL010 AND PL011 DRIVERS -- 2.21.0

1 0

[alsa-devel] [PATCH v3] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 04 Jun '19

04 Jun '19

The supported formats are S16_LE and S24_LE now. However, by datasheet of max98090, S24_LE is only supported when it is in the right justified mode. We should remove 24-bit format if it is not in that mode to avoid triggering error. Change-Id: I304882e0b67974f8fe4c2a47c61a41a04635b2df Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- Fix compile error. codec->dai => dai runtime => substream->runtime sound/soc/codecs/max98090.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..ada8c25e643d 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* Remove 24-bit format support if it is not in right justified mode. */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { + substream->runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + } + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2331,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.rc1.311.g5d7573a151-goog

1 0

[alsa-devel] [PATCH v2] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 04 Jun '19

04 Jun '19

The supported formats are S16_LE and S24_LE now. However, by datasheet of max98090, S24_LE is only supported when it is in the right justified mode. We should remove 24-bit format if it is not in that mode to avoid triggering error. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/max98090.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..a6c2cb89767c 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = codec_dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* Remove 24-bit format support if it is not in right justified mode. */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + } + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2331,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.rc1.311.g5d7573a151-goog

1 0

[alsa-devel] [PATCH] ASoC: max98090: remove 24-bit format support if RJ is 0
by Yu-Hsuan Hsu 04 Jun '19

04 Jun '19

The supported formats are S16_LE and S24_LE now. However, by datasheet of max98090, S24_LE is only supported when it is in the right justified mode. We should remove 24-bit format if it is not in that mode to avoid triggering error. Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/max98090.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..4cbfb0b95404 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1909,6 +1909,21 @@ static int max98090_configure_dmic(struct max98090_priv *max98090, return 0; } +static int max98090_dai_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = codec_dai->component; + struct max98090_priv *max98090 = snd_soc_component_get_drvdata(component); + unsigned int fmt = max98090->dai_fmt; + + /* Remove 24-bit format support if RJ is 1. */ + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) != SND_SOC_DAIFMT_RIGHT_J) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(substream->runtime, 0, 16, 16); + } + return 0; +} + static int max98090_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) @@ -2316,6 +2331,7 @@ EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) static const struct snd_soc_dai_ops max98090_dai_ops = { + .startup = max98090_dai_startup, .set_sysclk = max98090_dai_set_sysclk, .set_fmt = max98090_dai_set_fmt, .set_tdm_slot = max98090_set_tdm_slot, -- 2.22.0.rc1.311.g5d7573a151-goog

1 0

[alsa-devel] [PATCH 0/5] Assorted fixes discovered with gcc 4.1
by Geert Uytterhoeven 04 Jun '19

04 Jun '19

Hi all, Ever since commit cafa0010cd51fb71 ("Raise the minimum required gcc version to 4.6"), I felt bored when looking at my test build logs, as I was no longer discovering many real issues. Hence I started wondering if the modern gcc versions are really catching these classes of bugs caught before with gcc 4.1, or if they just go undetected. I reverted some changes and applied some fixes, which allowed me to compile most of the kernel with gcc 4.1 again. I built an m68k/allmodconfig kernel, looked at all new warnings, and fixed the ones that are not false positives. The result is a patch series of 5 patches, of which one or two fix real bugs. Thanks for your comments, and for applying where appropriate! Geert Uytterhoeven (5): lightnvm: Fix uninitialized pointer in nvm_remove_tgt() rxrpc: Fix uninitialized error code in rxrpc_send_data_packet() net: sched: pie: Use ULL suffix for 64-bit constant ALSA: fireface: Use ULL suffixes for 64-bit constants [RFC] devlink: Fix uninitialized error code in devlink_fmsg_prepare_skb() drivers/lightnvm/core.c | 2 +- net/core/devlink.c | 2 +- net/rxrpc/output.c | 4 +++- net/sched/sch_pie.c | 2 +- sound/firewire/fireface/ff-protocol-latter.c | 10 +++++----- 5 files changed, 11 insertions(+), 9 deletions(-) -- 2.17.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

9 19

Re: [alsa-devel] atmel SSC DSP A mode
by Codrin.Ciubotariu＠microchip.com 04 Jun '19

04 Jun '19

On 03.06.2019 13:19, Willem Romkes wrote: > > * > * > > Hi, > > We are using linux-at91 and lately I have been working on adding support > for a custom soundcard, a dual tlv320aic3x codec on SAMA5D3 platform. > This soundcard uses DSP A mode. > > We are facing audio issues: When we playback a WAV file with a sine > wave, the codec outputs far from a clean sine wave. The problem I think > is that the CKI settings for the TMCR register isn't set correctly. > see commit 20cf2603b122bf71fb54def1de6a2ad73d5ddb0b > Author: Bo Shen <voice.shen(a)atmel.com> > Date: Fri Jan 30 17:38:42 2015 +0800 > > ASoC: atmel_ssc_dai: fix the setting for DSP mode > > When SCC work in DSP A mode, the data outputs/inputs are shift out on > falling edge, the frame sync are sample on the rising edge. > > Reported-by: Songjun Wu <songjun.wu(a)atmel.com> > Signed-off-by: Bo Shen <voice.shen(a)atmel.com> > Signed-off-by: Mark Brown <broonie(a)kernel.org> > > diff --git a/sound/soc/atmel/atmel_ssc_dai.c > b/sound/soc/atmel/atmel_ssc_dai.c > index e691aab..35e44e4 100644 > --- a/sound/soc/atmel/atmel_ssc_dai.c > +++ b/sound/soc/atmel/atmel_ssc_dai.c > @@ -502,7 +502,7 @@ static int atmel_ssc_hw_params(struct > snd_pcm_substream *substream, > rcmr = SSC_BF(RCMR_PERIOD, ssc_p->rcmr_period) > | SSC_BF(RCMR_STTDLY, 1) > | SSC_BF(RCMR_START, SSC_START_RISING_RF) > - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) > + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) > | SSC_BF(RCMR_CKO, SSC_CKO_NONE) > | SSC_BF(RCMR_CKS, SSC_CKS_DIV); > > @@ -517,7 +517,7 @@ static int atmel_ssc_hw_params(struct > snd_pcm_substream *substream, > tcmr = SSC_BF(TCMR_PERIOD, ssc_p->tcmr_period) > | SSC_BF(TCMR_STTDLY, 1) > | SSC_BF(TCMR_START, SSC_START_RISING_RF) > - | SSC_BF(TCMR_CKI, SSC_CKI_RISING) > + | SSC_BF(TCMR_CKI, SSC_CKI_FALLING) > | SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS) > | SSC_BF(TCMR_CKS, SSC_CKS_DIV); > > @@ -546,7 +546,7 @@ static int atmel_ssc_hw_params(struct > snd_pcm_substream *substream, > rcmr = SSC_BF(RCMR_PERIOD, 0) > | SSC_BF(RCMR_STTDLY, START_DELAY) > | SSC_BF(RCMR_START, SSC_START_RISING_RF) > - | SSC_BF(RCMR_CKI, SSC_CKI_RISING) > + | SSC_BF(RCMR_CKI, SSC_CKI_FALLING) > | SSC_BF(RCMR_CKO, SSC_CKO_NONE) > | SSC_BF(RCMR_CKS, ssc->clk_from_rk_pin ? > SSC_CKS_PIN : SSC_CKS_CLOCK); > > For both the TCMR and RCMR the CKI is set to SSC_CKI_FALLING, but I > think for the TCMR the CKI should be set to SSC_CKI_RISING. See datasheet: > > TCMR CKI: "1: The data outputs (Data and Frame Sync signals) are shifted > out on Transmit Clock rising edge. The Frame sync signal > input is sampled on Transmit clock falling edge." > > RCMR CKI: "0: The data inputs (Data and Frame Sync signals) are sampled > on Receive Clock falling edge. The Frame Sync signal out- > put is shifted out on Receive Clock rising edge." > > Setting the CKI in TCMR to SSC_CKI_RISING to fixes our issues. Hi Willem, It this DS of the codec you are using? http://www.ti.com/lit/ds/symlink/tlv320aic33.pdf Are you using the DSP mode described in figure 25 DSP Serial Bus Mode Operation? If yes, then you should use the DSP_B mode... Thanks and best regards, Codrin

2 1

[alsa-devel] [PATCH] ASoC: core: move DAI pre-links initiation to snd_soc_instantiate_card
by Tzung-Bi Shih 04 Jun '19

04 Jun '19

Kernel crashes when an ASoC component rebinding. As an example, by using the following commands: - echo -n max98357a > /sys/bus/platform/drivers/max98357a/unbind - echo -n max98357a > /sys/bus/platform/drivers/max98357a/bind Got the error message: "Unable to handle kernel NULL pointer dereference at virtual address". The call trace: snd_soc_is_matching_component+0x30/0x6c soc_bind_dai_link+0x16c/0x240 snd_soc_bind_card+0x1e4/0xb10 snd_soc_add_component+0x270/0x300 snd_soc_register_component+0x54/0x6c devm_snd_soc_register_component+0x68/0xac max98357a_platform_probe+0x7c/0x94 The dai_link->platforms has been reset to NULL by soc_cleanup_platform() in soc_cleanup_card_resources() when un-registering component. However, it has no chance to re-allocate the dai_link->platforms when registering the component again. Move the DAI pre-links initiation from snd_soc_register_card() to snd_soc_instantiate_card() to make sure all DAI pre-links get initiated when component rebinding. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/soc-core.c | 28 ++++++++++------------------ 1 file changed, 10 insertions(+), 18 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2d3520fca613..82ff384753c7 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2072,6 +2072,15 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + for_each_card_prelinks(card, i, dai_link) { + ret = soc_init_dai_link(card, dai_link); + if (ret) { + dev_err(card->dev, "ASoC: failed to init link %s: %d\n", + dai_link->name, ret); + goto probe_end; + } + } + card->dapm.bias_level = SND_SOC_BIAS_OFF; card->dapm.dev = card->dev; card->dapm.card = card; @@ -2241,7 +2250,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) snd_soc_dapm_sync(&card->dapm); probe_end: - if (ret < 0) + if (ret < 0 && ret != -EPROBE_DEFER) soc_cleanup_card_resources(card); mutex_unlock(&card->mutex); @@ -2794,26 +2803,9 @@ static int snd_soc_bind_card(struct snd_soc_card *card) */ int snd_soc_register_card(struct snd_soc_card *card) { - int i, ret; - struct snd_soc_dai_link *link; - if (!card->name || !card->dev) return -EINVAL; - mutex_lock(&client_mutex); - for_each_card_prelinks(card, i, link) { - - ret = soc_init_dai_link(card, link); - if (ret) { - soc_cleanup_platform(card); - dev_err(card->dev, "ASoC: failed to init link %s\n", - link->name); - mutex_unlock(&client_mutex); - return ret; - } - } - mutex_unlock(&client_mutex); - dev_set_drvdata(card->dev, card); snd_soc_initialize_card_lists(card); -- 2.22.0.rc1.257.g3120a18244-goog

2 3

[alsa-devel] [PATCH 0/5] ASoC: SOF: suspend/resume debug tools
by Pierre-Louis Bossart 03 Jun '19

03 Jun '19

No new functionality per se, just utilities for developers. The vast majority of current integration issues are related to HDAudio + trace support combined with runtime suspend/resume. Add a couple of changes to make it easier for developers to see the firmware logs across suspend/resume transitions, disable the trace and/or disable runtime_pm to help narrow down the problems. Kai Vehmanen (2): ASoC: SOF: trace: remove code duplication in sof_wait_trace_avail() ASoC: SOF: force end-of-file for debugfs trace at suspend Pierre-Louis Bossart (3): ASoC: SOF: trace: move to opt-in with Kconfig and module parameter ASoC: SOF: acpi: add module param to disable pm_runtime ASoC: SOF: pci: add module param to disable pm_runtime sound/soc/sof/Kconfig | 8 +++++ sound/soc/sof/core.c | 26 ++++++++++---- sound/soc/sof/sof-acpi-dev.c | 12 ++++++- sound/soc/sof/sof-pci-dev.c | 12 ++++++- sound/soc/sof/sof-priv.h | 3 ++ sound/soc/sof/trace.c | 67 ++++++++++++++++++++++++++++++------ 6 files changed, 110 insertions(+), 18 deletions(-) base-commit: 188d45fe779eeb8e3521b59fcb12cc48a6f2c203 -- 2.20.1

2 9

[alsa-devel] [PATCH 0/9] ASoC: SOF: new functionality with MINOR ABI changes
by Pierre-Louis Bossart 03 Jun '19

03 Jun '19

The first patches are to align with firmware, we unfortunately had a couple of misses before the new ABI change process was introduced [1]. The new functionality added after the initial cleanup is: 1. sending the kernel time to firmware so that the trace logs use the same base across suspend-resume cycles. 2. introducing an IPC flood mechanism which already helped harden the IPC sequences, be it with QEMU or on real hardware. 3. support for mux/demux component 4. initial SoundWire support This patchset is independent from the IPC changes submitted on alsa-devel, but will only apply if previous trace related changes are applied. ([PATCH 3/5] ASoC: SOF: trace: move to opt-in with Kconfig and module parameter) [1] https://github.com/orgs/thesofproject/projects/2 Bard liao (1): ASoC: SOF: send time stamp to FW for alignment Jaska Uimonen (1): ASoC: SOF: topology: add support for mux/demux component Kai Vehmanen (1): ASoC: SOF: fix DSP oops definitions in FW ABI Pan Xiuli (1): ASoC: SOF: soundwire: add initial soundwire support Pierre-Louis Bossart (2): ASoC: SOF: uapi: mirror firmware changes ASoC: SOF: bump to ABI 3.6 Ranjani Sridharan (3): ASoC: SOF: add mode parameter for snd_sof_debugfs_buf_item ASoC: SOF: ipc: Introduce SOF_IPC_GLB_TEST_MSG IPC command ASoC: SOF: debug: add new debugfs entries for IPC flood test include/sound/sof/dai.h | 1 + include/sound/sof/header.h | 30 ++++- include/sound/sof/topology.h | 3 + include/sound/sof/trace.h | 10 ++ include/sound/sof/xtensa.h | 9 +- include/uapi/sound/sof/abi.h | 2 +- sound/soc/sof/Kconfig | 8 ++ sound/soc/sof/debug.c | 247 ++++++++++++++++++++++++++++++++++- sound/soc/sof/intel/bdw.c | 17 ++- sound/soc/sof/intel/byt.c | 15 ++- sound/soc/sof/intel/hda.c | 16 ++- sound/soc/sof/ipc.c | 9 ++ sound/soc/sof/loader.c | 4 +- sound/soc/sof/sof-priv.h | 8 +- sound/soc/sof/topology.c | 2 + sound/soc/sof/trace.c | 16 ++- sound/soc/sof/xtensa/core.c | 2 +- 17 files changed, 366 insertions(+), 33 deletions(-) -- 2.20.1

2 19

[alsa-devel] [PATCH 0/8] ASoC: SOF: IPC updates
by Pierre-Louis Bossart 03 Jun '19

03 Jun '19

Align the Intel IPC code with the recommended hardware sequences instead of confusing ones, remove useless command and clean-up interface definition. Daniel Baluta (1): ASoC: SOF: Do not send cmd via SHIM register Pierre-Louis Bossart (6): ASoC: SOF: Intel: cnl-ipc: read all IPC registers first ASoC: SOF: Intel: cnl-ipc: move code around for clarity ASoC: SOF: Intel: cnl-ipc: re-enable IPC IRQ at end of handler ASoC: SOF: Intel: hda-ipc: read all IPC registers first ASoC: SOF: Intel: hda-ipc: re-enable IPC IRQ at end of handler ASoC: SOF: Intel: ipc: don't check for HIPCCTL register value Slawomir Blauciak (1): ASoC: SOF: ipc: replace fw ready bitfield with explicit bit ordering include/sound/sof/info.h | 20 ++++++------- sound/soc/sof/intel/byt.c | 5 +--- sound/soc/sof/intel/cnl.c | 56 +++++++++++++++++------------------ sound/soc/sof/intel/hda-ipc.c | 43 +++++++++++++-------------- sound/soc/sof/ipc.c | 11 ++++--- 5 files changed, 67 insertions(+), 68 deletions(-) -- 2.20.1

2 16

[alsa-devel] [PATCH v2] ASoC: hda: fix unbalanced codec dev refcount for HDA_DEV_ASOC
by Ranjani Sridharan 03 Jun '19

03 Jun '19

HDA_DEV_ASOC type codec device refcounts are managed differently from HDA_DEV_LEGACY devices. The refcount is released explicitly in snd_hdac_ext_bus_device_remove() for ASOC type devices. So, remove the put_device() call in snd_hda_codec_dev_free() for such devices to make the refcount balanced. This will prevent the NULL pointer exception when the codec driver is released after the card is freed. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b20eb7fc83eb..fcdf2cd3783b 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -840,7 +840,14 @@ static int snd_hda_codec_dev_free(struct snd_device *device) if (codec->core.type == HDA_DEV_LEGACY) snd_hdac_device_unregister(&codec->core); codec_display_power(codec, false); - put_device(hda_codec_dev(codec)); + + /* + * In the case of ASoC HD-audio bus, the device refcount is released in + * snd_hdac_ext_bus_device_remove() explicitly. + */ + if (codec->core.type == HDA_DEV_LEGACY) + put_device(hda_codec_dev(codec)); + return 0; } -- 2.17.1

3 2

[alsa-devel] [PATCH] ASoC: stm32: sai: manage identification registers
by Olivier Moysan 03 Jun '19

03 Jun '19

Add support of identification registers in STM32 SAI. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> --- sound/soc/stm/stm32_sai.c | 44 +++++++++++++++++++++++++++++++---- sound/soc/stm/stm32_sai.h | 54 ++++++++++++++++++++++++++++++++----------- sound/soc/stm/stm32_sai_sub.c | 14 ++++++----- 3 files changed, 88 insertions(+), 24 deletions(-) diff --git a/sound/soc/stm/stm32_sai.c b/sound/soc/stm/stm32_sai.c index 7550d5f08be3..98b29f712831 100644 --- a/sound/soc/stm/stm32_sai.c +++ b/sound/soc/stm/stm32_sai.c @@ -30,13 +30,20 @@ #include "stm32_sai.h" static const struct stm32_sai_conf stm32_sai_conf_f4 = { - .version = SAI_STM32F4, - .has_spdif = false, + .version = STM_SAI_STM32F4, + .fifo_size = 8, + .has_spdif_pdm = false, }; +/* + * Default settings for stm32 H7 socs and next. + * These default settings will be overridden if the soc provides + * support of hardware configuration registers. + */ static const struct stm32_sai_conf stm32_sai_conf_h7 = { - .version = SAI_STM32H7, - .has_spdif = true, + .version = STM_SAI_STM32H7, + .fifo_size = 8, + .has_spdif_pdm = true, }; static const struct of_device_id stm32_sai_ids[] = { @@ -157,6 +164,8 @@ static int stm32_sai_probe(struct platform_device *pdev) struct reset_control *rst; struct resource *res; const struct of_device_id *of_id; + u32 val; + int ret; sai = devm_kzalloc(&pdev->dev, sizeof(*sai), GFP_KERNEL); if (!sai) @@ -169,7 +178,8 @@ static int stm32_sai_probe(struct platform_device *pdev) of_id = of_match_device(stm32_sai_ids, &pdev->dev); if (of_id) - sai->conf = (struct stm32_sai_conf *)of_id->data; + memcpy(&sai->conf, (const struct stm32_sai_conf *)of_id->data, + sizeof(struct stm32_sai_conf)); else return -EINVAL; @@ -208,6 +218,30 @@ static int stm32_sai_probe(struct platform_device *pdev) reset_control_deassert(rst); } + /* Enable peripheral clock to allow register access */ + ret = clk_prepare_enable(sai->pclk); + if (ret) { + dev_err(&pdev->dev, "failed to enable clock: %d\n", ret); + return ret; + } + + val = FIELD_GET(SAI_IDR_ID_MASK, + readl_relaxed(sai->base + STM_SAI_IDR)); + if (val == SAI_IPIDR_NUMBER) { + val = readl_relaxed(sai->base + STM_SAI_HWCFGR); + sai->conf.fifo_size = FIELD_GET(SAI_HWCFGR_FIFO_SIZE, val); + sai->conf.has_spdif_pdm = !!FIELD_GET(SAI_HWCFGR_SPDIF_PDM, + val); + + val = readl_relaxed(sai->base + STM_SAI_VERR); + sai->conf.version = val; + + dev_dbg(&pdev->dev, "SAI version: %lu.%lu registered\n", + FIELD_GET(SAI_VERR_MAJ_MASK, val), + FIELD_GET(SAI_VERR_MIN_MASK, val)); + } + clk_disable_unprepare(sai->pclk); + sai->pdev = pdev; sai->set_sync = &stm32_sai_set_sync; platform_set_drvdata(pdev, sai); diff --git a/sound/soc/stm/stm32_sai.h b/sound/soc/stm/stm32_sai.h index 9c36a393ab7b..158c73f557f7 100644 --- a/sound/soc/stm/stm32_sai.h +++ b/sound/soc/stm/stm32_sai.h @@ -37,6 +37,12 @@ #define STM_SAI_PDMCR_REGX 0x40 #define STM_SAI_PDMLY_REGX 0x44 +/* Hardware configuration registers */ +#define STM_SAI_HWCFGR 0x3F0 +#define STM_SAI_VERR 0x3F4 +#define STM_SAI_IDR 0x3F8 +#define STM_SAI_SIDR 0x3FC + /******************** Bit definition for SAI_GCR register *******************/ #define SAI_GCR_SYNCIN_SHIFT 0 #define SAI_GCR_SYNCIN_WDTH 2 @@ -82,7 +88,7 @@ #define SAI_XCR1_NODIV BIT(SAI_XCR1_NODIV_SHIFT) #define SAI_XCR1_MCKDIV_SHIFT 20 -#define SAI_XCR1_MCKDIV_WIDTH(x) (((x) == SAI_STM32F4) ? 4 : 6) +#define SAI_XCR1_MCKDIV_WIDTH(x) (((x) == STM_SAI_STM32F4) ? 4 : 6) #define SAI_XCR1_MCKDIV_MASK(x) GENMASK((SAI_XCR1_MCKDIV_SHIFT + (x) - 1),\ SAI_XCR1_MCKDIV_SHIFT) #define SAI_XCR1_MCKDIV_SET(x) ((x) << SAI_XCR1_MCKDIV_SHIFT) @@ -234,8 +240,33 @@ #define SAI_PDMDLY_4R_MASK GENMASK(30, SAI_PDMDLY_4R_SHIFT) #define SAI_PDMDLY_4R_WIDTH 3 -#define STM_SAI_IS_F4(ip) ((ip)->conf->version == SAI_STM32F4) -#define STM_SAI_IS_H7(ip) ((ip)->conf->version == SAI_STM32H7) +/* Registers below apply to SAI version 2.1 and more */ + +/* Bit definition for SAI_HWCFGR register */ +#define SAI_HWCFGR_FIFO_SIZE GENMASK(7, 0) +#define SAI_HWCFGR_SPDIF_PDM GENMASK(11, 8) +#define SAI_HWCFGR_REGOUT GENMASK(19, 12) + +/* Bit definition for SAI_VERR register */ +#define SAI_VERR_MIN_MASK GENMASK(3, 0) +#define SAI_VERR_MAJ_MASK GENMASK(7, 4) + +/* Bit definition for SAI_IDR register */ +#define SAI_IDR_ID_MASK GENMASK(31, 0) + +/* Bit definition for SAI_SIDR register */ +#define SAI_SIDR_ID_MASK GENMASK(31, 0) + +#define SAI_IPIDR_NUMBER 0x00130031 + +/* SAI version numbers are 1.x for F4. Major version number set to 1 for F4 */ +#define STM_SAI_STM32F4 BIT(4) +/* Dummy version number for H7 socs and next */ +#define STM_SAI_STM32H7 0x0 + +#define STM_SAI_IS_F4(ip) ((ip)->conf.version == STM_SAI_STM32F4) +#define STM_SAI_HAS_SPDIF_PDM(ip)\ + ((ip)->pdata->conf.has_spdif_pdm) enum stm32_sai_syncout { STM_SAI_SYNC_OUT_NONE, @@ -243,19 +274,16 @@ enum stm32_sai_syncout { STM_SAI_SYNC_OUT_B, }; -enum stm32_sai_version { - SAI_STM32F4, - SAI_STM32H7 -}; - /** * struct stm32_sai_conf - SAI configuration * @version: SAI version - * @has_spdif: SAI S/PDIF support flag + * @fifo_size: SAI fifo size as words number + * @has_spdif_pdm: SAI S/PDIF and PDM features support flag */ struct stm32_sai_conf { - int version; - bool has_spdif; + u32 version; + u32 fifo_size; + bool has_spdif_pdm; }; /** @@ -265,7 +293,7 @@ struct stm32_sai_conf { * @pclk: SAI bus clock * @clk_x8k: SAI parent clock for sampling frequencies multiple of 8kHz * @clk_x11k: SAI parent clock for sampling frequencies multiple of 11kHz - * @version: SOC version + * @conf: SAI hardware capabitilites * @irq: SAI interrupt line * @set_sync: pointer to synchro mode configuration callback * @gcr: SAI Global Configuration Register @@ -276,7 +304,7 @@ struct stm32_sai_data { struct clk *pclk; struct clk *clk_x8k; struct clk *clk_x11k; - struct stm32_sai_conf *conf; + struct stm32_sai_conf conf; int irq; int (*set_sync)(struct stm32_sai_data *sai, struct device_node *np_provider, int synco, int synci); diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 2a74ce7c9440..7d27efb19380 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -45,7 +45,6 @@ #define SAI_DATASIZE_24 0x6 #define SAI_DATASIZE_32 0x7 -#define STM_SAI_FIFO_SIZE 8 #define STM_SAI_DAI_NAME_SIZE 15 #define STM_SAI_IS_PLAYBACK(ip) ((ip)->dir == SNDRV_PCM_STREAM_PLAYBACK) @@ -63,7 +62,8 @@ #define SAI_SYNC_EXTERNAL 0x2 #define STM_SAI_PROTOCOL_IS_SPDIF(ip) ((ip)->spdif) -#define STM_SAI_HAS_SPDIF(x) ((x)->pdata->conf->has_spdif) +#define STM_SAI_HAS_SPDIF(x) ((x)->pdata->conf.has_spdif_pdm) +#define STM_SAI_HAS_PDM(x) ((x)->pdata->conf.has_spdif_pdm) #define STM_SAI_HAS_EXT_SYNC(x) (!STM_SAI_IS_F4(sai->pdata)) #define SAI_IEC60958_BLOCK_FRAMES 192 @@ -274,7 +274,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, unsigned long input_rate, unsigned long output_rate) { - int version = sai->pdata->conf->version; + int version = sai->pdata->conf.version; int div; div = DIV_ROUND_CLOSEST(input_rate, output_rate); @@ -295,7 +295,7 @@ static int stm32_sai_get_clk_div(struct stm32_sai_sub_data *sai, static int stm32_sai_set_clk_div(struct stm32_sai_sub_data *sai, unsigned int div) { - int version = sai->pdata->conf->version; + int version = sai->pdata->conf.version; int ret, cr1, mask; if (div > SAI_XCR1_MCKDIV_MAX(version)) { @@ -1148,6 +1148,8 @@ static int stm32_sai_dai_probe(struct snd_soc_dai *cpu_dai) * constraints). */ sai->dma_params.maxburst = 4; + if (sai->pdata->conf.fifo_size < 8) + sai->dma_params.maxburst = 1; /* Buswidth will be set by framework at runtime */ sai->dma_params.addr_width = DMA_SLAVE_BUSWIDTH_UNDEFINED; @@ -1315,8 +1317,8 @@ static int stm32_sai_sub_parse_of(struct platform_device *pdev, sai->phys_addr = res->start; sai->regmap_config = &stm32_sai_sub_regmap_config_f4; - /* Note: PDM registers not available for H7 sub-block B */ - if (STM_SAI_IS_H7(sai->pdata) && STM_SAI_IS_SUB_A(sai)) + /* Note: PDM registers not available for sub-block B */ + if (STM_SAI_HAS_PDM(sai) && STM_SAI_IS_SUB_A(sai)) sai->regmap_config = &stm32_sai_sub_regmap_config_h7; sai->regmap = devm_regmap_init_mmio_clk(&pdev->dev, "sai_ck", -- 2.7.4

2 1

[alsa-devel] [PATCH v2] Revert "ASoC: core: use component driver name as component name"
by Krzysztof Kozlowski 03 Jun '19

03 Jun '19

Using component driver as a name is not unique and it breaks audio in certain configurations, e.g. Hardkernel Odroid XU3 board where following components are registered: - "3830000.i2s" with driver name "snd_dmaengine_pcm" - "3830000.i2s-sec" with driver name "snd_dmaengine_pcm" - "3830000.i2s" with driver name "samsung-i2s" This reverts commit b19671d6caf1ac393681864d5d85dda9fa99a448. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Changes since v1: 1. Fix driver name in commit msg. --- sound/soc/soc-core.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2d3520fca613..7abb017a83f3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3113,10 +3113,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, { struct snd_soc_dapm_context *dapm; - if (driver->name) - component->name = kstrdup(driver->name, GFP_KERNEL); - else - component->name = fmt_single_name(dev, &component->id); + component->name = fmt_single_name(dev, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.7.4

2 2

[alsa-devel] Applied "ASoC: SOF: debug: add new debugfs entries for IPC flood test" to the asoc tree
by Mark Brown 03 Jun '19

03 Jun '19

The patch ASoC: SOF: debug: add new debugfs entries for IPC flood test 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 091c12e1f50cce93b1af90e56cad88787ec86dfb Mon Sep 17 00:00:00 2001 From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Date: Mon, 3 Jun 2019 11:18:20 -0500 Subject: [PATCH] ASoC: SOF: debug: add new debugfs entries for IPC flood test Add a couple of new debugfs entries "ipc_flood_count" and "ipc_flood_duration_ms" that can be used to execute the IPC flood test. "ipc_flood_count" floods the DSP with the number of test IPCs specified and ipc_flood_duration_ms floods the DSP with test IPC's for the duration(in ms) specified. The test stats such as average, min and max IPC response times are logged in the dmesg and saved in the debugfs entry cache buffer. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/Kconfig | 8 ++ sound/soc/sof/debug.c | 243 +++++++++++++++++++++++++++++++++++++++ sound/soc/sof/sof-priv.h | 6 +- 3 files changed, 256 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index ca3ad3362478..1d4b4dced4b6 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -131,6 +131,14 @@ config SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE Say Y if you want to enable caching the memory windows. If unsure, select "N". +config SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST + bool "SOF enable IPC flood test" + help + This option enables the IPC flood test which can be used to flood + the DSP with test IPCs and gather stats about response times. + Say Y if you want to enable IPC flood test. + If unsure, select "N". + endif ## SND_SOC_SOF_DEBUG endif ## SND_SOC_SOF_OPTIONS diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 219c3becf670..54bb53bfc81b 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -17,6 +17,203 @@ #include "sof-priv.h" #include "ops.h" +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) +#define MAX_IPC_FLOOD_DURATION_MS 1000 +#define MAX_IPC_FLOOD_COUNT 10000 +#define IPC_FLOOD_TEST_RESULT_LEN 512 + +static int sof_debug_ipc_flood_test(struct snd_sof_dev *sdev, + struct snd_sof_dfsentry *dfse, + bool flood_duration_test, + unsigned long ipc_duration_ms, + unsigned long ipc_count) +{ + struct sof_ipc_cmd_hdr hdr; + struct sof_ipc_reply reply; + u64 min_response_time = U64_MAX; + ktime_t start, end, test_end; + u64 avg_response_time = 0; + u64 max_response_time = 0; + u64 ipc_response_time; + int i = 0; + int ret; + + /* configure test IPC */ + hdr.cmd = SOF_IPC_GLB_TEST_MSG | SOF_IPC_TEST_IPC_FLOOD; + hdr.size = sizeof(hdr); + + /* set test end time for duration flood test */ + if (flood_duration_test) + test_end = ktime_get_ns() + ipc_duration_ms * NSEC_PER_MSEC; + + /* send test IPC's */ + while (1) { + start = ktime_get(); + ret = sof_ipc_tx_message(sdev->ipc, hdr.cmd, &hdr, hdr.size, + &reply, sizeof(reply)); + end = ktime_get(); + + if (ret < 0) + break; + + /* compute min and max response times */ + ipc_response_time = ktime_to_ns(ktime_sub(end, start)); + min_response_time = min(min_response_time, ipc_response_time); + max_response_time = max(max_response_time, ipc_response_time); + + /* sum up response times */ + avg_response_time += ipc_response_time; + i++; + + /* test complete? */ + if (flood_duration_test) { + if (ktime_to_ns(end) >= test_end) + break; + } else { + if (i == ipc_count) + break; + } + } + + if (ret < 0) + dev_err(sdev->dev, + "error: ipc flood test failed at %d iterations\n", i); + + /* return if the first IPC fails */ + if (!i) + return ret; + + /* compute average response time */ + do_div(avg_response_time, i); + + /* clear previous test output */ + memset(dfse->cache_buf, 0, IPC_FLOOD_TEST_RESULT_LEN); + + if (flood_duration_test) { + dev_dbg(sdev->dev, "IPC Flood test duration: %lums\n", + ipc_duration_ms); + snprintf(dfse->cache_buf, IPC_FLOOD_TEST_RESULT_LEN, + "IPC Flood test duration: %lums\n", ipc_duration_ms); + } + + dev_dbg(sdev->dev, + "IPC Flood count: %d, Avg response time: %lluns\n", + i, avg_response_time); + dev_dbg(sdev->dev, "Max response time: %lluns\n", + max_response_time); + dev_dbg(sdev->dev, "Min response time: %lluns\n", + min_response_time); + + /* format output string */ + snprintf(dfse->cache_buf + strlen(dfse->cache_buf), + IPC_FLOOD_TEST_RESULT_LEN - strlen(dfse->cache_buf), + "IPC Flood count: %d\nAvg response time: %lluns\n", + i, avg_response_time); + + snprintf(dfse->cache_buf + strlen(dfse->cache_buf), + IPC_FLOOD_TEST_RESULT_LEN - strlen(dfse->cache_buf), + "Max response time: %lluns\nMin response time: %lluns\n", + max_response_time, min_response_time); + + return ret; +} +#endif + +static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, + size_t count, loff_t *ppos) +{ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) + struct snd_sof_dfsentry *dfse = file->private_data; + struct snd_sof_dev *sdev = dfse->sdev; + unsigned long ipc_duration_ms = 0; + bool flood_duration_test = false; + unsigned long ipc_count = 0; + int err; +#endif + size_t size; + char *string; + int ret; + + string = kzalloc(count, GFP_KERNEL); + if (!string) + return -ENOMEM; + + size = simple_write_to_buffer(string, count, ppos, buffer, count); + ret = size; + +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) + /* + * write op is only supported for ipc_flood_count or + * ipc_flood_duration_ms debugfs entries atm. + * ipc_flood_count floods the DSP with the number of IPC's specified. + * ipc_duration_ms test floods the DSP for the time specified + * in the debugfs entry. + */ + if (strcmp(dfse->dfsentry->d_name.name, "ipc_flood_count") && + strcmp(dfse->dfsentry->d_name.name, "ipc_flood_duration_ms")) + return -EINVAL; + + if (!strcmp(dfse->dfsentry->d_name.name, "ipc_flood_duration_ms")) + flood_duration_test = true; + + /* test completion criterion */ + if (flood_duration_test) + ret = kstrtoul(string, 0, &ipc_duration_ms); + else + ret = kstrtoul(string, 0, &ipc_count); + if (ret < 0) + return ret; + + /* limit max duration/ipc count for flood test */ + if (flood_duration_test) { + if (!ipc_duration_ms) { + ret = size; + goto out; + } + + /* find the minimum. min() is not used to avoid warnings */ + if (ipc_duration_ms > MAX_IPC_FLOOD_DURATION_MS) + ipc_duration_ms = MAX_IPC_FLOOD_DURATION_MS; + } else { + if (!ipc_count) { + ret = size; + goto out; + } + + /* find the minimum. min() is not used to avoid warnings */ + if (ipc_count > MAX_IPC_FLOOD_COUNT) + ipc_count = MAX_IPC_FLOOD_COUNT; + } + + ret = pm_runtime_get_sync(sdev->dev); + if (ret < 0) { + dev_err_ratelimited(sdev->dev, + "error: debugfs write failed to resume %d\n", + ret); + pm_runtime_put_noidle(sdev->dev); + return ret; + } + + /* flood test */ + ret = sof_debug_ipc_flood_test(sdev, dfse, flood_duration_test, + ipc_duration_ms, ipc_count); + + pm_runtime_mark_last_busy(sdev->dev); + err = pm_runtime_put_autosuspend(sdev->dev); + if (err < 0) + dev_err_ratelimited(sdev->dev, + "error: debugfs write failed to idle %d\n", + err); + + /* return size if test is successful */ + if (ret >= 0) + ret = size; +out: +#endif + kfree(string); + return ret; +} + static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer, size_t count, loff_t *ppos) { @@ -28,6 +225,22 @@ static ssize_t sof_dfsentry_read(struct file *file, char __user *buffer, int size; u8 *buf; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) + if ((!strcmp(dfse->dfsentry->d_name.name, "ipc_flood_count") || + !strcmp(dfse->dfsentry->d_name.name, "ipc_flood_duration_ms")) && + dfse->cache_buf) { + if (*ppos) + return 0; + + count = strlen(dfse->cache_buf); + size_ret = copy_to_user(buffer, dfse->cache_buf, count); + if (size_ret) + return -EFAULT; + + *ppos += count; + return count; + } +#endif size = dfse->size; /* validate position & count */ @@ -107,6 +320,7 @@ static const struct file_operations sof_dfs_fops = { .open = simple_open, .read = sof_dfsentry_read, .llseek = default_llseek, + .write = sof_dfsentry_write, }; /* create FS entry for debug files that can expose DSP memories, registers */ @@ -177,6 +391,17 @@ int snd_sof_debugfs_buf_item(struct snd_sof_dev *sdev, dfse->size = size; dfse->sdev = sdev; +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) + /* + * cache_buf is unused for SOF_DFSENTRY_TYPE_BUF debugfs entries. + * So, use it to save the results of the last IPC flood test. + */ + dfse->cache_buf = devm_kzalloc(sdev->dev, IPC_FLOOD_TEST_RESULT_LEN, + GFP_KERNEL); + if (!dfse->cache_buf) + return -ENOMEM; +#endif + dfse->dfsentry = debugfs_create_file(name, mode, sdev->debugfs_root, dfse, &sof_dfs_fops); if (!dfse->dfsentry) { @@ -221,6 +446,24 @@ int snd_sof_dbg_init(struct snd_sof_dev *sdev) return err; } +#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST) + /* create read-write ipc_flood_count debugfs entry */ + err = snd_sof_debugfs_buf_item(sdev, NULL, 0, + "ipc_flood_count", 0666); + + /* errors are only due to memory allocation, not debugfs */ + if (err < 0) + return err; + + /* create read-write ipc_flood_duration_ms debugfs entry */ + err = snd_sof_debugfs_buf_item(sdev, NULL, 0, + "ipc_flood_duration_ms", 0666); + + /* errors are only due to memory allocation, not debugfs */ + if (err < 0) + return err; +#endif + return 0; } EXPORT_SYMBOL_GPL(snd_sof_dbg_init); diff --git a/sound/soc/sof/sof-priv.h b/sound/soc/sof/sof-priv.h index ce22240cd93a..17f3d2a5a701 100644 --- a/sound/soc/sof/sof-priv.h +++ b/sound/soc/sof/sof-priv.h @@ -56,6 +56,10 @@ #define SOF_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_FLOAT) +#define ENABLE_DEBUGFS_CACHEBUF \ + (IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) || \ + IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_IPC_FLOOD_TEST)) + struct snd_sof_dev; struct snd_sof_ipc_msg; struct snd_sof_ipc; @@ -226,7 +230,7 @@ struct snd_sof_dfsentry { * or if it is accessible only when the DSP is in D0. */ enum sof_debugfs_access_type access_type; -#if IS_ENABLED(CONFIG_SND_SOC_SOF_DEBUG_ENABLE_DEBUGFS_CACHE) +#if ENABLE_DEBUGFS_CACHEBUF char *cache_buf; /* buffer to cache the contents of debugfs memory */ #endif struct snd_sof_dev *sdev; -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: soc-core: fix ifnullfree.cocci warnings
by Julia Lawall 03 Jun '19

03 Jun '19

From: kbuild test robot <lkp(a)intel.com> sound/soc/soc-core.c:391:2-7: WARNING: NULL check before some freeing functions is not needed. NULL check before some freeing functions is not needed. Based on checkpatch warning "kfree(NULL) is safe this check is probably not required" and kfreeaddr.cocci by Julia Lawall. Generated by: scripts/coccinelle/free/ifnullfree.cocci Fixes: 8e144273005d ("ASoC: soc-core: merge soc_new_pcm_runtime() and soc_rtd_init()") Signed-off-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Julia Lawall <julia.lawall(a)lip6.fr> --- tree: https://github.com/morimoto/linux fw-cleanup-2019-05-30-v3 head: 7f2c61a9097fcf25dac44865db31e7eab5c8fc9d commit: 8e144273005d3caf8b9978e407e3d00b6180dd49 [96/103] ASoC: soc-core: merge soc_new_pcm_runtime() and soc_rtd_init() :::::: branch date: 12 hours ago :::::: commit date: 12 hours ago Please take the patch only if it's a positive warning. Thanks! soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -387,8 +387,7 @@ static void soc_free_pcm_runtime(struct snd_soc_rtdcom_del_all(rtd); - if (rtd->codec_dais) - kfree(rtd->codec_dais); + kfree(rtd->codec_dais); list_del(&rtd->list); kfree(rtd);

2 1

[alsa-devel] [PATCH v3 00/30] Update cros_ec_commands.h
by Gwendal Grignou 03 Jun '19

03 Jun '19

The interface between CrosEC embedded controller and the host, described by cros_ec_commands.h, as diverged from what the embedded controller really support. The source of thruth is at https://chromium.googlesource.com/chromiumos/platform/ec/+/master/include/e… That include file is converted to remove ACPI and Embedded only code. >From now on, cros_ec_commands.h will be automatically generated from the file above, do not modify directly. Fell free to squash the commits below. Changes in v3: - Rebase after commit 81888d8ab1532 ("mfd: cros_ec: Update the EC feature codes") - Add Acked-by: Benson Leung <bleung(a)chromium.org> Changes in v2: - Move I2S changes at the end of the patchset, squashed with change in sound/soc/codecs/cros_ec_codec.c to match new interface. - Add Acked-by: Enric Balletbo i Serra <enric.balletbo(a)collabora.com> Gwendal Grignou (30): mfd: cros_ec: Update license term mfd: cros_ec: Zero BUILD_ macro mfd: cros_ec: set comments properly mfd: cros_ec: add ec_align macros mfd: cros_ec: Define commands as 4-digit UPPER CASE hex values mfd: cros_ec: use BIT macro mfd: cros_ec: Update ACPI interface definition mfd: cros_ec: move HDMI CEC API definition mfd: cros_ec: Remove zero-size structs mfd: cros_ec: Add Flash V2 commands API mfd: cros_ec: Add PWM_SET_DUTY API mfd: cros_ec: Add lightbar v2 API mfd: cros_ec: Expand hash API mfd: cros_ec: Add EC transport protocol v4 mfd: cros_ec: Complete MEMS sensor API mfd: cros_ec: Fix event processing API mfd: cros_ec: Add fingerprint API mfd: cros_ec: Fix temperature API mfd: cros_ec: Complete Power and USB PD API mfd: cros_ec: Add API for keyboard testing mfd: cros_ec: Add Hibernate API mfd: cros_ec: Add Smart Battery Firmware update API mfd: cros_ec: Add I2C passthru protection API mfd: cros_ec: Add API for EC-EC communication mfd: cros_ec: Add API for Touchpad support mfd: cros_ec: Add API for Fingerprint support mfd: cros_ec: Add API for rwsig mfd: cros_ec: Add SKU ID and Secure storage API mfd: cros_ec: Add Management API entry points mfd: cros_ec: Update I2S API include/linux/mfd/cros_ec_commands.h | 3658 ++++++++++++++++++++------ sound/soc/codecs/cros_ec_codec.c | 8 +- 2 files changed, 2915 insertions(+), 751 deletions(-) -- 2.21.0.1020.gf2820cf01a-goog

4 37

[alsa-devel] [PATCH v2] mfd: madera: Add missing of table registration
by Daniel Gomez 03 Jun '19

03 Jun '19

MODULE_DEVICE_TABLE(of, <of_match_table> should be called to complete DT OF mathing mechanism and register it. Before this patch: modinfo ./drivers/mfd/madera.ko | grep alias After this patch: modinfo ./drivers/mfd/madera.ko | grep alias alias: of:N*T*Ccirrus,wm1840C* alias: of:N*T*Ccirrus,wm1840 alias: of:N*T*Ccirrus,cs47l91C* alias: of:N*T*Ccirrus,cs47l91 alias: of:N*T*Ccirrus,cs47l90C* alias: of:N*T*Ccirrus,cs47l90 alias: of:N*T*Ccirrus,cs47l85C* alias: of:N*T*Ccirrus,cs47l85 alias: of:N*T*Ccirrus,cs47l35C* alias: of:N*T*Ccirrus,cs47l35 Reported-by: Javier Martinez Canillas <javier(a)dowhile0.org> Signed-off-by: Daniel Gomez <dagmcr(a)gmail.com> --- drivers/mfd/madera-core.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/mfd/madera-core.c b/drivers/mfd/madera-core.c index 2a77988..826b971 100644 --- a/drivers/mfd/madera-core.c +++ b/drivers/mfd/madera-core.c @@ -286,6 +286,7 @@ const struct of_device_id madera_of_match[] = { { .compatible = "cirrus,wm1840", .data = (void *)WM1840 }, {} }; +MODULE_DEVICE_TABLE(of, madera_of_match); EXPORT_SYMBOL_GPL(madera_of_match); static int madera_get_reset_gpio(struct madera *madera) -- 2.7.4

3 2

[alsa-devel] [PATCH] treewide: fix typos of SPDX-License-Identifier
by Masahiro Yamada 03 Jun '19

03 Jun '19

Prior to the adoption of SPDX, it was difficult for tools to determine the correct license due to incomplete or badly formatted license text. The SPDX solves this issue, assuming people can correctly spell "SPDX-License-Identifier" although this assumption is broken in some places. Since scripts/spdxcheck.py parses only lines that exactly matches to the correct tag, it cannot (should not) detect this kind of error. If the correct tag is missing, scripts/checkpatch.pl warns like this: WARNING: Missing or malformed SPDX-License-Identifier tag in line * So, people should notice it before the patch submission, but in reality broken tags sometimes slip in. The checkpatch warning is not useful for checking the committed files globally since large number of files still have no SPDX tag. Also, I am not sure about the legal effect when the SPDX tag is broken. Anyway, these typos are absolutely worth fixing. It is pretty easy to find suspicious lines by grep. $ git grep --not -e SPDX-License-Identifier --and -e SPDX- -- \ :^LICENSES :^scripts/spdxcheck.py :^*/license-rules.rst arch/arm/kernel/bugs.c:// SPDX-Identifier: GPL-2.0 drivers/phy/st/phy-stm32-usbphyc.c:// SPDX-Licence-Identifier: GPL-2.0 drivers/pinctrl/sh-pfc/pfc-r8a77980.c:// SPDX-Lincense-Identifier: GPL 2.0 lib/test_stackinit.c:// SPDX-Licenses: GPLv2 sound/soc/codecs/max9759.c:// SPDX-Licence-Identifier: GPL-2.0 Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> --- arch/arm/kernel/bugs.c | 2 +- drivers/phy/st/phy-stm32-usbphyc.c | 2 +- drivers/pinctrl/sh-pfc/pfc-r8a77980.c | 2 +- lib/test_stackinit.c | 2 +- sound/soc/codecs/max9759.c | 2 +- 5 files changed, 5 insertions(+), 5 deletions(-) diff --git a/arch/arm/kernel/bugs.c b/arch/arm/kernel/bugs.c index d41d3598e5e5..14c8dbbb7d2d 100644 --- a/arch/arm/kernel/bugs.c +++ b/arch/arm/kernel/bugs.c @@ -1,4 +1,4 @@ -// SPDX-Identifier: GPL-2.0 +// SPDX-License-Identifier: GPL-2.0 #include <linux/init.h> #include <asm/bugs.h> #include <asm/proc-fns.h> diff --git a/drivers/phy/st/phy-stm32-usbphyc.c b/drivers/phy/st/phy-stm32-usbphyc.c index 1255cd1d9a60..56bdea4b0bd9 100644 --- a/drivers/phy/st/phy-stm32-usbphyc.c +++ b/drivers/phy/st/phy-stm32-usbphyc.c @@ -1,4 +1,4 @@ -// SPDX-Licence-Identifier: GPL-2.0 +// SPDX-License-Identifier: GPL-2.0 /* * STMicroelectronics STM32 USB PHY Controller driver * diff --git a/drivers/pinctrl/sh-pfc/pfc-r8a77980.c b/drivers/pinctrl/sh-pfc/pfc-r8a77980.c index 473da65890a7..9ed4ead2dafb 100644 --- a/drivers/pinctrl/sh-pfc/pfc-r8a77980.c +++ b/drivers/pinctrl/sh-pfc/pfc-r8a77980.c @@ -1,4 +1,4 @@ -// SPDX-Lincense-Identifier: GPL 2.0 +// SPDX-License-Identifier: GPL-2.0 /* * R8A77980 processor support - PFC hardware block. * diff --git a/lib/test_stackinit.c b/lib/test_stackinit.c index 13115b6f2b88..e97dc54b4fdf 100644 --- a/lib/test_stackinit.c +++ b/lib/test_stackinit.c @@ -1,4 +1,4 @@ -// SPDX-Licenses: GPLv2 +// SPDX-License-Identifier: GPL-2.0 /* * Test cases for compiler-based stack variable zeroing via future * compiler flags or CONFIG_GCC_PLUGIN_STRUCTLEAK*. diff --git a/sound/soc/codecs/max9759.c b/sound/soc/codecs/max9759.c index ecfb4a80424b..00e9d4fd1651 100644 --- a/sound/soc/codecs/max9759.c +++ b/sound/soc/codecs/max9759.c @@ -1,4 +1,4 @@ -// SPDX-Licence-Identifier: GPL-2.0 +// SPDX-License-Identifier: GPL-2.0 /* * MAX9759 Amplifier Driver * -- 2.17.1

3 2

[alsa-devel] Why open-coding in sof_hda_bus_init()?
by Takashi Iwai 03 Jun '19

03 Jun '19

Hi, while looking at SOF code due to the recent debugging session, I noticed that sof_hda_bus_init() is basically an open-code of the existing snd_hdac_ext_bus_init(). Why don't we simply call snd_hdac_ext_bus_init() like below? The only thing that differs is the handling of bus->id number, where SOF fixes to zero (which was actually done in a patch after the first commit). But judging from the comment, this doesn't seem like a big matter, as we assume a single bus, so far... thanks, Takashi --- diff --git a/sound/soc/sof/intel/Makefile b/sound/soc/sof/intel/Makefile index b8f58e006e29..f42450a9a7b6 100644 --- a/sound/soc/sof/intel/Makefile +++ b/sound/soc/sof/intel/Makefile @@ -7,7 +7,7 @@ snd-sof-intel-ipc-objs := intel-ipc.o snd-sof-intel-hda-common-objs := hda.o hda-loader.o hda-stream.o hda-trace.o \ hda-dsp.o hda-ipc.o hda-ctrl.o hda-pcm.o \ - hda-dai.o hda-bus.o \ + hda-dai.o \ apl.o cnl.o snd-sof-intel-hda-objs := hda-codec.o diff --git a/sound/soc/sof/intel/hda-bus.c b/sound/soc/sof/intel/hda-bus.c deleted file mode 100644 index a7e6d8227df6..000000000000 --- a/sound/soc/sof/intel/hda-bus.c +++ /dev/null @@ -1,111 +0,0 @@ -// SPDX-License-Identifier: (GPL-2.0 OR BSD-3-Clause) -// -// This file is provided under a dual BSD/GPLv2 license. When using or -// redistributing this file, you may do so under either license. -// -// Copyright(c) 2018 Intel Corporation. All rights reserved. -// -// Authors: Keyon Jie <yang.jie(a)linux.intel.com> - -#include <linux/io.h> -#include <sound/hdaudio.h> -#include "../sof-priv.h" -#include "hda.h" - -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - -static const struct hdac_bus_ops bus_ops = { - .command = snd_hdac_bus_send_cmd, - .get_response = snd_hdac_bus_get_response, -}; - -#endif - -static void sof_hda_writel(u32 value, u32 __iomem *addr) -{ - writel(value, addr); -} - -static u32 sof_hda_readl(u32 __iomem *addr) -{ - return readl(addr); -} - -static void sof_hda_writew(u16 value, u16 __iomem *addr) -{ - writew(value, addr); -} - -static u16 sof_hda_readw(u16 __iomem *addr) -{ - return readw(addr); -} - -static void sof_hda_writeb(u8 value, u8 __iomem *addr) -{ - writeb(value, addr); -} - -static u8 sof_hda_readb(u8 __iomem *addr) -{ - return readb(addr); -} - -static int sof_hda_dma_alloc_pages(struct hdac_bus *bus, int type, - size_t size, struct snd_dma_buffer *buf) -{ - return snd_dma_alloc_pages(type, bus->dev, size, buf); -} - -static void sof_hda_dma_free_pages(struct hdac_bus *bus, - struct snd_dma_buffer *buf) -{ - snd_dma_free_pages(buf); -} - -static const struct hdac_io_ops io_ops = { - .reg_writel = sof_hda_writel, - .reg_readl = sof_hda_readl, - .reg_writew = sof_hda_writew, - .reg_readw = sof_hda_readw, - .reg_writeb = sof_hda_writeb, - .reg_readb = sof_hda_readb, - .dma_alloc_pages = sof_hda_dma_alloc_pages, - .dma_free_pages = sof_hda_dma_free_pages, -}; - -/* - * This can be used for both with/without hda link support. - */ -void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev, - const struct hdac_ext_bus_ops *ext_ops) -{ - memset(bus, 0, sizeof(*bus)); - bus->dev = dev; - - bus->io_ops = &io_ops; - INIT_LIST_HEAD(&bus->stream_list); - - bus->irq = -1; - bus->ext_ops = ext_ops; - - /* - * There is only one HDA bus atm. keep the index as 0. - * Need to fix when there are more than one HDA bus. - */ - bus->idx = 0; - - spin_lock_init(&bus->reg_lock); - -#if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) - INIT_LIST_HEAD(&bus->codec_list); - INIT_LIST_HEAD(&bus->hlink_list); - - mutex_init(&bus->cmd_mutex); - mutex_init(&bus->lock); - bus->ops = &bus_ops; - INIT_WORK(&bus->unsol_work, snd_hdac_bus_process_unsol_events); - bus->cmd_dma_state = true; -#endif - -} diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 92d45c43b4b1..a9de6a297b56 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -532,8 +532,11 @@ int hda_dsp_ctrl_init_chip(struct snd_sof_dev *sdev, bool full_reset); /* * HDA bus operations. */ -void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev, - const struct hdac_ext_bus_ops *ext_ops); +static inline void sof_hda_bus_init(struct hdac_bus *bus, struct device *dev, + const struct hdac_ext_bus_ops *ext_ops) +{ + snd_hdac_ext_bus_init(bus, dev, NULL, NULL, ext_ops); +} #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) /*

3 10

[alsa-devel] [PATCH] ALSA: pcm: Check for integer overflow during multiplication
by bgoswami＠codeaurora.org 03 Jun '19

03 Jun '19

From: Phani Kumar Uppalapati <phaniu(a)codeaurora.org> Channel info data structure is parsed from userspace and if the number of channels is not set correctly, it could lead to integer overflow when the number of channels is multiplied with pcm bit width. Add a condition to check for integer overflow during the multiplication operationi, and return error if overflow detected. Signed-off-by: Phani Kumar Uppalapati <phaniu(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- sound/core/pcm_lib.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index 345ab1a..f45ae3a 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1718,6 +1718,11 @@ static int snd_pcm_lib_ioctl_channel_info(struct snd_pcm_substream *substream, switch (runtime->access) { case SNDRV_PCM_ACCESS_MMAP_INTERLEAVED: case SNDRV_PCM_ACCESS_RW_INTERLEAVED: + if ((UINT_MAX/width) < info->channel) { + snd_printd("%s: integer overflow in multiplication\n", + __func__); + return -EINVAL; + } info->first = info->channel * width; info->step = runtime->channels * width; break; @@ -1725,6 +1730,12 @@ static int snd_pcm_lib_ioctl_channel_info(struct snd_pcm_substream *substream, case SNDRV_PCM_ACCESS_RW_NONINTERLEAVED: { size_t size = runtime->dma_bytes / runtime->channels; + + if ((size > 0) && ((UINT_MAX/(size * 8)) < info->channel)) { + snd_printd("%s: integer overflow in multiplication\n", + __func__); + return -EINVAL; + } info->first = info->channel * size * 8; info->step = width; break; -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

3 2

[alsa-devel] [PATCH V3] ASoC: Intel: Baytrail: add quirk for Aegex 10 (RU2) tablet
by Kovács Tamás 31 May '19

31 May '19

This tablet has an incorrect acpi identifier just like Thinkpad10 tablet, which is why it is trying to load the RT5640 driver instead of the RT5762 driver. The RT5640 driver, on the other hand, checks the hardware ID, so no driver are loaded during boot. This fix resolves to load the RT5672 driver on this tablet during boot. It also provides the correct IO configuration, like the jack detect mode 3, for 1.8V pullup. I would like to thank Pierre-Louis Bossart for helping with this patch. Signed-off-by: Kovács Tamás <kepszlok(a)zohomail.eu> --- sound/soc/codecs/rt5670.c | 12 ++++++++++++ .../soc/intel/common/soc-acpi-intel-byt-match.c | 17 +++++++++++++++++ 2 files changed, 29 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9a037108b1ae..a746e11ccfe3 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2882,6 +2882,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { RT5670_DEV_GPIO | RT5670_JD_MODE3), }, + { + .callback = rt5670_quirk_cb, + .ident = "Aegex 10 tablet (RU2)", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AEGEX"), + DMI_MATCH(DMI_PRODUCT_VERSION, "RU2"), + }, + .driver_data = (unsigned long *)(RT5670_DMIC_EN | + RT5670_DMIC2_INR | + RT5670_DEV_GPIO | + RT5670_JD_MODE3), + }, {} }; diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index 0cfab247876a..9fb58f1f095f 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -22,6 +22,7 @@ static unsigned long byt_machine_id; #define BYT_THINKPAD_10 1 #define BYT_POV_P1006W 2 +#define BYT_AEGEX_10 3 static int byt_thinkpad10_quirk_cb(const struct dmi_system_id *id) { @@ -35,6 +36,12 @@ static int byt_pov_p1006w_quirk_cb(const struct dmi_system_id *id) return 1; } +static int byt_aegex10_quirk_cb(const struct dmi_system_id *id) +{ + byt_machine_id = BYT_AEGEX_10; + return 1; +} + static const struct dmi_system_id byt_table[] = { { .callback = byt_thinkpad10_quirk_cb, @@ -75,9 +82,18 @@ static const struct dmi_system_id byt_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "0E57"), }, }, + { + /* Aegex 10 tablet (RU2) */ + .callback = byt_aegex10_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AEGEX"), + DMI_MATCH(DMI_PRODUCT_VERSION, "RU2"), + }, + }, { } }; +/* The Thinkapd 10 and Aegex 10 tablets have the same ID problem */ static struct snd_soc_acpi_mach byt_thinkpad_10 = { .id = "10EC5640", .drv_name = "cht-bsw-rt5672", @@ -104,6 +120,7 @@ static struct snd_soc_acpi_mach *byt_quirk(void *arg) switch (byt_machine_id) { case BYT_THINKPAD_10: + case BYT_AEGEX_10: return &byt_thinkpad_10; case BYT_POV_P1006W: return &byt_pov_p1006w;

2 1

[alsa-devel] [PATCH] ASoC: hda: increment codec device refcount when it is added to the card
by Ranjani Sridharan 31 May '19

31 May '19

Calling snd_device_new() makes the codec devices managed by the card. So, when the card is removed, the refcount for the codec device is decremented and results in the codec device's kobject being cleaned up if the refcount is 0. But, this leads to a NULL pointer exception while attempting to remove the symlinks when the codec driver is released later on. Therefore, increment the codec device's refcount before adding it to the card to prevent this. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/pci/hda/hda_codec.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b20eb7fc83eb..0d5d95b07e19 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -985,6 +985,14 @@ int snd_hda_codec_device_new(struct hda_bus *bus, struct snd_card *card, codec->core.subsystem_id, codec->core.revision_id); snd_component_add(card, component); + /* + * snd_device_new() makes the codec device managed by the card. + * When the card is removed, the device reference count is + * decremented. Therefore, increment it here to prevent removing + * the codec device's kobject when the card is removed. + */ + get_device(hda_codec_dev(codec)); + err = snd_device_new(card, SNDRV_DEV_CODEC, codec, &dev_ops); if (err < 0) goto error; -- 2.17.1

3 11

[alsa-devel] [PATCH] ASoC: hdac_hda: Fix unbalanced bus link refcount at probe error
by Takashi Iwai 31 May '19

31 May '19

The error paths in hdac_hda_codec_probe() don't take care of the bus link refcount properly, which leave the refcount still high. This patch addresses them. Fixes: 6bae5ea94989 ("ASoC: hdac_hda: add asoc extension for legacy HDA codec drivers") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- A bug I found while digging for another problem :) sound/soc/codecs/hdac_hda.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index 7d4940256914..b55deaeb1ebf 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -475,8 +475,10 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) snd_hdac_ext_bus_link_get(hdev->bus, hlink); hda_pvt = hdac_to_hda_priv(hdev); - if (!hda_pvt) - return -ENOMEM; + if (!hda_pvt) { + ret = -ENOMEM; + goto error; + } /* ASoC specific initialization */ ret = devm_snd_soc_register_component(&hdev->dev, @@ -484,12 +486,13 @@ static int hdac_hda_dev_probe(struct hdac_device *hdev) ARRAY_SIZE(hdac_hda_dais)); if (ret < 0) { dev_err(&hdev->dev, "failed to register HDA codec %d\n", ret); - return ret; + goto error; } dev_set_drvdata(&hdev->dev, hda_pvt); - snd_hdac_ext_bus_link_put(hdev->bus, hlink); + error: + snd_hdac_ext_bus_link_put(hdev->bus, hlink); return ret; } -- 2.16.4

2 2

[alsa-devel] [PATCH v8 0/6] drm/bridge: sii902x: HDMI-audio support and some fixes
by Jyri Sarha 31 May '19

31 May '19

I think these should be ready for applying to drm-misc. Changes since v7: - Debased on top of the lasts drm-misc-next and tested - "dt-bindings: display: sii902x: Add HDMI audio bindings" - Dropped off "or higher to avoid conflict with video ports" and added "Reviewed-by: Rob Herring <robh(a)kernel.org>" Ther previous round: https://patchwork.kernel.org/cover/10919173/ 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 | 42 +- drivers/gpu/drm/bridge/sii902x.c | 488 +++++++++++++++++- 2 files changed, 522 insertions(+), 8 deletions(-) -- Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 7

[alsa-devel] [BISECT] No audio after "ASoC: core: use component driver name as component name"
by Krzysztof Kozlowski 31 May '19

31 May '19

Hi, Bisect points to commit b19671d6caf1ac393681864d5d85dda9fa99a448 Author: Tzung-Bi Shih <tzungbi(a)google.com> Date: Wed May 29 14:22:14 2019 +0800 ASoC: core: use component driver name as component name as a reason of failure of missing Audio card on Odroid XU3 board (ARMv7, Exynos5422, max98090 codec). Full kernel log: https://krzk.eu/#/builders/1/builds/3349/steps/14/logs/serial0 The problem might be in component name. The driver->name and fmt_single_name(dev, &component->id) are: snd_dmaengine_pcm != 3830000.i2s snd_dmaengine_pcm != 3830000.i2s-sec samsung-i2s != 3830000.i2s This commit should not go in without fixing the users of old behavior... I could adjust the platform names for primary and secondary links... but now it looks like two components will have the same name. Best regards, Krzysztof

3 2

[alsa-devel] aplay - pcm_read: Input/output error
by GitHub issues - edited 31 May '19

31 May '19

alsa-project/alsa-lib issue #7 was edited from mrajwa: Hello, I came across an issue with custom built aplay (system one works OK on Ubuntu 18.04): I have fetched both alsa-lib and alsa-utils repositories, built them now try to run: `path_to_aplay/aplay -C -D hw:0,8 -r 16000 -f S16_LE -c 2 tmp.wav -vvv` Immediately after the start it prints hw params and the error message saying: `arecord: pcm_read:2103: read errorL Input/output error` As I said system aplay works fine. Moreover I compared hw params for both system and this custom built aplay and they are exactly the same! I also tried to built alsa-utils from debian package here I did: `apt source alsa-lib && sudo apt build-dep alsa-utils && fakeroot debian/rules clean && fakeroot debian/rules build && fakeroot debian/rules binary` All finished fine, now I repeated above steps for alsa-utils and here is a problem at `fakeroot debian/rules binary` step. At some point it stops with an error message saying: `dpkg-shlibdeps: error: no dependency information found for /usr/lib/libasound.co.s2 (used by debian/alsa-utils/usr/bin/alsaucm)` Does anybody have any idea what is going on here? Issue URL : https://github.com/alsa-project/alsa-lib/issues/7 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] aplay - pcm_read: Input/output error
by GitHub issues - edited 31 May '19

31 May '19

alsa-project/alsa-lib issue #7 was edited from mrajwa: Hello, I came across an issue with custom built aplay (system one works OK on Ubuntu 18.04): I have fetched both alsa-lib and alsa-utils repositories, built them now try to run: `path_to_aplay/aplay -C -D hw:0,8 -r 16000 -f S16_LE -c 2 tmp.wav -vvv` Immediately after the start it prints hw params and the error message saying: `arecord: pcm_read:2103: read errorL Input/output error` As I said system aplay works fine. Moreover I compared hw params for both system and this custom built aplay and they are exactly the same! I also tried to built alsa-utils from debian package here I did: apt source alsa-lib && sudo apt build-dep alsa-utils && fakeroot debian/rules clean && fakeroot debian/rules build && fakeroot debian/rules binary All finished fine, now I repeated above steps for alsa-utils and here is aproblem at fakeroot debian/rules binary step. At some point it stops with error message saying: `dpkg-shlibdeps: error: no dependency information found for /usr/lib/libasound.co.s2 (used by debian/alsa-utils/usr/bin/alsaucm)` Does anybody have any idea what is going on here? Issue URL : https://github.com/alsa-project/alsa-lib/issues/7 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] aplay - pcm_read: Input/output error
by GitHub issues - opened 31 May '19

31 May '19

alsa-project/alsa-lib issue #7 was opened from mrajwa: Hello, I came across an issue with custom built aplay (system one works OK on Ubuntu 18.04): I have fetched both alsa-lib and alsa-utils repositories, built them now try to run: `path_to_aplay/aplay -C -D hw:0,8 -r 16000 -f S16_LE -c 2 tmp.wav -vvv` Immediately after the start it prints hw_params and the and error message saying: `arecord: pcm_read:2103: read errorL Input/output error` As I said system aplay works fine. Moreover I compared hw_params for both system and this custom built aplay and they are exactly the same! I also tried to built alsa-utils from debian package here I did: apt source alsa-lib && sudo apt build-dep alsa-utils && fakeroot debian/rules clean && fakeroot debian/rules build && fakeroot debian/rules binary All finished fine, now I repeated above steps for alsa-utils and here is aproblem at fakeroot debian/rules binary step. At some point it stops with error message saying: `dpkg-shlibdeps: error: no dependency information found for /usr/lib/libasound.co.s2 (used by debian/alsa-utils/usr/bin/alsaucm)` Does anybody have any idea what is going on here? Issue URL : https://github.com/alsa-project/alsa-lib/issues/7 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[alsa-devel] [PATCH] Revert "ASoC: core: use component driver name as component name"
by Krzysztof Kozlowski 31 May '19

31 May '19

Using component driver as a name is not unique and it breaks audio in certain configurations, e.g. Hardkernel Odroid XU3 board where following components are registered: - 3830000.i2s with driver snd_dmaengine_pcm - 3830000.i2s-sec with driver snd_dmaengine_pcm - 3830000.i2s with driver 3830000.i2s-sec This reverts commit b19671d6caf1ac393681864d5d85dda9fa99a448. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- sound/soc/soc-core.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2d3520fca613..7abb017a83f3 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3113,10 +3113,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, { struct snd_soc_dapm_context *dapm; - if (driver->name) - component->name = kstrdup(driver->name, GFP_KERNEL); - else - component->name = fmt_single_name(dev, &component->id); + component->name = fmt_single_name(dev, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.7.4

1 0

[alsa-devel] [PATCH 0/3] ALSA: trivial conversions to struct_size()
by Takashi Iwai 31 May '19

31 May '19

Hi, there are a few more places that can be better written with struct_size() macro. Let's replace them. Takashi === Takashi Iwai (3): ALSA: control: Use struct_size() ALSA: hda: Use struct_size() ALSA: hda/ca0132: Use struct_size() sound/core/control.c | 6 +----- sound/pci/hda/hda_codec.c | 2 +- sound/pci/hda/patch_ca0132.c | 2 +- 3 files changed, 3 insertions(+), 7 deletions(-) -- 2.16.4

1 3

[alsa-devel] ALC256 headset mode update
by Kailang 31 May '19

31 May '19

Hi Takashi, Since ALC256 was used for many machines. So, update ALC256 headset mode for separate with ALC255. Many Thanks. BR, Kailang

2 1

[alsa-devel] Fwd: Problem with aplay params
by Martin R 31 May '19

31 May '19

Hello, I have built aplay using github repository https://github.com/alsa-project/alsa-utils/ as well as https://github.com/thesofproject/alsa-lib. Now when I try to run it by "./path_to_aplay/aplay -C -D hw:0,8 -r 16000 -f S16_LE -c 2 tmp.wav -vvv I get immediate error message saying: arecord: pcm_read:2150 read error: Input/output error. Why is it failing? The system aplay/arecord works OK - only the one I built fails with above error. PS: system aplay version is 1.1.8 while the one built by me is 1.19 Regards, Marcin

1 0

[alsa-devel] Question about dapm setup
by Kuninori Morimoto 31 May '19

31 May '19

Hi ALSA ML snd_soc_instantiate_card() setups dapm, but its timing seems very randomly for me. In my understanding, dapm setup timing is not so serious. So, I think we can do it in one place, but are there some reasons ? For example, "xxxx should be called after yyyy" static int snd_soc_instantiate_card(struct snd_soc_card *card) { ... => snd_soc_dapm_debugfs_init(&card->dapm, card->debugfs_card_root); ... if (card->dapm_widgets) snd_soc_dapm_new_controls(&card->dapm, card->dapm_widgets, card->num_dapm_widgets); if (card->of_dapm_widgets) snd_soc_dapm_new_controls(&card->dapm, card->of_dapm_widgets, card->num_of_dapm_widgets); ... snd_soc_dapm_link_dai_widgets(card); snd_soc_dapm_connect_dai_link_widgets(card); if (card->controls) snd_soc_add_card_controls(card, card->controls, card->num_controls); if (card->dapm_routes) snd_soc_dapm_add_routes(&card->dapm, card->dapm_routes, card->num_dapm_routes); if (card->of_dapm_routes) snd_soc_dapm_add_routes(&card->dapm, card->of_dapm_routes, card->num_of_dapm_routes); ... snd_soc_dapm_new_widgets(card); ... dapm_mark_endpoints_dirty(card); snd_soc_dapm_sync(&card->dapm); ... } Thank you for your help !! Best regards --- Kuninori Morimoto

2 4

[alsa-devel] [PATCH V2] ASoC: Intel: Baytrail: add quirk for Aegex 10 (RU2) tablet
by Kovács Tamás 30 May '19

30 May '19

This tablet has an incorrect acpi identifier just like Thinkpad10 tablet, which is why it is trying to load the RT5640 driver instead of the RT5762 driver. The RT5640 driver, on the other hand, checks the hardware ID, so no driver are loaded during boot. This fix resolves to load the RT5672 driver on this tablet during boot. It also provides the correct IO configuration, like the jack detect mode 3, for 1.8V pullup. I would like to thank Pierre-Louis Bossart for helping with this patch. --- sound/soc/codecs/rt5670.c | 14 +------------- .../intel/common/soc-acpi-intel-byt-match.c | 19 +------------------ 2 files changed, 2 insertions(+), 31 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 37cce0439568..9a037108b1ae 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2882,18 +2882,6 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { RT5670_DEV_GPIO | RT5670_JD_MODE3), }, - { - .callback = rt5670_quirk_cb, - .ident = "Aegex 10 tablet (RU2)", - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "AEGEX"), - DMI_MATCH(DMI_PRODUCT_VERSION, "RU2"), - }, - .driver_data = (unsigned long *)(RT5670_DMIC_EN | - RT5670_DMIC2_INR | - RT5670_DEV_GPIO | - RT5670_JD_MODE3), - }, {} }; diff --git a/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/sound/soc/intel/common/soc-acpi-intel-byt-match.c index cfed58c23e0b..fe812a909db4 100644 --- a/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -22,7 +22,6 @@ static unsigned long byt_machine_id; #define BYT_THINKPAD_10 1 #define BYT_POV_P1006W 2 -#define BYT_AEGEX_10 3 static int byt_thinkpad10_quirk_cb(const struct dmi_system_id *id) { @@ -36,12 +35,6 @@ static int byt_pov_p1006w_quirk_cb(const struct dmi_system_id *id) return 1; } -static int byt_aegex10_quirk_cb(const struct dmi_system_id *id) -{ - byt_machine_id = BYT_AEGEX_10; - return 1; -} - static const struct dmi_system_id byt_table[] = { { .callback = byt_thinkpad10_quirk_cb, @@ -82,18 +75,9 @@ static const struct dmi_system_id byt_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "0E57"), }, }, - { - /* Aegex 10 tablet (RU2) */ - .callback = byt_aegex10_quirk_cb, - .matches = { - DMI_MATCH(DMI_SYS_VENDOR, "AEGEX"), - DMI_MATCH(DMI_PRODUCT_VERSION, "RU2"), - }, - }, { } }; -/* The Thinkapd 10 and Aegex 10 tablets have the same ID problem */ static struct snd_soc_acpi_mach byt_thinkpad_10 = { .id = "10EC5640", .drv_name = "cht-bsw-rt5672", @@ -120,7 +104,6 @@ static struct snd_soc_acpi_mach *byt_quirk(void *arg) switch (byt_machine_id) { case BYT_THINKPAD_10: - case BYT_AEGEX_10: return &byt_thinkpad_10; case BYT_POV_P1006W: return &byt_pov_p1006w;

2 1

[alsa-devel] [PATCH] ASoC: rt5677: Add missing voice activation register definitions
by Curtis Malainey 30 May '19

30 May '19

Most of the voice activation definitions were missing, they will be needed for when hotwording is added. Also the source bits are only 2 wide not 3 and needed to be corrected. Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677.h | 30 +++++++++++++++++++++++++++++- 1 file changed, 29 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5677.h b/sound/soc/codecs/rt5677.h index 11a2ffceec3b..076e5161d8da 100644 --- a/sound/soc/codecs/rt5677.h +++ b/sound/soc/codecs/rt5677.h @@ -1456,9 +1456,37 @@ #define RT5677_I2S4_CLK_SEL_MASK (0xf) #define RT5677_I2S4_CLK_SEL_SFT 0 +/* VAD Function Control 1 (0x9c) */ +#define RT5677_VAD_MIN_DUR_MASK (0x3 << 13) +#define RT5677_VAD_MIN_DUR_SFT 13 +#define RT5677_VAD_ADPCM_BYPASS (1 << 10) +#define RT5677_VAD_ADPCM_BYPASS_BIT 10 +#define RT5677_VAD_FG2ENC (1 << 9) +#define RT5677_VAD_FG2ENC_BIT 9 +#define RT5677_VAD_BUF_OW (1 << 8) +#define RT5677_VAD_BUF_OW_BIT 8 +#define RT5677_VAD_CLR_FLAG (1 << 7) +#define RT5677_VAD_CLR_FLAG_BIT 7 +#define RT5677_VAD_BUF_POP (1 << 6) +#define RT5677_VAD_BUF_POP_BIT 6 +#define RT5677_VAD_BUF_PUSH (1 << 5) +#define RT5677_VAD_BUF_PUSH_BIT 5 +#define RT5677_VAD_DET_ENABLE (1 << 4) +#define RT5677_VAD_DET_ENABLE_BIT 4 +#define RT5677_VAD_FUNC_ENABLE (1 << 3) +#define RT5677_VAD_FUNC_ENABLE_BIT 3 +#define RT5677_VAD_FUNC_RESET (1 << 2) +#define RT5677_VAD_FUNC_RESET_BIT 2 + /* VAD Function Control 4 (0x9f) */ -#define RT5677_VAD_SRC_MASK (0x7 << 8) +#define RT5677_VAD_OUT_SRC_RATE_MASK (0x1 << 11) +#define RT5677_VAD_OUT_SRC_RATE_SFT 11 +#define RT5677_VAD_OUT_SRC_MASK (0x1 << 10) +#define RT5677_VAD_OUT_SRC_SFT 10 +#define RT5677_VAD_SRC_MASK (0x3 << 8) #define RT5677_VAD_SRC_SFT 8 +#define RT5677_VAD_LV_DIFF_MASK (0xff << 0) +#define RT5677_VAD_LV_DIFF_SFT 0 /* DSP InBound Control (0xa3) */ #define RT5677_IB01_SRC_MASK (0x7 << 12) -- 2.22.0.rc1.257.g3120a18244-goog

2 1

[alsa-devel] [PATCH 0/5] ASoC: Intel: fix cross-compilation/COMPILE_TEST issues
by Pierre-Louis Bossart 30 May '19

30 May '19

The Intel code has dependencies that aren't defined on all platforms and prevent COMPILE_TEST from working. The first patch was submitted as an RFC and includes Takashi Reviewed-by tag. The second patch was submitted by Randy Dunlap, I only corrected minor issues to make it more generic. The rest are just simple fixes to remove useless asm/ headers, make sure all machine drivers use COMPILE_TEST and add dependencies on CONFIG_PCI to let the SOF code support COMPILE_TEST. Tested for the following ARCH: x86_64, i386, ia64, arm, arm64, sh, mips, openrisc, sparc, m68k, s390 Pierre-Louis Bossart (4): ASoC: Intel: use common helpers to detect CPUs ASoC: Intel: boards: remove dependency on asm/platform_sst_audio.h ASoC: Intel: boards: Add COMPILE_TEST for new machine drivers ASoC: SOF: add COMPILE_TEST for PCI options Randy Dunlap (1): ASoc: fix sound/soc/intel/skylake/slk-ssp-clk.c build error on IA64 sound/soc/intel/Kconfig | 9 ++ sound/soc/intel/atom/sst/sst_acpi.c | 65 +--------- sound/soc/intel/boards/Kconfig | 7 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 11 +- sound/soc/intel/boards/bytcht_cx2072x.c | 1 - sound/soc/intel/boards/bytcht_da7213.c | 1 - sound/soc/intel/boards/bytcht_es8316.c | 13 +- sound/soc/intel/boards/bytcr_rt5640.c | 16 +-- sound/soc/intel/boards/bytcr_rt5651.c | 17 +-- sound/soc/intel/boards/cht_bsw_rt5645.c | 16 +-- sound/soc/intel/boards/sof_rt5682.c | 11 +- sound/soc/intel/common/soc-intel-quirks.h | 115 ++++++++++++++++++ sound/soc/sof/Kconfig | 2 +- sound/soc/sof/intel/hda.c | 13 +- sound/soc/sof/sof-acpi-dev.c | 57 +-------- sound/soc/sof/sof-pci-dev.c | 4 + 16 files changed, 164 insertions(+), 194 deletions(-) create mode 100644 sound/soc/intel/common/soc-intel-quirks.h -- 2.20.1

2 10

[alsa-devel] [PATCH v3 0/9] ASoC: SOF: stability fixes
by Pierre-Louis Bossart 30 May '19

30 May '19

No new functionality here but fixes to the IPC, memory allocation, workqueues, control, runtime_pm and HDaudio support. These issues were identified during the integration/productization of new platforms. I added a 'Fixes' tag to make sure backports/stable can pick up these patches, in case it's too late for them to land in 5.2 The next batch for SOF will be a set of new capabilities for trace, IPC test, Hdaudio support, they will be submitted after additional tests and once these fixes are reviewed/merged. Thanks! Changes since v2: Removed 3 patches (Hdaudio memory allocation changes, will be resubmitted when unclear system freeze is sorted out + patch already merged by Mark) Corrected merge issue with IPC patch from Guennadi Changes since v1: Feedback from Takashi: use spin_lock_irq instead of spin_lock_irq_save Added Takashi's Reviewed-by tag Guennadi Liakhovetski (1): ASoC: SOF: ipc: fix a race, leading to IPC timeouts Keyon Jie (1): ASoC: SOF: control: correct the copy size for bytes kcontrol put Libin Yang (1): ASoC: SOF: pcm: clear hw_params_upon_resume flag correctly Pierre-Louis Bossart (2): ASoC: SOF: core: fix error handling with the probe workqueue ASoC: SOF: pcm: remove warning - initialize workqueue on open Ranjani Sridharan (2): ASoC: SOF: core: remove DSP after unregistering machine driver ASoC: SOF: core: remove snd_soc_unregister_component in case of error Zhu Yingjiang (2): ASoC: SOF: Intel: hda: fix the hda init chip ASoC: SOF: Intel: hda: use the defined ppcap functions sound/soc/sof/control.c | 9 +-- sound/soc/sof/core.c | 29 +++++++-- sound/soc/sof/intel/bdw.c | 9 ++- sound/soc/sof/intel/byt.c | 10 +-- sound/soc/sof/intel/cnl.c | 4 ++ sound/soc/sof/intel/hda-ctrl.c | 102 +++++++++++++++++++++++++++--- sound/soc/sof/intel/hda-ipc.c | 17 ++++- sound/soc/sof/intel/hda.c | 109 +++++++-------------------------- sound/soc/sof/ipc.c | 13 ---- sound/soc/sof/pcm.c | 8 +-- 10 files changed, 176 insertions(+), 134 deletions(-) base-commit: 188d45fe779eeb8e3521b59fcb12cc48a6f2c203 -- 2.20.1

2 18

[alsa-devel] [PATCH] ASoC: SOF: uapi headers - add missing include for stdint.h
by Jaroslav Kysela 30 May '19

30 May '19

The modified header files depend on types defined in <stdint.h>. Signed-off-by: Jaroslav Kysela <perex(a)perex.cz> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Mark Brown <broonie(a)kernel.org> --- include/uapi/sound/sof/eq.h | 4 ++++ include/uapi/sound/sof/fw.h | 4 ++++ include/uapi/sound/sof/header.h | 4 ++++ include/uapi/sound/sof/manifest.h | 4 ++++ include/uapi/sound/sof/trace.h | 4 ++++ 5 files changed, 20 insertions(+) diff --git a/include/uapi/sound/sof/eq.h b/include/uapi/sound/sof/eq.h index 666c2b6a3229..106d56e357e0 100644 --- a/include/uapi/sound/sof/eq.h +++ b/include/uapi/sound/sof/eq.h @@ -9,6 +9,10 @@ #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_EQ_H__ #define __INCLUDE_UAPI_SOUND_SOF_USER_EQ_H__ +#ifndef __KERNEL__ +#include <stdint.h> +#endif + /* FIR EQ type */ #define SOF_EQ_FIR_IDX_SWITCH 0 diff --git a/include/uapi/sound/sof/fw.h b/include/uapi/sound/sof/fw.h index 1afca973eb09..4f2de19a3b86 100644 --- a/include/uapi/sound/sof/fw.h +++ b/include/uapi/sound/sof/fw.h @@ -13,6 +13,10 @@ #ifndef __INCLUDE_UAPI_SOF_FW_H__ #define __INCLUDE_UAPI_SOF_FW_H__ +#ifndef __KERNEL__ +#include <stdint.h> +#endif + #define SND_SOF_FW_SIG_SIZE 4 #define SND_SOF_FW_ABI 1 #define SND_SOF_FW_SIG "Reef" diff --git a/include/uapi/sound/sof/header.h b/include/uapi/sound/sof/header.h index 7868990b0d6f..6a62ae8f0eb9 100644 --- a/include/uapi/sound/sof/header.h +++ b/include/uapi/sound/sof/header.h @@ -9,6 +9,10 @@ #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__ #define __INCLUDE_UAPI_SOUND_SOF_USER_HEADER_H__ +#ifndef __KERNEL__ +#include <stdint.h> +#endif + /* * Header for all non IPC ABI data. * diff --git a/include/uapi/sound/sof/manifest.h b/include/uapi/sound/sof/manifest.h index 2009ee30fad0..d57aa2bc4764 100644 --- a/include/uapi/sound/sof/manifest.h +++ b/include/uapi/sound/sof/manifest.h @@ -9,6 +9,10 @@ #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_MANIFEST_H__ #define __INCLUDE_UAPI_SOUND_SOF_USER_MANIFEST_H__ +#ifndef __KERNEL__ +#include <stdint.h> +#endif + /* start offset for base FW module */ #define SOF_MAN_ELF_TEXT_OFFSET 0x2000 diff --git a/include/uapi/sound/sof/trace.h b/include/uapi/sound/sof/trace.h index ffa7288a0f16..1652bc08d576 100644 --- a/include/uapi/sound/sof/trace.h +++ b/include/uapi/sound/sof/trace.h @@ -9,6 +9,10 @@ #ifndef __INCLUDE_UAPI_SOUND_SOF_USER_TRACE_H__ #define __INCLUDE_UAPI_SOUND_SOF_USER_TRACE_H__ +#ifndef __KERNEL__ +#include <stdint.h> +#endif + /* * Host system time. * -- 2.20.1

3 3

Re: [alsa-devel] Merging the new firmware files for CA0132 into alsa-firmware
by Jaroslav Kysela 30 May '19

30 May '19

Dne 28. 05. 19 v 16:54 Takashi Iwai napsal(a): > Hi, > > it seems that Connor's previous attempt to put a couple of ca0132 > firmware files into linux-firmware tree didn't go through, > unfortunately. And now I'm thinking of taking them into alsa-firmware > package as a stop-gap. We already distribute other ca0132 firmware > files, so the addition shouldn't be a big problem. > > Jaroslav, what do you think? No problem. The same situation is for the SoC SOF firmware files (drivers are in kernel, firmware files are missing). Perhaps, we can release those files quickly in alsa-firmware and then migrate them slowly to linux-firmware. Jaroslav -- Jaroslav Kysela <perex(a)perex.cz> Linux Sound Maintainer; ALSA Project; Red Hat, Inc.

5 14

[alsa-devel] [PATCH] ALSA: hda: fix: lock reg_lock before calling snd_hdac_bus_update_rirb
by Bard liao 30 May '19

30 May '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> The patch is to fix commit 5e13cf6cd64c (ALSA: hda: add polling mode in snd_hdac_bus_get_response) spin_lock_irq should be called before snd_hdac_bus_update_rirb. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- Takashi, Sorry, I make a mistake on the "ALSA: hda: add polling mode in snd_hdac_bus_get_response" patch. This patch is a fix for that. Thanks, Bard --- sound/hda/hdac_controller.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c index a16ac31bda83..d6a91429c058 100644 --- a/sound/hda/hdac_controller.c +++ b/sound/hda/hdac_controller.c @@ -239,9 +239,9 @@ int snd_hdac_bus_get_response(struct hdac_bus *bus, unsigned int addr, timeout = jiffies + msecs_to_jiffies(1000); for (loopcounter = 0;; loopcounter++) { + spin_lock_irq(&bus->reg_lock); if (bus->polling_mode) snd_hdac_bus_update_rirb(bus); - spin_lock_irq(&bus->reg_lock); if (!bus->rirb.cmds[addr]) { if (res) *res = bus->rirb.res[addr]; /* the last value */ -- 2.17.1

2 1

[alsa-devel] [PATCH 4.9 107/128] ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit c705247136a523488eac806bd357c3e5d79a7acd ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index b9e42b503a377..4f8bdb7650e84 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -75,6 +75,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); -- 2.20.1

1 0

[alsa-devel] [PATCH 4.9 106/128] ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 883087f2b092b..38132143b7d5e 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -119,13 +119,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* -- 2.20.1

1 0

[alsa-devel] [PATCH 4.14 159/193] ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit c705247136a523488eac806bd357c3e5d79a7acd ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index b9e42b503a377..4f8bdb7650e84 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -75,6 +75,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); -- 2.20.1

1 0

[alsa-devel] [PATCH 4.14 158/193] ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 84ef6385736cd..4c6f19ef98b25 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -119,13 +119,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* -- 2.20.1

1 0

[alsa-devel] [PATCH 4.19 215/276] ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit c705247136a523488eac806bd357c3e5d79a7acd ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 7f0fa4b522231..cca33ab7020a4 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -71,6 +71,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); -- 2.20.1

1 0

[alsa-devel] [PATCH 4.19 214/276] ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 191426a6d9adf..30a3d68b5c033 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -118,13 +118,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* -- 2.20.1

1 0

[alsa-devel] [PATCH 5.0 249/346] ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit c705247136a523488eac806bd357c3e5d79a7acd ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 9981668ab5909..040d06b89f00a 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -71,6 +71,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); -- 2.20.1

1 0

[alsa-devel] [PATCH 5.0 248/346] ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 191426a6d9adf..30a3d68b5c033 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -118,13 +118,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* -- 2.20.1

1 0

[alsa-devel] [PATCH 5.1 276/405] ASoC: wcd9335: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit 64b92de9603f22b5455da925ee57268ef7fb4e80 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/codecs/wcd9335.c:5193:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 5183, but without a correspon ding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> (commit_signer:1/11=9%,authored:1/11=9%) Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/codecs/wcd9335.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wcd9335.c b/sound/soc/codecs/wcd9335.c index 981f88a5f6154..a04a7cedd99de 100644 --- a/sound/soc/codecs/wcd9335.c +++ b/sound/soc/codecs/wcd9335.c @@ -5188,6 +5188,7 @@ static int wcd9335_slim_status(struct slim_device *sdev, wcd->slim = sdev; wcd->slim_ifc_dev = of_slim_get_device(sdev->ctrl, ifc_dev_np); + of_node_put(ifc_dev_np); if (!wcd->slim_ifc_dev) { dev_err(dev, "Unable to get SLIM Interface device\n"); return -EINVAL; -- 2.20.1

1 0

[alsa-devel] [PATCH 5.1 275/405] ASoC: fsl_utils: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit c705247136a523488eac806bd357c3e5d79a7acd ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/fsl_utils.c:74:2-8: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 38, but without a corresponding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Timur Tabi <timur(a)kernel.org> Cc: Nicolin Chen <nicoleotsuka(a)gmail.com> Cc: Xiubo Li <Xiubo.Lee(a)gmail.com> Cc: Fabio Estevam <festevam(a)gmail.com> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/fsl_utils.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_utils.c b/sound/soc/fsl/fsl_utils.c index 9981668ab5909..040d06b89f00a 100644 --- a/sound/soc/fsl/fsl_utils.c +++ b/sound/soc/fsl/fsl_utils.c @@ -71,6 +71,7 @@ int fsl_asoc_get_dma_channel(struct device_node *ssi_np, iprop = of_get_property(dma_np, "cell-index", NULL); if (!iprop) { of_node_put(dma_np); + of_node_put(dma_channel_np); return -EINVAL; } *dma_id = be32_to_cpup(iprop); -- 2.20.1

1 0

[alsa-devel] [PATCH 5.1 274/405] ASoC: eukrea-tlv320: fix a leaked reference by adding missing of_node_put
by Greg Kroah-Hartman 30 May '19

30 May '19

[ Upstream commit b820d52e7eed7b30b2dfef5f4213a2bc3cbea6f3 ] The call to of_parse_phandle returns a node pointer with refcount incremented thus it must be explicitly decremented after the last usage. Detected by coccinelle with the following warnings: ./sound/soc/fsl/eukrea-tlv320.c:121:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. ./sound/soc/fsl/eukrea-tlv320.c:127:3-9: ERROR: missing of_node_put; acquired a node pointer with refcount incremented on line 102, but without a correspo nding object release within this function. Signed-off-by: Wen Yang <wen.yang99(a)zte.com.cn> Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/fsl/eukrea-tlv320.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/eukrea-tlv320.c b/sound/soc/fsl/eukrea-tlv320.c index 191426a6d9adf..30a3d68b5c033 100644 --- a/sound/soc/fsl/eukrea-tlv320.c +++ b/sound/soc/fsl/eukrea-tlv320.c @@ -118,13 +118,13 @@ static int eukrea_tlv320_probe(struct platform_device *pdev) if (ret) { dev_err(&pdev->dev, "fsl,mux-int-port node missing or invalid.\n"); - return ret; + goto err; } ret = of_property_read_u32(np, "fsl,mux-ext-port", &ext_port); if (ret) { dev_err(&pdev->dev, "fsl,mux-ext-port node missing or invalid.\n"); - return ret; + goto err; } /* -- 2.20.1

1 0

[alsa-devel] [PATCH 00/22] Some documentation fixes
by Mauro Carvalho Chehab 30 May '19

30 May '19

Fix several warnings and broken links. This series was generated against linux-next, but was rebased to be applied at docs-next. It should apply cleanly on either tree. There's a git tree with all of them applied on the top of docs/docs-next at: https://git.linuxtv.org/mchehab/experimental.git/log/?h=fix_doc_links_v2 Mauro Carvalho Chehab (21): ABI: sysfs-devices-system-cpu: point to the right docs isdn: mISDN: remove a bogus reference to a non-existing doc dt: fix broken references to nand.txt docs: zh_CN: get rid of basic_profiling.txt doc: it_IT: fix reference to magic-number.rst docs: mm: numaperf.rst: get rid of a build warning docs: bpf: get rid of two warnings docs: mark orphan documents as such docs: amd-memory-encryption.rst get rid of warnings gpu: amdgpu: fix broken amdgpu_dma_buf.c references gpu: i915.rst: Fix references to renamed files docs: zh_CN: avoid duplicate citation references docs: vm: hmm.rst: fix some warnings docs: it: license-rules.rst: get rid of warnings docs: gpio: driver.rst: fix a bad tag docs: soundwire: locking: fix tags for a code-block docs: security: trusted-encrypted.rst: fix code-block tag docs: security: core.rst: Fix several warnings docs: net: dpio-driver.rst: fix two codeblock warnings docs: net: sja1105.rst: fix table format docs: fix broken documentation links Otto Sabart (1): mfd: madera: Fix bad reference to pinctrl.txt file .../ABI/testing/sysfs-devices-system-cpu | 3 +- Documentation/accelerators/ocxl.rst | 2 + Documentation/acpi/dsd/leds.txt | 2 +- .../admin-guide/kernel-parameters.rst | 6 +- .../admin-guide/kernel-parameters.txt | 16 ++--- Documentation/admin-guide/mm/numaperf.rst | 5 +- Documentation/admin-guide/ras.rst | 2 +- Documentation/arm/stm32/overview.rst | 2 + .../arm/stm32/stm32f429-overview.rst | 2 + .../arm/stm32/stm32f746-overview.rst | 2 + .../arm/stm32/stm32f769-overview.rst | 2 + .../arm/stm32/stm32h743-overview.rst | 2 + .../arm/stm32/stm32mp157-overview.rst | 2 + Documentation/bpf/btf.rst | 2 + .../bindings/mtd/amlogic,meson-nand.txt | 2 +- .../devicetree/bindings/mtd/gpmc-nand.txt | 2 +- .../devicetree/bindings/mtd/marvell-nand.txt | 2 +- .../devicetree/bindings/mtd/tango-nand.txt | 2 +- .../devicetree/bindings/net/fsl-enetc.txt | 7 +- .../bindings/pci/amlogic,meson-pcie.txt | 2 +- .../regulator/qcom,rpmh-regulator.txt | 2 +- .../devicetree/booting-without-of.txt | 2 +- Documentation/driver-api/gpio/board.rst | 2 +- Documentation/driver-api/gpio/consumer.rst | 2 +- Documentation/driver-api/gpio/driver.rst | 2 +- .../driver-api/soundwire/locking.rst | 4 +- .../firmware-guide/acpi/enumeration.rst | 2 +- .../firmware-guide/acpi/method-tracing.rst | 2 +- Documentation/gpu/amdgpu.rst | 4 +- Documentation/gpu/i915.rst | 6 +- Documentation/gpu/msm-crash-dump.rst | 2 + Documentation/i2c/instantiating-devices | 2 +- Documentation/interconnect/interconnect.rst | 2 + Documentation/laptops/lg-laptop.rst | 2 + .../freescale/dpaa2/dpio-driver.rst | 4 +- Documentation/networking/dsa/sja1105.rst | 6 +- Documentation/powerpc/isa-versions.rst | 2 + Documentation/security/keys/core.rst | 16 +++-- .../security/keys/trusted-encrypted.rst | 4 +- Documentation/sysctl/kernel.txt | 4 +- .../translations/it_IT/process/howto.rst | 2 +- .../it_IT/process/license-rules.rst | 28 ++++---- .../it_IT/process/magic-number.rst | 2 +- .../it_IT/process/stable-kernel-rules.rst | 4 +- .../translations/zh_CN/basic_profiling.txt | 71 ------------------- .../translations/zh_CN/process/4.Coding.rst | 2 +- .../zh_CN/process/management-style.rst | 4 +- .../zh_CN/process/programming-language.rst | 28 ++++---- .../virtual/kvm/amd-memory-encryption.rst | 5 ++ Documentation/virtual/kvm/vcpu-requests.rst | 2 + Documentation/vm/hmm.rst | 9 ++- Documentation/x86/x86_64/5level-paging.rst | 2 +- Documentation/x86/x86_64/boot-options.rst | 4 +- .../x86/x86_64/fake-numa-for-cpusets.rst | 2 +- MAINTAINERS | 6 +- arch/arm/Kconfig | 2 +- arch/arm64/kernel/kexec_image.c | 2 +- arch/powerpc/Kconfig | 2 +- arch/x86/Kconfig | 16 ++--- arch/x86/Kconfig.debug | 2 +- arch/x86/boot/header.S | 2 +- arch/x86/entry/entry_64.S | 2 +- arch/x86/include/asm/bootparam_utils.h | 2 +- arch/x86/include/asm/page_64_types.h | 2 +- arch/x86/include/asm/pgtable_64_types.h | 2 +- arch/x86/kernel/cpu/microcode/amd.c | 2 +- arch/x86/kernel/kexec-bzimage64.c | 2 +- arch/x86/kernel/pci-dma.c | 2 +- arch/x86/mm/tlb.c | 2 +- arch/x86/platform/pvh/enlighten.c | 2 +- drivers/acpi/Kconfig | 10 +-- drivers/isdn/mISDN/dsp_core.c | 2 - drivers/net/ethernet/faraday/ftgmac100.c | 2 +- .../fieldbus/Documentation/fieldbus_dev.txt | 4 +- drivers/vhost/vhost.c | 2 +- include/acpi/acpi_drivers.h | 2 +- include/linux/fs_context.h | 2 +- include/linux/lsm_hooks.h | 2 +- include/linux/mfd/madera/pdata.h | 3 +- mm/Kconfig | 2 +- security/Kconfig | 2 +- tools/include/linux/err.h | 2 +- .../Documentation/stack-validation.txt | 4 +- tools/testing/selftests/x86/protection_keys.c | 2 +- 84 files changed, 183 insertions(+), 212 deletions(-) delete mode 100644 Documentation/translations/zh_CN/basic_profiling.txt -- 2.21.0

1 2

[alsa-devel] [PATCH v2] ASoC: sun4i-codec: fix first delay on Speaker
by Georgii Staroselskii 29 May '19

29 May '19

Allwinner DAC seems to have a delay in the Speaker audio routing. When playing a sound for the first time, the sound gets chopped. On a second play the sound is played correctly. After some time (~5s) the issue gets back. This commit seems to be fixing the same issue as bf14da7 but for another codepath. This is the DTS that was used to debug the problem. &codec { allwinner,pa-gpios = <&r_pio 0 11 GPIO_ACTIVE_HIGH>; /* PL11 */ allwinner,audio-routing = "Speaker", "LINEOUT"; status = "okay"; } Signed-off-by: Georgii Staroselskii <georgii.staroselskii(a)emlid.com> --- sound/soc/sunxi/sun4i-codec.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index f2deffe..9e1f00e 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1320,6 +1320,15 @@ static int sun4i_codec_spk_event(struct snd_soc_dapm_widget *w, gpiod_set_value_cansleep(scodec->gpio_pa, !!SND_SOC_DAPM_EVENT_ON(event)); + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* + * Need a delay to wait for DAC to push the data. 700ms seems + * to be the best compromise not to feel this delay while + * playing a sound. + */ + msleep(700); + } + return 0; } -- 2.7.4

3 4

[alsa-devel] [PATCH] ASoC: Intel: sof-rt5682: fix undefined references with Baytrail-only support
by Pierre-Louis Bossart 29 May '19

29 May '19

The sof-rt5682 machine driver supports both legacy Baytrail devices and more recent ApolloLake/CometLake platforms. When only Baytrail is selected, the compilation fails with the following errors: ERROR: "hdac_hdmi_jack_port_init" [sound/soc/intel/boards/snd-soc-sof_rt5682.ko] undefined! ERROR: "hdac_hdmi_jack_init" [sound/soc/intel/boards/snd-soc-sof_rt5682.ko] undefined! Fix by selecting SND_SOC_HDAC_HDMI unconditionally. The code for HDMI support is not reachable on Baytrail so this change has no functional impact. Fixes: f70abd75b7c6 ("ASoC: Intel: add sof-rt5682 machine driver") Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/boards/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 59e366edc16b..317ee1e8c8c6 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -402,7 +402,7 @@ config SND_SOC_INTEL_SOF_RT5682_MACH (SND_SOC_SOF_BAYTRAIL && X86_INTEL_LPSS) select SND_SOC_RT5682 select SND_SOC_DMIC - select SND_SOC_HDAC_HDMI if SND_SOC_SOF_HDA_COMMON + select SND_SOC_HDAC_HDMI help This adds support for ASoC machine driver for SOF platforms with rt5682 codec. -- 2.20.1

2 1

[alsa-devel] [PATCH] ASoC: cs42xx8: Fix build error with CONFIG_GPIOLIB is not set
by shengjiu.wang＠nxp.com 29 May '19

29 May '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> config: x86_64-randconfig-x000201921-201921 compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: make ARCH=x86_64 sound/soc/codecs/cs42xx8.c: In function ‘cs42xx8_probe’: sound/soc/codecs/cs42xx8.c:472:25: error: implicit declaration of function ‘devm_gpiod_get_optional’; did you mean ‘devm_clk_get_optional’? [-Werror=implicit-function-declaration] cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset", ^~~~~~~~~~~~~~~~~~~~~~~ devm_clk_get_optional sound/soc/codecs/cs42xx8.c:473:8: error: ‘GPIOD_OUT_HIGH’ undeclared (first use in this function); did you mean ‘GPIOF_INIT_HIGH’? GPIOD_OUT_HIGH); ^~~~~~~~~~~~~~ GPIOF_INIT_HIGH sound/soc/codecs/cs42xx8.c:473:8: note: each undeclared identifier is reported only once for each function it appears in sound/soc/codecs/cs42xx8.c:477:2: error: implicit declaration of function ‘gpiod_set_value_cansleep’; did you mean ‘gpio_set_value_cansleep’? [-Werror=implicit-function-declaration] gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); ^~~~~~~~~~~~~~~~~~~~~~~~ gpio_set_value_cansleep Fixes: bfe95dfa4dac ("ASoC: cs42xx8: Add reset gpio handling") Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 3e8dbf63adbe..3bbc62322dfe 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -14,7 +14,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/of_device.h> -#include <linux/of_gpio.h> +#include <linux/gpio/consumer.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <sound/pcm_params.h> -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: core: use component driver name as component name
by Tzung-Bi Shih 29 May '19

29 May '19

fmt_single_name() uses device name to determine component name. If multiple components bind to the same device, the debugfs creation in soc_init_component_debugfs() would fail due to duplicated entity names. Name provided by component driver is unique enough to represent each component. Use component driver name as the component name to avoid name duplication. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- Hi, We got "Failed to create component debugfs directory" in our environment. But the patch does not resolve the issue. (https://mailman.alsa-project.org/pipermail/alsa-devel/2018-February/132391.…) The entity name duplicated when creating the debugfs directory, because there is a device registered 2 components in mt8183_afe_pcm_dev_probe() in sound/soc/mediatek/mt8183/mt8183-afe-pcm.c. sound/soc/soc-core.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7abb017a83f3..2d3520fca613 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3113,7 +3113,10 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, { struct snd_soc_dapm_context *dapm; - component->name = fmt_single_name(dev, &component->id); + if (driver->name) + component->name = kstrdup(driver->name, GFP_KERNEL); + else + component->name = fmt_single_name(dev, &component->id); if (!component->name) { dev_err(dev, "ASoC: Failed to allocate name\n"); return -ENOMEM; -- 2.22.0.rc1.257.g3120a18244-goog

2 1

[alsa-devel] [PATCH 000/145] ASoC: modern dai_link style support
by Kuninori Morimoto 29 May '19

29 May '19

Hi Mark These are for modern dai_link style support patches. [001/145] adds missing modern dai_link style for CPU. and others are switch to modern style from legacy style. Last patch removes legacy style. These are based on mark/for-5.3 + v5.2-rc1 I hope these work well for all sound card, but I can't test for all. Please test these at each sound cards. Kuninori Morimoto (145): ASoC: soc-core: use snd_soc_dai_link_component for CPU ASoC: simple-card: support snd_soc_dai_link_component style for cpu ASoC: soc.h: add sound dai_link connection macro ASoC: mediatek: mt6797-mt6351: use modern dai_link style ASoC: mediatek: mt2701-wm8960: use modern dai_link style ASoC: mediatek: mt2701-cs42448: use modern dai_link style ASoC: mediatek: mt8173-max98090: use modern dai_link style ASoC: mediatek: mt8173-rt5650: use modern dai_link style ASoC: mediatek: mt8173-rt5650-rt5676: use modern dai_link style ASoC: mediatek: mt8173-rt5650-rt5514: use modern dai_link style ASoC: mediatek: mt8183-da7219-max98357: use modern dai_link style ASoC: mediatek: mt8183-mt6358-ts3a227-max98357: use modern dai_link style ASoC: samsung: bells: use modern dai_link style ASoC: samsung: smdk_wm8580: use modern dai_link style ASoC: samsung: smdk_wm8994: use modern dai_link style ASoC: samsung: smdk_wm8994pcm: use modern dai_link style ASoC: samsung: smdk_spdif: use modern dai_link style ASoC: samsung: h1940_uda1380: use modern dai_link style ASoC: samsung: arndale_rt5631: use modern dai_link style ASoC: samsung: tobermory: use modern dai_link style ASoC: samsung: tm2_wm5110: use modern dai_link style ASoC: samsung: speyside: use modern dai_link style ASoC: samsung: snow: use modern dai_link style ASoC: samsung: smartq_wm8987: use modern dai_link style ASoC: samsung: s3c24xx_uda134x: use modern dai_link style ASoC: samsung: s3c24xx_simtec_tlv320aic23: use modern dai_link style ASoC: samsung: s3c24xx_simtec_hermes: use modern dai_link style ASoC: samsung: rx1950_uda1380: use modern dai_link style ASoC: samsung: odroid: use modern dai_link style ASoC: samsung: neo1973_wm8753: use modern dai_link style ASoC: samsung: lowland: use modern dai_link style ASoC: samsung: littlemill: use modern dai_link style ASoC: samsung: jive_wm8750: use modern dai_link style ASoC: pxa: brownstone: use modern dai_link style ASoC: pxa: corgi: use modern dai_link style ASoC: pxa: e740_wm9705: use modern dai_link style ASoC: pxa: e750_wm9705: use modern dai_link style ASoC: pxa: e800_wm9712: use modern dai_link style ASoC: pxa: em-x270: use modern dai_link style ASoC: pxa: hx4700: use modern dai_link style ASoC: pxa: imote2: use modern dai_link style ASoC: pxa: magician: use modern dai_link style ASoC: pxa: mioa701_wm9713: use modern dai_link style ASoC: pxa: palm27x: use modern dai_link style ASoC: pxa: zylonite: use modern dai_link style ASoC: pxa: z2: use modern dai_link style ASoC: pxa: ttc-dkb: use modern dai_link style ASoC: pxa: tosa: use modern dai_link style ASoC: pxa: spitz: use modern dai_link style ASoC: pxa: poodle: use modern dai_link style ASoC: kirkwood: armada-370-db: use modern dai_link style ASoC: ti: n810: use modern dai_link style ASoC: ti: osk5912: use modern dai_link style ASoC: ti: ams-delta: use modern dai_link style ASoC: ti: omap3pandora: use modern dai_link style ASoC: ti: davinci-evm: use modern dai_link style ASoC: ti: omap-hdmi: use modern dai_link style ASoC: ti: omap-abe-twl6040: use modern dai_link style ASoC: ti: omap-twl4030: use modern dai_link style ASoC: ti: rx51: use modern dai_link style ASoC: sh: migor: use modern dai_link style ASoC: sh: sh7760-ac97: use modern dai_link style ASoC: amd: acp-da7219-max98357a: use modern dai_link style ASoC: amd: acp-rt5645: use modern dai_link style ASoC: atmel: mikroe-proto: use modern dai_link style ASoC: atmel: sam9x5_wm8731: use modern dai_link style ASoC: atmel: sam9g20_wm8731: use modern dai_link style ASoC: atmel: tse850-pcm5142: use modern dai_link style ASoC: atmel: atmel-wm8904: use modern dai_link style ASoC: atmel: atmel-pdmic: use modern dai_link style ASoC: atmel: atmel-classd: use modern dai_link style ASoC: au1x: db1000: use modern dai_link style ASoC: au1x: db1200: use modern dai_link style ASoC: cirrus: edb93xx: use modern dai_link style ASoC: cirrus: simone: use modern dai_link style ASoC: cirrus: snappercl15: use modern dai_link style ASoC: fsl: efika-audio-fabric: use modern dai_link style ASoC: fsl: imx-spdif: use modern dai_link style ASoC: fsl: mx27vis-aic32x4: use modern dai_link style ASoC: fsl: phycore-ac97: use modern dai_link style ASoC: fsl: wm1133-ev1: use modern dai_link style ASoC: fsl: fsl-asoc-card: use modern dai_link style ASoC: fsl: mpc8610_hpcd/p1022_ds/p1022_rdk: use modern dai_link style ASoC: fsl: imx-es8328: use modern dai_link style ASoC: fsl: eukrea-tlv320: use modern dai_link style ASoC: fsl: imx-sgtl5000: use modern dai_link style ASoC: fsl: imx-mc13783: use modern dai_link style ASoC: fsl: pcm030-audio-fabric: use modern dai_link style ASoC: fsl: imx-audmix: use modern dai_link style ASoC: jz4740: qi_lb60: use modern dai_link style ASoC: meson: axg-card: use modern dai_link style ASoC: mxs: mxs-sgtl5000: use modern dai_link style ASoC: nuc900: nuc900-audio: use modern dai_link style ASoC: qcom: storm: use modern dai_link style ASoC: qcom: common: use modern dai_link style ASoC: qcom: apq8016_sbc: use modern dai_link style ASoC: rockchip: rk3399_gru_sound: use modern dai_link style ASoC: rockchip: rk3288_hdmi_analog: use modern dai_link style ASoC: rockchip: rockchip_rt5645: use modern dai_link style ASoC: rockchip: rockchip_max98090: use modern dai_link style ASoC: sirf: sirf-audio: use modern dai_link style ASoC: ux500: mop500: use modern dai_link style ASoC: sunxi: sun4i-codec: use modern dai_link style ASoC: txx9: txx9aclc-generic: use modern dai_link style ASoC: tegra: tegra_rt5677: use modern dai_link style ASoC: tegra: tegra_wm8753: use modern dai_link style ASoC: tegra: trimslice: use modern dai_link style ASoC: tegra: tegra_alc5632: use modern dai_link style ASoC: tegra: tegra_sgtl5000: use modern dai_link style ASoC: tegra: tegra_rt5640: use modern dai_link style ASoC: tegra: tegra_wm8903: use modern dai_link style ASoC: tegra: tegra_wm9712: use modern dai_link style ASoC: tegra: tegra_max98090: use modern dai_link style ASoC: soc-topology: use modern dai_link style ASoC: vc4: vc4_htmi: use modern dai_link style ASoC: sof: use modern dai_link style ASoC: Intel: bdw-rt5677: use modern dai_link style ASoC: Intel: haswell: use modern dai_link style ASoC: Intel: broadwell: use modern dai_link style ASoC: Intel: bxt_da7219_max98357a: use modern dai_link style ASoC: Intel: bxt_rt298: use modern dai_link style ASoC: Intel: byt-rt5640: use modern dai_link style ASoC: Intel: byt-max98090: use modern dai_link style ASoC: Intel: bytcht_es8316: use modern dai_link style ASoC: Intel: bytcht_nocodec: use modern dai_link style ASoC: Intel: bytcht_da7213: use modern dai_link style ASoC: Intel: bytcr_rt5651: use modern dai_link style ASoC: Intel: bytcr_rt5640: use modern dai_link style ASoC: Intel: cht_bsw_rt5645: use modern dai_link style ASoC: Intel: cht_bsw_max98090_ti: use modern dai_link style ASoC: Intel: cht_bsw_rt5672: use modern dai_link style ASoC: Intel: cht_bsw_nau8824: use modern dai_link style ASoC: Intel: kbl_rt5660: use modern dai_link style ASoC: Intel: kbl_rt5663_max98927: use modern dai_link style ASoC: Intel: kbl_rt5663_rt5514_max98927: use modern dai_link style ASoC: Intel: kbl_da7219_max98927: use modern dai_link style ASoC: Intel: kbl_da7219_max98357a: use modern dai_link style ASoC: Intel: skl_nau88l25_ssm4567: use modern dai_link style ASoC: Intel: skl_rt286: use modern dai_link style ASoC: Intel: skl_nau88l25_max98357a: use modern dai_link style ASoC: Intel: skl-pcm: use modern dai_link style ASoC: Intel: skl_hda_dsp_common: use modern dai_link style ASoC: Intel: glk_rt5682_max98357a: use modern dai_link style ASoC: Intel: sof_rt5682: use modern dai_link style ASoC: soc-core: remove legacy style dai_link drivers/gpu/drm/vc4/vc4_hdmi.c | 19 +- include/sound/simple_card_utils.h | 20 +- include/sound/soc.h | 153 +++++++++------ sound/soc/amd/acp-da7219-max98357a.c | 42 ++-- sound/soc/amd/acp-rt5645.c | 21 +- sound/soc/atmel/atmel-classd.c | 21 +- sound/soc/atmel/atmel-pdmic.c | 21 +- sound/soc/atmel/atmel_wm8904.c | 17 +- sound/soc/atmel/mikroe-proto.c | 21 +- sound/soc/atmel/sam9g20_wm8731.c | 22 ++- sound/soc/atmel/sam9x5_wm8731.c | 20 +- sound/soc/atmel/tse850-pcm5142.c | 13 +- sound/soc/au1x/db1000.c | 10 +- sound/soc/au1x/db1200.c | 50 +++-- sound/soc/cirrus/edb93xx.c | 10 +- sound/soc/cirrus/simone.c | 10 +- sound/soc/cirrus/snappercl15.c | 11 +- sound/soc/fsl/efika-audio-fabric.c | 22 ++- sound/soc/fsl/eukrea-tlv320.c | 19 +- sound/soc/fsl/fsl-asoc-card.c | 47 +++-- sound/soc/fsl/fsl_utils.c | 2 +- sound/soc/fsl/imx-audmix.c | 45 ++++- sound/soc/fsl/imx-es8328.c | 23 ++- sound/soc/fsl/imx-mc13783.c | 10 +- sound/soc/fsl/imx-sgtl5000.c | 23 ++- sound/soc/fsl/imx-spdif.c | 20 +- sound/soc/fsl/mpc8610_hpcd.c | 33 +++- sound/soc/fsl/mx27vis-aic32x4.c | 11 +- sound/soc/fsl/p1022_ds.c | 36 +++- sound/soc/fsl/p1022_rdk.c | 35 +++- sound/soc/fsl/pcm030-audio-fabric.c | 20 +- sound/soc/fsl/phycore-ac97.c | 10 +- sound/soc/fsl/wm1133-ev1.c | 10 +- sound/soc/generic/audio-graph-card.c | 30 +-- sound/soc/generic/simple-card-utils.c | 21 +- sound/soc/generic/simple-card.c | 39 ++-- sound/soc/intel/boards/bdw-rt5677.c | 22 ++- sound/soc/intel/boards/broadwell.c | 46 +++-- sound/soc/intel/boards/bxt_da7219_max98357a.c | 140 ++++++++------ sound/soc/intel/boards/bxt_rt298.c | 129 +++++++------ sound/soc/intel/boards/byt-max98090.c | 10 +- sound/soc/intel/boards/byt-rt5640.c | 10 +- sound/soc/intel/boards/bytcht_da7213.c | 37 ++-- sound/soc/intel/boards/bytcht_es8316.c | 38 ++-- sound/soc/intel/boards/bytcht_nocodec.c | 30 +-- sound/soc/intel/boards/bytcr_rt5640.c | 43 +++-- sound/soc/intel/boards/bytcr_rt5651.c | 40 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 32 ++-- sound/soc/intel/boards/cht_bsw_nau8824.c | 41 ++-- sound/soc/intel/boards/cht_bsw_rt5645.c | 42 ++-- sound/soc/intel/boards/cht_bsw_rt5672.c | 37 ++-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 137 +++++++------ sound/soc/intel/boards/haswell.c | 46 +++-- sound/soc/intel/boards/kbl_da7219_max98357a.c | 114 ++++++----- sound/soc/intel/boards/kbl_da7219_max98927.c | 212 +++++++++------------ sound/soc/intel/boards/kbl_rt5660.c | 83 ++++---- sound/soc/intel/boards/kbl_rt5663_max98927.c | 185 +++++++++--------- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 126 ++++++------ sound/soc/intel/boards/skl_hda_dsp_common.c | 66 ++++--- sound/soc/intel/boards/skl_hda_dsp_generic.c | 4 +- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 120 +++++++----- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 132 +++++++------ sound/soc/intel/boards/skl_rt286.c | 118 ++++++------ sound/soc/intel/boards/sof_rt5682.c | 37 ++-- sound/soc/intel/skylake/skl-pcm.c | 2 +- sound/soc/jz4740/qi_lb60.c | 10 +- sound/soc/kirkwood/armada-370-db.c | 42 ++-- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 95 ++++++--- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 34 ++-- sound/soc/mediatek/mt6797/mt6797-mt6351.c | 115 +++++++---- sound/soc/mediatek/mt8173/mt8173-max98090.c | 34 ++-- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 51 ++--- sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 85 +++++---- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 74 +++---- sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c | 182 ++++++++++-------- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 168 ++++++++++------ sound/soc/meson/axg-card.c | 57 ++++-- sound/soc/mxs/mxs-sgtl5000.c | 27 ++- sound/soc/nuc900/nuc900-audio.c | 10 +- sound/soc/pxa/brownstone.c | 10 +- sound/soc/pxa/corgi.c | 10 +- sound/soc/pxa/e740_wm9705.c | 20 +- sound/soc/pxa/e750_wm9705.c | 20 +- sound/soc/pxa/e800_wm9712.c | 21 +- sound/soc/pxa/em-x270.c | 20 +- sound/soc/pxa/hx4700.c | 10 +- sound/soc/pxa/imote2.c | 11 +- sound/soc/pxa/magician.c | 22 ++- sound/soc/pxa/mioa701_wm9713.c | 20 +- sound/soc/pxa/palm27x.c | 20 +- sound/soc/pxa/poodle.c | 10 +- sound/soc/pxa/spitz.c | 10 +- sound/soc/pxa/tosa.c | 20 +- sound/soc/pxa/ttc-dkb.c | 10 +- sound/soc/pxa/z2.c | 10 +- sound/soc/pxa/zylonite.c | 30 +-- sound/soc/qcom/apq8016_sbc.c | 19 +- sound/soc/qcom/common.c | 33 +++- sound/soc/qcom/storm.c | 17 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 21 +- sound/soc/rockchip/rk3399_gru_sound.c | 51 +++-- sound/soc/rockchip/rockchip_max98090.c | 17 +- sound/soc/rockchip/rockchip_rt5645.c | 33 ++-- sound/soc/samsung/arndale_rt5631.c | 25 ++- sound/soc/samsung/bells.c | 87 +++++---- sound/soc/samsung/h1940_uda1380.c | 10 +- sound/soc/samsung/jive_wm8750.c | 10 +- sound/soc/samsung/littlemill.c | 19 +- sound/soc/samsung/lowland.c | 26 ++- sound/soc/samsung/neo1973_wm8753.c | 18 +- sound/soc/samsung/odroid.c | 29 ++- sound/soc/samsung/rx1950_uda1380.c | 11 +- sound/soc/samsung/s3c24xx_simtec_hermes.c | 11 +- sound/soc/samsung/s3c24xx_simtec_tlv320aic23.c | 11 +- sound/soc/samsung/s3c24xx_uda134x.c | 10 +- sound/soc/samsung/smartq_wm8987.c | 10 +- sound/soc/samsung/smdk_spdif.c | 10 +- sound/soc/samsung/smdk_wm8580.c | 20 +- sound/soc/samsung/smdk_wm8994.c | 30 +-- sound/soc/samsung/smdk_wm8994pcm.c | 10 +- sound/soc/samsung/snow.c | 40 ++-- sound/soc/samsung/speyside.c | 26 ++- sound/soc/samsung/tm2_wm5110.c | 42 ++-- sound/soc/samsung/tobermory.c | 10 +- sound/soc/sh/migor.c | 10 +- sound/soc/sh/sh7760-ac97.c | 11 +- sound/soc/sirf/sirf-audio.c | 13 +- sound/soc/soc-core.c | 149 +++------------ sound/soc/soc-topology.c | 22 ++- sound/soc/sof/nocodec.c | 21 +- sound/soc/sof/topology.c | 20 +- sound/soc/sunxi/sun4i-codec.c | 20 +- sound/soc/tegra/tegra_alc5632.c | 37 ++-- sound/soc/tegra/tegra_max98090.c | 17 +- sound/soc/tegra/tegra_rt5640.c | 17 +- sound/soc/tegra/tegra_rt5677.c | 37 ++-- sound/soc/tegra/tegra_sgtl5000.c | 37 ++-- sound/soc/tegra/tegra_wm8753.c | 17 +- sound/soc/tegra/tegra_wm8903.c | 17 +- sound/soc/tegra/tegra_wm9712.c | 14 +- sound/soc/tegra/trimslice.c | 19 +- sound/soc/ti/ams-delta.c | 10 +- sound/soc/ti/davinci-evm.c | 105 ++++++---- sound/soc/ti/n810.c | 11 +- sound/soc/ti/omap-abe-twl6040.c | 36 +++- sound/soc/ti/omap-hdmi.c | 20 +- sound/soc/ti/omap-twl4030.c | 36 ++-- sound/soc/ti/omap3pandora.c | 20 +- sound/soc/ti/osk5912.c | 11 +- sound/soc/ti/rx51.c | 23 ++- sound/soc/txx9/txx9aclc-generic.c | 10 +- sound/soc/ux500/mop500.c | 36 ++-- 152 files changed, 3382 insertions(+), 2344 deletions(-) -- 2.7.4

3 149

[alsa-devel] [PATCH 1/6] device property: Add new array helper
by Charles Keepax 29 May '19

29 May '19

It is fairly common to want to read an integer array property that is composed of an unknown number of fixed size integer groups. For example, say each group consists of three values which correspond to the settings for one input on the device and the driver supports several chips with different numbers of inputs. Add a new helper function to provide this functionality, it differs for the existing helpers in that it allows reading a smaller number of values than the full array size and checks that the number of values read is a multiple of the group size. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- drivers/base/property.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ include/linux/property.h | 2 ++ 2 files changed, 50 insertions(+) diff --git a/drivers/base/property.c b/drivers/base/property.c index 348b37e64944c..656d21e01a648 100644 --- a/drivers/base/property.c +++ b/drivers/base/property.c @@ -133,6 +133,54 @@ int device_property_read_u32_array(struct device *dev, const char *propname, EXPORT_SYMBOL_GPL(device_property_read_u32_array); /** + * device_property_read_u32_2darray - return a 2d u32 array property of a device + * @dev: Device to get the property of + * @propname: Name of the property + * @val: The values are stored here or %NULL to return the number of values + * @nval: Size of the @val array + * @multiple: Number of entries in each block of data + * + * Function reads an array of u32 properties split up into fixed size + * sub-groups, with @propname from the device firmware description and + * stores them to @val if found. + * + * Return: Number of values read + * %0 if the property was not found, + * %-EINVAL if given arguments are not valid, + * %-ENODATA if the property does not have a value, + * %-EPROTO if the property is not an array of numbers, + * %-EOVERFLOW if the size of the property is not as expected. + * %-ENXIO if no suitable firmware interface is present. + */ +int device_property_read_u32_2darray(struct device *dev, const char *propname, + u32 *val, size_t nval, int multiple) +{ + int n, ret; + + n = device_property_read_u32_array(dev, propname, NULL, 0); + if (n == -EINVAL) { + return 0; /* missing, ignore */ + } else if (n < 0) { + dev_warn(dev, "%s malformed (%d)\n", propname, n); + return n; + } else if ((n % multiple) != 0) { + dev_warn(dev, "%s not a multiple of %d entries\n", + propname, multiple); + return -EOVERFLOW; + } + + if (n > nval) + n = nval; + + ret = device_property_read_u32_array(dev, propname, val, n); + if (ret < 0) + return ret; + else + return n; +} +EXPORT_SYMBOL_GPL(device_property_read_u32_2darray); + +/** * device_property_read_u64_array - return a u64 array property of a device * @dev: Device to get the property of * @propname: Name of the property diff --git a/include/linux/property.h b/include/linux/property.h index a29369c89e6ef..854867f0d139f 100644 --- a/include/linux/property.h +++ b/include/linux/property.h @@ -43,6 +43,8 @@ int device_property_read_u16_array(struct device *dev, const char *propname, u16 *val, size_t nval); int device_property_read_u32_array(struct device *dev, const char *propname, u32 *val, size_t nval); +int device_property_read_u32_2darray(struct device *dev, const char *propname, + u32 *val, size_t nval, int multiple); int device_property_read_u64_array(struct device *dev, const char *propname, u64 *val, size_t nval); int device_property_read_string_array(struct device *dev, const char *propname, -- 2.11.0

1 5

Re: [alsa-devel] Dell XPS 9360 continuous hissing sound
by Paulo Fidalgo 29 May '19

29 May '19

Since I don't know if you are waiting for the command output or not I will provide it: After a reboot with the headset unplugged: amixer -c0 cget numid=10 numid=10,iface=MIXER,name='Headphone Mic Boost Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0 : values=0,0 | dBscale-min=0.00dB,step=10.00dB,mute=0 After a reboot with the headset plugged I have the same value: amixer -c0 cget numid=10 numid=10,iface=MIXER,name='Headphone Mic Boost Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0 : values=0,0 | dBscale-min=0.00dB,step=10.00dB,mute=0 Also when the headset is plugged the mic, seems to not be detected, as it keeps with the internal mic of the laptop, but I will create another thread for that. Kind regards, Paulo Fidalgo On Thu, 16 May 2019 at 15:45, Paulo Fidalgo <paulo.fidalgo.pt(a)gmail.com> wrote: > > Since I don't know if you are waiting for the command output or not I > will provide it: > > After a reboot with the headset unplugged: > > amixer -c0 cget numid=10 > numid=10,iface=MIXER,name='Headphone Mic Boost Volume' > ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0 > : values=0,0 > | dBscale-min=0.00dB,step=10.00dB,mute=0 > > After a reboot with the headset plugged I have the same value: > > amixer -c0 cget numid=10 > numid=10,iface=MIXER,name='Headphone Mic Boost Volume' > ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0 > : values=0,0 > | dBscale-min=0.00dB,step=10.00dB,mute=0 > > > Also when the headset is plugged the mic, seems to not be detected, as > it keeps with the internal mic of the laptop, but I will create > another thread for that. > > Kind regards, > > Paulo Fidalgo > > > On Fri, 22 Mar 2019 at 16:54, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > On Fri, 22 Mar 2019 17:07:15 +0100, > > Paulo Fidalgo wrote: > > > > > > Strangely the command returns amixer: Unable to find simple control 'numid=10',0 > > > So I've runned "amixer -c0" and you can find the output in before.txt > > > > Ah sorry, it was a typo: use "cget" instead of "get". > > > > > > > Then I've runned amixer -c 0 cset 'numid=10' 1 > > > and again the "amixer -c0" and you can find the output in after.txt > > > > > > Also I've notice that the sound appears after I playing a music on > > > computer and goes a few (like 10 ) seconds after the music stopped. > > > > It sounds like a state after runtime PM. > > > > > > Takashi > > > > > > > > On Fri, 22 Mar 2019 at 15:37, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > > > On Fri, 22 Mar 2019 12:40:22 +0100, > > > > Paulo Fidalgo wrote: > > > > > > > > > > Hi! > > > > > > > > > > After a bug report on Fedora > > > > > (https://bugzilla.redhat.com/show_bug.cgi?id=1677353) > > > > > > > > > > ---- > > > > > When I start t he computer I hears a continuous background noise. The > > > > > problem is also described here: > > > > > https://wiki.archlinux.org/index.php/Dell_XPS_13_(9360)#Continuous_hissing_… > > > > > > > > > > I need to issue the command `amixer -c 0 cset 'numid=10' 1` to solve > > > > > the problem, but every time I reboot the computer I need to run it > > > > > again. > > > > > This has been happening for quite some time, but I've been post-poning > > > > > the bug report. > > > > > > > > What is the original value before setting this amixer command above? > > > > You can get via "amixer -c0 get numid=10'. > > > > > > > > Does the noise again when you restore the value after setting to 1 in > > > > the above? > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > Current kernel: 4.20.7-200.fc29.x86_64 > > > > > > > > > > Alsa information: > > > > > http://www.alsa-project.org/db/?f=df4834c53335709abb521c767e7b1da2f4371744 > > > > > ---- > > > > > They advise me to contact you, to see if we can get a solution on this. > > > > > > > > > > Note: The computer also does not recognise (switch to) when I plug an headset. > > > > > > > > > > If you need more information, I'm eager to help diagnose and help to > > > > > solve this, but I have no Kernel programming skills, other than that > > > > > count on me. > > > > > > > > > > Kind regards, > > > > > > > > > > Paulo Fidalgo > > > > > > > > Simple mixer control 'Master',0 > > > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > > > Playback channels: Mono > > > Limits: Playback 0 - 87 > > > Mono: Playback 87 [100%] [0.00dB] [on] > > > Simple mixer control 'Headphone',0 > > > Capabilities: pvolume pswitch > > > Playback channels: Front Left - Front Right > > > Limits: Playback 0 - 87 > > > Mono: > > > Front Left: Playback 87 [100%] [0.00dB] [on] > > > Front Right: Playback 87 [100%] [0.00dB] [on] > > > Simple mixer control 'Headphone Mic',0 > > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > > Capture exclusive group: 0 > > > Capture channels: Mono > > > Mono: Capture [off] > > > Simple mixer control 'Headphone Mic Boost',0 > > > Capabilities: volume > > > Playback channels: Front Left - Front Right > > > Capture channels: Front Left - Front Right > > > Limits: 0 - 3 > > > Front Left: 1 [33%] [10.00dB] > > > Front Right: 1 [33%] [10.00dB] > > > Simple mixer control 'Speaker',0 > > > Capabilities: pvolume pswitch > > > Playback channels: Front Left - Front Right > > > Limits: Playback 0 - 87 > > > Mono: > > > Front Left: Playback 0 [0%] [-65.25dB] [off] > > > Front Right: Playback 0 [0%] [-65.25dB] [off] > > > Simple mixer control 'PCM',0 > > > Capabilities: pvolume > > > Playback channels: Front Left - Front Right > > > Limits: Playback 0 - 255 > > > Mono: > > > Front Left: Playback 255 [100%] [0.00dB] > > > Front Right: Playback 255 [100%] [0.00dB] > > > Simple mixer control 'IEC958',0 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [off] > > > Simple mixer control 'IEC958',1 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [on] > > > Simple mixer control 'IEC958',2 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [on] > > > Simple mixer control 'IEC958',3 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [on] > > > Simple mixer control 'IEC958',4 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [on] > > > Simple mixer control 'Capture',0 > > > Capabilities: cvolume cswitch > > > Capture channels: Front Left - Front Right > > > Limits: Capture 0 - 63 > > > Front Left: Capture 63 [100%] [30.00dB] [on] > > > Front Right: Capture 63 [100%] [30.00dB] [on] > > > Simple mixer control 'Auto-Mute Mode',0 > > > Capabilities: enum > > > Items: 'Disabled' 'Enabled' > > > Item0: 'Disabled' > > > Simple mixer control 'Headset Mic',0 > > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > > Capture exclusive group: 0 > > > Capture channels: Mono > > > Mono: Capture [on] > > > Simple mixer control 'Headset Mic Boost',0 > > > Capabilities: volume > > > Playback channels: Front Left - Front Right > > > Capture channels: Front Left - Front Right > > > Limits: 0 - 3 > > > Front Left: 1 [33%] [10.00dB] > > > Front Right: 1 [33%] [10.00dB] > > > Simple mixer control 'Internal Mic',0 > > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > > Capture exclusive group: 0 > > > Capture channels: Mono > > > Mono: Capture [off] > > > Simple mixer control 'Internal Mic Boost',0 > > > Capabilities: volume > > > Playback channels: Front Left - Front Right > > > Capture channels: Front Left - Front Right > > > Limits: 0 - 3 > > > Front Left: 0 [0%] [0.00dB] > > > Front Right: 0 [0%] [0.00dB] > > > Simple mixer control 'Master',0 > > > Capabilities: pvolume pvolume-joined pswitch pswitch-joined > > > Playback channels: Mono > > > Limits: Playback 0 - 87 > > > Mono: Playback 83 [95%] [-3.00dB] [on] > > > Simple mixer control 'Headphone',0 > > > Capabilities: pvolume pswitch > > > Playback channels: Front Left - Front Right > > > Limits: Playback 0 - 87 > > > Mono: > > > Front Left: Playback 87 [100%] [0.00dB] [on] > > > Front Right: Playback 87 [100%] [0.00dB] [on] > > > Simple mixer control 'Headphone Mic',0 > > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > > Capture exclusive group: 0 > > > Capture channels: Mono > > > Mono: Capture [off] > > > Simple mixer control 'Headphone Mic Boost',0 > > > Capabilities: volume > > > Playback channels: Front Left - Front Right > > > Capture channels: Front Left - Front Right > > > Limits: 0 - 3 > > > Front Left: 0 [0%] [0.00dB] > > > Front Right: 0 [0%] [0.00dB] > > > Simple mixer control 'Speaker',0 > > > Capabilities: pvolume pswitch > > > Playback channels: Front Left - Front Right > > > Limits: Playback 0 - 87 > > > Mono: > > > Front Left: Playback 0 [0%] [-65.25dB] [off] > > > Front Right: Playback 0 [0%] [-65.25dB] [off] > > > Simple mixer control 'PCM',0 > > > Capabilities: pvolume > > > Playback channels: Front Left - Front Right > > > Limits: Playback 0 - 255 > > > Mono: > > > Front Left: Playback 255 [100%] [0.00dB] > > > Front Right: Playback 255 [100%] [0.00dB] > > > Simple mixer control 'IEC958',0 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [off] > > > Simple mixer control 'IEC958',1 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [on] > > > Simple mixer control 'IEC958',2 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [on] > > > Simple mixer control 'IEC958',3 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [on] > > > Simple mixer control 'IEC958',4 > > > Capabilities: pswitch pswitch-joined > > > Playback channels: Mono > > > Mono: Playback [on] > > > Simple mixer control 'Capture',0 > > > Capabilities: cvolume cswitch > > > Capture channels: Front Left - Front Right > > > Limits: Capture 0 - 63 > > > Front Left: Capture 63 [100%] [30.00dB] [on] > > > Front Right: Capture 63 [100%] [30.00dB] [on] > > > Simple mixer control 'Auto-Mute Mode',0 > > > Capabilities: enum > > > Items: 'Disabled' 'Enabled' > > > Item0: 'Disabled' > > > Simple mixer control 'Headset Mic',0 > > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > > Capture exclusive group: 0 > > > Capture channels: Mono > > > Mono: Capture [on] > > > Simple mixer control 'Headset Mic Boost',0 > > > Capabilities: volume > > > Playback channels: Front Left - Front Right > > > Capture channels: Front Left - Front Right > > > Limits: 0 - 3 > > > Front Left: 1 [33%] [10.00dB] > > > Front Right: 1 [33%] [10.00dB] > > > Simple mixer control 'Internal Mic',0 > > > Capabilities: cswitch cswitch-joined cswitch-exclusive > > > Capture exclusive group: 0 > > > Capture channels: Mono > > > Mono: Capture [off] > > > Simple mixer control 'Internal Mic Boost',0 > > > Capabilities: volume > > > Playback channels: Front Left - Front Right > > > Capture channels: Front Left - Front Right > > > Limits: 0 - 3 > > > Front Left: 0 [0%] [0.00dB] > > > Front Right: 0 [0%] [0.00dB]

2 6

[alsa-devel] Question about aux device
by Kuninori Morimoto 29 May '19

29 May '19

Hi ALSA ML I wonder about aux device probe method. Does someone know about it ? (1) snd_soc_instantiate_card() binds all (normal) components by soc_bind_dai_link() ~~~~~~~~~~~~~~~~~ (2) And initialize/probe all binded (normal) components after that. Here, in this operation, it will calls soc_probe_component(). ~~~~~~~~~~~~~~~~~~~ (3) After that, auxiliary component is probed by using soc_probe_component() ~~~~~~~~~~~~~~~~~~~ (4) And, auxiliary component is binded by soc_bind_dai_link() after that. ~~~~~~~~~~~~~~~~~ static int snd_soc_instantiate_card(struct snd_soc_card *card) { ... /* probe normal components here */ for_each_card_prelinks(card, i, dai_link) { (1)=> ret = soc_bind_dai_link(card, dai_link); if (ret != 0) goto probe_end; } ... /* probe all components used by DAI links on this card */ for_each_comp_order(order) { for_each_card_rtds(card, rtd) { (2)=> ret = soc_probe_link_components(card, rtd, order); ... } } } /* probe auxiliary components */ (3)=> ret = soc_probe_aux_devices(card); ... for_each_card_links(card, dai_link) { ... (4)=> ret = soc_bind_dai_link(card, dai_link); ... } } 1st question is it seems the order of (1)/(2) and (3)/(4) are different. are there some reason about it ? or no reason ? 2nd question is can we merge these ? (1)+(4) and (2)+(3). Or do we have some reason ? For example (3)/(4) need to be called after xxxx. Thank you for your help !! Best regards --- Kuninori Morimoto

1 2

[alsa-devel] Question about card names
by Kuninori Morimoto 29 May '19

29 May '19

Hi ALSA ML snd_soc_instantiate_card() setups card->snd_card's "shortname", "longname", "driver" names. (= 1,2,3). But, it cares only card->snd_card->driver (= 4). I think we need to care about all naming, but why only driver ?? static int snd_soc_instantiate_card(struct snd_soc_card *card) { ... 1=> snprintf(card->snd_card->shortname, sizeof(card->snd_card->shortname), "%s", card->name); 2=> snprintf(card->snd_card->longname, sizeof(card->snd_card->longname), "%s", card->long_name ? card->long_name : card->name); 3=> snprintf(card->snd_card->driver, sizeof(card->snd_card->driver), "%s", card->driver_name ? card->driver_name : card->name); for (i = 0; i < ARRAY_SIZE(card->snd_card->driver); i++) { switch (card->snd_card->driver[i]) { x=> case '_': x=> case '-': case '\0': break; default: if (!isalnum(card->snd_card->driver[i])) 4=> card->snd_card->driver[i] = '_'; break; } } ... } And, it seems in this loop, it breaks check if it finds "_" or "-" (= x). If my understanding was correct, the naming "abcd??efg" will be fixed to "abcd__efg", but, it do nothing if name was for example "-abcd??efg". Is this bug ?? Thank you for your help !! Best regards --- Kuninori Morimoto

2 3

[alsa-devel] [PATCH] ALSA: hda/realtek - Improve the headset mic for Acer Aspire laptops
by Hui Wang 29 May '19

29 May '19

We met another Acer Aspire laptop which has the problem on the headset-mic, the Pin 0x19 is not set the corret configuration for a mic and the pin presence can't be detected too after plugging a headset. Kailang suggested that we should set the coeff to enable the mic and apply the ALC269_FIXUP_LIFEBOOK_EXTMIC. After doing that, both headset-mic presence and headset-mic work well. The existing ALC255_FIXUP_ACER_MIC_NO_PRESENCE set the headset-mic jack to be a phantom jack. Now since the jack can support presence unsol event, let us imporve it to set the jack to be a normal jack. https://bugs.launchpad.net/bugs/1821269 Fixes: 5824ce8de7b1c ("ALSA: hda/realtek - Add support for Acer Aspire E5-475 headset mic") Cc: Chris Chiu <chiu(a)endlessm.com> CC: Daniel Drake <drake(a)endlessm.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Kailang Yang <kailang(a)realtek.com> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f1bac03e954b..f42a097f2e16 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6223,13 +6223,15 @@ static const struct hda_fixup alc269_fixups[] = { .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, [ALC255_FIXUP_ACER_MIC_NO_PRESENCE] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x19, 0x01a1913c }, /* use as headset mic, without its own jack detect */ - { } + .type = HDA_FIXUP_VERBS, + .v.verbs = (const struct hda_verb[]) { + /* Enable the Mic */ + { 0x20, AC_VERB_SET_COEF_INDEX, 0x45 }, + { 0x20, AC_VERB_SET_PROC_COEF, 0x5089 }, + {} }, .chained = true, - .chain_id = ALC255_FIXUP_HEADSET_MODE + .chain_id = ALC269_FIXUP_LIFEBOOK_EXTMIC }, [ALC255_FIXUP_ASUS_MIC_NO_PRESENCE] = { .type = HDA_FIXUP_PINS, @@ -7273,6 +7275,10 @@ static const struct snd_hda_pin_quirk alc269_pin_fixup_tbl[] = { {0x18, 0x02a11030}, {0x19, 0x0181303F}, {0x21, 0x0221102f}), + SND_HDA_PIN_QUIRK(0x10ec0255, 0x1025, "Acer", ALC255_FIXUP_ACER_MIC_NO_PRESENCE, + {0x12, 0x90a60140}, + {0x14, 0x90170120}, + {0x21, 0x02211030}), SND_HDA_PIN_QUIRK(0x10ec0255, 0x1025, "Acer", ALC255_FIXUP_ACER_MIC_NO_PRESENCE, {0x12, 0x90a601c0}, {0x14, 0x90171120}, -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: Baytrail: add quirk for Aegex 10 (RU2) tablet
by Tamás Kovács 28 May '19

28 May '19

This tablet has an incorrect acpi identifier, which is why it is trying to load the RT5640 driver instead of the RT5762 driver. The RT5640 driver, on the other hand, checks the hardware ID, so no drivers are loaded during boot. This fix resolves to load the RT5672 driver on this tablet during boot. It also provides the correct IO configuration, like the jack detect mode 3, for 1.8V pullup. I would like to thank Pierre-Louis Bossart for helping with this patch. Signed-off-by: Kovács Tamás <kepszlok(a)gmail.com> --- diff --git a/ori/sound/soc/intel/common/soc-acpi-intel-byt-match.c b/new/sound/soc/intel/common/soc-acpi-intel-byt-match.c index fe812a9..e549950 100644 --- a/ori/sound/soc/intel/common/soc-acpi-intel-byt-match.c +++ b/new/sound/soc/intel/common/soc-acpi-intel-byt-match.c @@ -1,4 +1,4 @@ -/* +/* * soc-apci-intel-byt-match.c - tables and support for BYT ACPI enumeration. * * Copyright (c) 2017, Intel Corporation. @@ -22,6 +22,7 @@ static unsigned long byt_machine_id; #define BYT_THINKPAD_10 1 #define BYT_POV_P1006W 2 +#define BYT_AEGEX_10 3 static int byt_thinkpad10_quirk_cb(const struct dmi_system_id *id) { @@ -35,6 +36,12 @@ static int byt_pov_p1006w_quirk_cb(const struct dmi_system_id *id) return 1; } +static int byt_aegex10_quirk_cb(const struct dmi_system_id *id) +{ + byt_machine_id = BYT_AEGEX_10; + return 1; +} + static const struct dmi_system_id byt_table[] = { { .callback = byt_thinkpad10_quirk_cb, @@ -75,6 +82,14 @@ static const struct dmi_system_id byt_table[] = { DMI_EXACT_MATCH(DMI_BOARD_NAME, "0E57"), }, }, + { + /* Aegex 10 tablet (RU2) */ + .callback = byt_aegex10_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AEGEX"), + DMI_MATCH(DMI_PRODUCT_VERSION, "RU2"), + }, + }, { } }; @@ -96,6 +111,15 @@ static struct snd_soc_acpi_mach byt_pov_p1006w = { .sof_tplg_filename = "sof-byt-rt5651.tplg", }; +static struct snd_soc_acpi_mach byt_aegex_10 = { + .id = "10EC5640", + .drv_name = "cht-bsw-rt5672", + .fw_filename = "intel/fw_sst_0f28.bin", + .board = "cht-bsw", + .sof_fw_filename = "intel/sof-byt.ri", + .sof_tplg_filename = "intel/sof-byt-rt5670.tplg", +}; + static struct snd_soc_acpi_mach *byt_quirk(void *arg) { struct snd_soc_acpi_mach *mach = arg; @@ -107,6 +131,8 @@ static struct snd_soc_acpi_mach *byt_quirk(void *arg) return &byt_thinkpad_10; case BYT_POV_P1006W: return &byt_pov_p1006w; + case BYT_AEGEX_10: + return &byt_aegex_10; default: return mach; } diff --git a/ori/sound/soc/codecs/rt5670.c b/new/sound/soc/codecs/rt5670.c index 9a03710..37cce04 100644 --- a/ori/sound/soc/codecs/rt5670.c +++ b/new/sound/soc/codecs/rt5670.c @@ -1,4 +1,4 @@ -/* +/* * rt5670.c -- RT5670 ALSA SoC audio codec driver * * Copyright 2014 Realtek Semiconductor Corp. @@ -2882,6 +2882,18 @@ static const struct dmi_system_id dmi_platform_intel_quirks[] = { RT5670_DEV_GPIO | RT5670_JD_MODE3), }, + { + .callback = rt5670_quirk_cb, + .ident = "Aegex 10 tablet (RU2)", + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "AEGEX"), + DMI_MATCH(DMI_PRODUCT_VERSION, "RU2"), + }, + .driver_data = (unsigned long *)(RT5670_DMIC_EN | + RT5670_DMIC2_INR | + RT5670_DEV_GPIO | + RT5670_JD_MODE3), + }, {} };

2 2

[alsa-devel] Cannot build broadwell on for-5.3
by Curtis Malainey 28 May '19

28 May '19

It appears the for-5.3 branch had some broken commits pulled in at 1c7c3237c0cc4ad3c7b0df458290c8e2a652f178 ("Merge tag 'v5.2-rc1' into asoc-5.3 ") building for the pixelbook I get the following build errors: Invalid absolute R_X86_64_32S relocation: _etext make[3]: *** [/mnt/host/source/src/third_party/kernel/v4.14/arch/x86/boot/compressed/Makefile:130: arch/x86/boot/compressed/vmlinux.relocs] Error 1 make[3]: *** Deleting file 'arch/x86/boot/compressed/vmlinux.relocs' Mainline is still broken as well. Not sure how to propagate this up to the responsible parties, figured I would start a discussion here first. I am also open to suggestions if there is a chance I am missing something.

4 5

[alsa-devel] [PATCH] ASoC: Add a debug log to track DAPM widget power up/down
by bgoswami＠codeaurora.org 28 May '19

28 May '19

From: Banajit Goswami <bgoswami(a)codeaurora.org> Add a debug log to help track widgets being powered-up and powered-down by DAPM. Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- sound/soc/soc-dapm.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 29cdfbf4c888..9bab90410de5 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1665,6 +1665,13 @@ static void dapm_seq_run(struct snd_soc_card *card, break; } + /* + * Add this debug log to keep track of widgets being + * powered-up and powered-down. + */ + dev_dbg(w->dapm->dev, "dapm: powering %s widget %s\n", + power_up ? "up" : "down", w->name); + if (ret < 0) dev_err(w->dapm->dev, "ASoC: Failed to apply widget power: %d\n", ret); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

3 3

[alsa-devel] [PATCH] ASoC: soc-core: fixup references at soc_cleanup_card_resources()
by Kuninori Morimoto 28 May '19

28 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit 53e947a0e1f7 ("ASoC: soc-core: merge card resources cleanup method") merged cleanup method of snd_soc_instantiate_card() and soc_cleanup_card_resources(). But, after this commit, if user uses unbind/bind to Component factor drivers, Kernel might indicates refcount error at soc_cleanup_card_resources(). The 1st reason is card->snd_card is still exist even though snd_card_free() was called, but it is already cleaned. We need to set NULL to it. 2nd is card->dapm and card create debugfs, but its dentry is still exist even though it was removed. We need to set NULL to it. Fixes: 53e947a0e1f7 ("ASoC: soc-core: merge card resources cleanup method") Cc: stable(a)vger.kernel.org # for v5.1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 7 ++++++- sound/soc/soc-dapm.c | 3 +++ 2 files changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 46e3ab0..758159a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -228,7 +228,10 @@ static void soc_init_card_debugfs(struct snd_soc_card *card) static void soc_cleanup_card_debugfs(struct snd_soc_card *card) { + if (!card->debugfs_card_root) + return; debugfs_remove_recursive(card->debugfs_card_root); + card->debugfs_card_root = NULL; } static void snd_soc_debugfs_init(void) @@ -2034,8 +2037,10 @@ static void soc_check_tplg_fes(struct snd_soc_card *card) static int soc_cleanup_card_resources(struct snd_soc_card *card) { /* free the ALSA card at first; this syncs with pending operations */ - if (card->snd_card) + if (card->snd_card) { snd_card_free(card->snd_card); + card->snd_card = NULL; + } /* remove and free each DAI */ soc_remove_dai_links(card); diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 0382a47..5d9d767 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2192,7 +2192,10 @@ static void dapm_debugfs_add_widget(struct snd_soc_dapm_widget *w) static void dapm_debugfs_cleanup(struct snd_soc_dapm_context *dapm) { + if (!dapm->debugfs_dapm) + return; debugfs_remove_recursive(dapm->debugfs_dapm); + dapm->debugfs_dapm = NULL; } #else -- 2.7.4

2 1

[alsa-devel] [PATCH][next] ASoC: cx2072x: remove redundant assignment to pulse_len
by Colin King 28 May '19

28 May '19

From: Colin Ian King <colin.king(a)canonical.com> Variable pulse_len is being initialized to 1 however this value is never read and pulse_len is being re-assigned later in a switch statement. Clean up the code by removing the redundant initialization. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/cx2072x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index 23d2b25fe04c..c11a585bbf70 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -679,7 +679,7 @@ static int cx2072x_config_i2spcm(struct cx2072x_priv *cx2072x) int is_right_j = 0; int is_frame_inv = 0; int is_bclk_inv = 0; - int pulse_len = 1; + int pulse_len; int frame_len = cx2072x->frame_size; int sample_size = cx2072x->sample_size; int i2s_right_slot; -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: SOF: topology: Use struct_size() helper
by Gustavo A. R. Silva 28 May '19

28 May '19

Make use of the struct_size() helper instead of an open-coded version in order to avoid any potential type mistakes, in particular in the context in which this code is being used. So, replace the following form: sizeof(struct sof_ipc_ctrl_data) + sizeof(struct sof_ipc_ctrl_value_chan) * le32_to_cpu(mc->num_channels) with: struct_size(scontrol->control_data, chanv, le32_to_cpu(mc->num_channels)) and so on... This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/sof/topology.c | 10 ++++------ 1 file changed, 4 insertions(+), 6 deletions(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index c88afa872a58..745cb875863c 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -442,9 +442,8 @@ static int sof_control_load_volume(struct snd_soc_component *scomp, return -EINVAL; /* init the volume get/put data */ - scontrol->size = sizeof(struct sof_ipc_ctrl_data) + - sizeof(struct sof_ipc_ctrl_value_chan) * - le32_to_cpu(mc->num_channels); + scontrol->size = struct_size(scontrol->control_data, chanv, + le32_to_cpu(mc->num_channels)); scontrol->control_data = kzalloc(scontrol->size, GFP_KERNEL); if (!scontrol->control_data) return -ENOMEM; @@ -501,9 +500,8 @@ static int sof_control_load_enum(struct snd_soc_component *scomp, return -EINVAL; /* init the enum get/put data */ - scontrol->size = sizeof(struct sof_ipc_ctrl_data) + - sizeof(struct sof_ipc_ctrl_value_chan) * - le32_to_cpu(ec->num_channels); + scontrol->size = struct_size(scontrol->control_data, chanv, + le32_to_cpu(ec->num_channels)); scontrol->control_data = kzalloc(scontrol->size, GFP_KERNEL); if (!scontrol->control_data) return -ENOMEM; -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: max98357a: Show KConfig entry
by Maxime Jourdan 28 May '19

28 May '19

The SEI510 board features a standalone MAX98357A codec. Add a tristate prompt to allow selecting the codec. Signed-off-by: Maxime Jourdan <mjourdan(a)baylibre.com> --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e730d47ac85b..48c94065072a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -708,7 +708,8 @@ config SND_SOC_MAX98095 tristate config SND_SOC_MAX98357A - tristate + tristate "Maxim MAX98357A CODEC" + depends on GPIOLIB config SND_SOC_MAX98371 tristate -- 2.21.0

2 1

[alsa-devel] [PATCH] ASoC: SOF: Use struct_size() in kmemdup()
by Gustavo A. R. Silva 28 May '19

28 May '19

Make use of the struct_size() helper instead of an open-coded version in order to avoid any potential type mistakes, in particular in the context in which this code is being used. So, replace code of the following form: sizeof(*w) + sizeof(struct sof_ipc_window_elem) * w->num_windows with: struct_size(w, window, w->num_windows) Notice that variable size is unnecessary, hence it is removed. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/sof/loader.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c index 628fae552442..16b016b76fd8 100644 --- a/sound/soc/sof/loader.c +++ b/sound/soc/sof/loader.c @@ -19,15 +19,13 @@ static int get_ext_windows(struct snd_sof_dev *sdev, { struct sof_ipc_window *w = container_of(ext_hdr, struct sof_ipc_window, ext_hdr); - size_t size; if (w->num_windows == 0 || w->num_windows > SOF_IPC_MAX_ELEMS) return -EINVAL; - size = sizeof(*w) + sizeof(struct sof_ipc_window_elem) * w->num_windows; - /* keep a local copy of the data */ - sdev->info_window = kmemdup(w, size, GFP_KERNEL); + sdev->info_window = kmemdup(w, struct_size(w, window, w->num_windows), + GFP_KERNEL); if (!sdev->info_window) return -ENOMEM; -- 2.21.0

2 1

[alsa-devel] [PATCH 1/2] dt-bindings: sound: Convert Allwinner I2S binding to YAML
by Maxime Ripard 28 May '19

28 May '19

The Allwinner SoCs feature an I2S controller across multiple SoC generations. However, earlier generations were a bit simpler than the subsequent ones, and for example would always have RX and TX capabilities, and no reset lines. Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml | 100 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++- Documentation/devicetree/bindings/sound/sun4i-i2s.txt | 45 +-------------------------------- 2 files changed, 100 insertions(+), 45 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sun4i-i2s.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml new file mode 100644 index 000000000000..85b2d6d84055 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-i2s.yaml @@ -0,0 +1,100 @@ +# SPDX-License-Identifier: (GPL-2.0+ OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-i2s.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 I2S Controller Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: allwinner,sun4i-a10-i2s + - const: allwinner,sun6i-a31-i2s + - const: allwinner,sun8i-a83t-i2s + - const: allwinner,sun8i-h3-i2s + - const: allwinner,sun50i-a64-codec-i2s + - items: + - const: allwinner,sun50i-a64-i2s + - const: allwinner,sun8i-h3-i2s + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: mod + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + # Even though it only applies to subschemas under the conditionals, + # not listing them here will trigger a warning because of the + # additionalsProperties set to false. + resets: + maxItems: 1 + +allOf: + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun6i-a31-i2s + - allwinner,sun8i-a83t-i2s + - allwinner,sun8i-h3-i2s + - allwinner,sun50i-a64-codec-i2s + + then: + required: + - resets + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + i2s0: i2s@1c22400 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-i2s"; + reg = <0x01c22400 0x400>; + interrupts = <0 16 4>; + clocks = <&apb0_gates 3>, <&i2s0_clk>; + clock-names = "apb", "mod"; + dmas = <&dma 0 3>, <&dma 0 3>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt b/Documentation/devicetree/bindings/sound/sun4i-i2s.txt deleted file mode 100644 index 61e71c1729e0..000000000000 --- a/Documentation/devicetree/bindings/sound/sun4i-i2s.txt +++ /dev/null @@ -1,45 +0,0 @@ -* Allwinner A10 I2S controller - -The I2S bus (Inter-IC sound bus) is a serial link for digital -audio data transfer between devices in the system. - -Required properties: - -- compatible: should be one of the following: - - "allwinner,sun4i-a10-i2s" - - "allwinner,sun6i-a31-i2s" - - "allwinner,sun8i-a83t-i2s" - - "allwinner,sun8i-h3-i2s" - - "allwinner,sun50i-a64-codec-i2s" -- reg: physical base address of the controller and length of memory mapped - region. -- interrupts: should contain the I2S interrupt. -- dmas: DMA specifiers for tx and rx dma. See the DMA client binding, - Documentation/devicetree/bindings/dma/dma.txt -- dma-names: should include "tx" and "rx". -- clocks: a list of phandle + clock-specifer pairs, one for each entry in clock-names. -- clock-names: should contain the following: - - "apb" : clock for the I2S bus interface - - "mod" : module clock for the I2S controller -- #sound-dai-cells : Must be equal to 0 - -Required properties for the following compatibles: - - "allwinner,sun6i-a31-i2s" - - "allwinner,sun8i-a83t-i2s" - - "allwinner,sun8i-h3-i2s" - - "allwinner,sun50i-a64-codec-i2s" -- resets: phandle to the reset line for this codec - -Example: - -i2s0: i2s@1c22400 { - #sound-dai-cells = <0>; - compatible = "allwinner,sun4i-a10-i2s"; - reg = <0x01c22400 0x400>; - interrupts = <GIC_SPI 16 IRQ_TYPE_LEVEL_HIGH>; - clocks = <&apb0_gates 3>, <&i2s0_clk>; - clock-names = "apb", "mod"; - dmas = <&dma SUN4I_DMA_NORMAL 3>, - <&dma SUN4I_DMA_NORMAL 3>; - dma-names = "rx", "tx"; -}; base-commit: 86c9acd4fdf5b2e1648a6cf3d6e99a7f0d35cb4f -- git-series 0.9.1

3 5

[alsa-devel] [PATCH][next] ASoC: cx2072x: fix integer overflow on unsigned int multiply
by Colin King 28 May '19

28 May '19

From: Colin Ian King <colin.king(a)canonical.com> In the case where frac_div larger than 96 the result of an unsigned multiplication overflows an unsigned int. For example, this can happen when the sample_rate is 192000 and pll_input is 122. Fix this by casing the first term of the mutiply to a u64. Also remove the extraneous parentheses around the expression. Addresses-Coverity: ("Unintentional integer overflow") Fixes: a497a4363706 ("ASoC: Add support for Conexant CX2072X CODEC") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/cx2072x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index c11a585bbf70..e8e6fd2e97b6 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -627,7 +627,7 @@ static int cx2072x_config_pll(struct cx2072x_priv *cx2072x) if (frac_div) { frac_div *= 1000; frac_div /= pll_input; - frac_num = ((4000 + frac_div) * ((1 << 20) - 4)); + frac_num = (u64)(4000 + frac_div) * ((1 << 20) - 4); do_div(frac_num, 7); frac = ((u32)frac_num + 499) / 1000; } -- 2.20.1

3 2

[alsa-devel] [PATCH][next] ASoC: cx2072x: fix spelling mistake "configued" -> "configured"
by Colin King 28 May '19

28 May '19

From: Colin Ian King <colin.king(a)canonical.com> There is a spelling mistake in a dev_err error message. Fit it. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/cx2072x.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cx2072x.c b/sound/soc/codecs/cx2072x.c index e8e6fd2e97b6..f25376789c5c 100644 --- a/sound/soc/codecs/cx2072x.c +++ b/sound/soc/codecs/cx2072x.c @@ -933,7 +933,7 @@ static int cx2072x_hw_params(struct snd_pcm_substream *substream, return frame_size; if (cx2072x->mclk_rate == 0) { - dev_err(dev, "Master clock rate is not configued\n"); + dev_err(dev, "Master clock rate is not configured\n"); return -EINVAL; } -- 2.20.1

3 2

[alsa-devel] [PATCH] ASoC: fsl: sai: Fix clock source for mclk0
by Daniel Baluta 28 May '19

28 May '19

SAI provide multiple master clock source options selectable via bit MSEL of TCR2/RCR2. All possible master clock sources are stored in sai->mclk_clk array. Current implementation assumes that MCLK0 source is always busclk, but this is wrong! For example, on i.MX8QM we have: 00b - Bus Clock selected. 01b - Master Clock (MCLK) 1 option selected. 10b - Master Clock (MCLK) 2 option selected. 11b - Master Clock (MCLK) 3 option selected. while on i.MX6SX we have: 00b - Master Clock (MCLK) 1 option selected. 01b - Master Clock (MCLK) 1 option selected. 10b - Master Clock (MCLK) 2 option selected. 11b - Master Clock (MCLK) 3 option selected. So, this patch will read mclk0 source clock from device tree. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Daniel Baluta <daniel.baluta(a)nxp.com> --- sound/soc/fsl/fsl_sai.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index d2a4dc744fd7..faa8de87ff83 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -829,8 +829,7 @@ static int fsl_sai_probe(struct platform_device *pdev) sai->bus_clk = NULL; } - sai->mclk_clk[0] = sai->bus_clk; - for (i = 1; i < FSL_SAI_MCLK_MAX; i++) { + for (i = 0; i < FSL_SAI_MCLK_MAX; i++) { sprintf(tmp, "mclk%d", i); sai->mclk_clk[i] = devm_clk_get(&pdev->dev, tmp); if (IS_ERR(sai->mclk_clk[i])) { -- 2.17.1

3 6

[alsa-devel] [PATCH 1/3][V2] lib: fix match_string() helper on -1 array size
by Alexandru Ardelean 28 May '19

28 May '19

The documentation the `_match_string()` helper mentions that `n` should be: * @n: number of strings in the array or -1 for NULL terminated arrays The behavior of the function is different, in the sense that it exits on the first NULL element in the array, regardless of whether `n` is -1 or a positive number. This patch changes the behavior, to exit the loop when a NULL element is found and n == -1. Essentially, this aligns the behavior with the doc-string. There are currently many users of `match_string()`, and so, in order to go through them, the next patches in the series will focus on doing some cosmetic changes, which are aimed at grouping the users of `match_string()`. Signed-off-by: Alexandru Ardelean <alexandru.ardelean(a)analog.com> --- Changelog v1 -> v2: * split the initial series into just 3 patches that fix the `match_string()` helper and start introducing a new version of this helper, which computes array-size of static arrays lib/string.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/lib/string.c b/lib/string.c index 6016eb3ac73d..e2cf5acc83bd 100644 --- a/lib/string.c +++ b/lib/string.c @@ -681,8 +681,11 @@ int match_string(const char * const *array, size_t n, const char *string) for (index = 0; index < n; index++) { item = array[index]; - if (!item) + if (!item) { + if (n != (size_t)-1) + continue; break; + } if (!strcmp(item, string)) return index; } -- 2.20.1

1 2

[alsa-devel] [PATCH v2 0/5] ALSA: line6 fixes and cleanups
by Takashi Iwai 28 May '19

28 May '19

Hi, this is a patchset I forgot to merge for fixing and impriving LINE6 drivers. The v2 is rebased after the latest toneport fix, and contains a cleanup with container_of(), too. Takashi === Takashi Iwai (5): ALSA: line6: pod: Rewrite complex timer & work combo with a delayed work ALSA: line6: podhd: Rewrite complex timer & work combo with a delayed work ALSA: line6: variax: Rewrite complex timer & work combo with a delayed work ALSA: line6: Drop superfluous timer helper function ALSA: line6: Use container_of() sound/usb/line6/driver.c | 11 ---- sound/usb/line6/driver.h | 9 --- sound/usb/line6/pod.c | 108 +++++++++++------------------------ sound/usb/line6/podhd.c | 80 ++++---------------------- sound/usb/line6/toneport.c | 14 ++--- sound/usb/line6/variax.c | 138 ++++++++++++++------------------------------- 6 files changed, 94 insertions(+), 266 deletions(-) -- 2.16.4

1 5

[alsa-devel] [PATCH] ALSA: line6: Assure canceling delayed work at disconnection
by Takashi Iwai 28 May '19

28 May '19

The current code performs the cancel of a delayed work at the late stage of disconnection procedure, which may lead to the access to the already cleared state. This patch assures to call cancel_delayed_work_sync() at the beginning of the disconnection procedure for avoiding that race. The delayed work object is now assigned in the common line6 object instead of its derivative, so that we can call cancel_delayed_work_sync(). Along with the change, the startup function is called via the new callback instead. This will make it easier to port other LINE6 drivers to use the delayed work for startup in later patches. Reported-by: syzbot+5255458d5e0a2b10bbb9(a)syzkaller.appspotmail.com Fixes: 7f84ff68be05 ("ALSA: line6: toneport: Fix broken usage of timer for delayed execution") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/line6/driver.c | 12 ++++++++++++ sound/usb/line6/driver.h | 4 ++++ sound/usb/line6/toneport.c | 15 +++------------ 3 files changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index b61f65bed4e4..2b57854335b3 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -720,6 +720,15 @@ static int line6_init_cap_control(struct usb_line6 *line6) return 0; } +static void line6_startup_work(struct work_struct *work) +{ + struct usb_line6 *line6 = + container_of(work, struct usb_line6, startup_work.work); + + if (line6->startup) + line6->startup(line6); +} + /* Probe USB device. */ @@ -755,6 +764,7 @@ int line6_probe(struct usb_interface *interface, line6->properties = properties; line6->usbdev = usbdev; line6->ifcdev = &interface->dev; + INIT_DELAYED_WORK(&line6->startup_work, line6_startup_work); strcpy(card->id, properties->id); strcpy(card->driver, driver_name); @@ -825,6 +835,8 @@ void line6_disconnect(struct usb_interface *interface) if (WARN_ON(usbdev != line6->usbdev)) return; + cancel_delayed_work(&line6->startup_work); + if (line6->urb_listen != NULL) line6_stop_listen(line6); diff --git a/sound/usb/line6/driver.h b/sound/usb/line6/driver.h index 61425597eb61..650d909c9c4f 100644 --- a/sound/usb/line6/driver.h +++ b/sound/usb/line6/driver.h @@ -178,11 +178,15 @@ struct usb_line6 { fifo; } messages; + /* Work for delayed PCM startup */ + struct delayed_work startup_work; + /* If MIDI is supported, buffer_message contains the pre-processed data; * otherwise the data is only in urb_listen (buffer_incoming). */ void (*process_message)(struct usb_line6 *); void (*disconnect)(struct usb_line6 *line6); + void (*startup)(struct usb_line6 *line6); }; extern char *line6_alloc_sysex_buffer(struct usb_line6 *line6, int code1, diff --git a/sound/usb/line6/toneport.c b/sound/usb/line6/toneport.c index e28368d8eba2..55865f7e437d 100644 --- a/sound/usb/line6/toneport.c +++ b/sound/usb/line6/toneport.c @@ -54,9 +54,6 @@ struct usb_line6_toneport { /* Firmware version (x 100) */ u8 firmware_version; - /* Work for delayed PCM startup */ - struct delayed_work pcm_work; - /* Device type */ enum line6_device_type type; @@ -241,12 +238,8 @@ static int snd_toneport_source_put(struct snd_kcontrol *kcontrol, return 1; } -static void toneport_start_pcm(struct work_struct *work) +static void toneport_startup(struct usb_line6 *line6) { - struct usb_line6_toneport *toneport = - container_of(work, struct usb_line6_toneport, pcm_work.work); - struct usb_line6 *line6 = &toneport->line6; - line6_pcm_acquire(line6->line6pcm, LINE6_STREAM_MONITOR, true); } @@ -394,7 +387,7 @@ static int toneport_setup(struct usb_line6_toneport *toneport) if (toneport_has_led(toneport)) toneport_update_led(toneport); - schedule_delayed_work(&toneport->pcm_work, + schedule_delayed_work(&toneport->line6.startup_work, msecs_to_jiffies(TONEPORT_PCM_DELAY * 1000)); return 0; } @@ -407,8 +400,6 @@ static void line6_toneport_disconnect(struct usb_line6 *line6) struct usb_line6_toneport *toneport = (struct usb_line6_toneport *)line6; - cancel_delayed_work_sync(&toneport->pcm_work); - if (toneport_has_led(toneport)) toneport_remove_leds(toneport); } @@ -424,9 +415,9 @@ static int toneport_init(struct usb_line6 *line6, struct usb_line6_toneport *toneport = (struct usb_line6_toneport *) line6; toneport->type = id->driver_info; - INIT_DELAYED_WORK(&toneport->pcm_work, toneport_start_pcm); line6->disconnect = line6_toneport_disconnect; + line6->startup = toneport_startup; /* initialize PCM subsystem: */ err = line6_init_pcm(line6, &toneport_pcm_properties); -- 2.16.4

1 0

[alsa-devel] [PATCH 1/5] ALSA: hda - Force polling mode on CNL for fixing codec communication
by Bard liao 28 May '19

28 May '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> We observed the same issue as reported by commit a8d7bde23e7130686b7662 ("ALSA: hda - Force polling mode on CFL for fixing codec communication") We don't have a better solution. So apply the same workaround to CNL. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 2ec91085fa3e..a93468ffb85c 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -375,6 +375,7 @@ enum { #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) #define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) +#define IS_CNL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x9dc8) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -1700,8 +1701,8 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, else chip->bdl_pos_adj = bdl_pos_adj[dev]; - /* Workaround for a communication error on CFL (bko#199007) */ - if (IS_CFL(pci)) + /* Workaround for a communication error on CFL (bko#199007) and CNL */ + if (IS_CFL(pci) || IS_CNL(pci)) chip->polling_mode = 1; err = azx_bus_init(chip, model[dev], &pci_hda_io_ops); -- 2.17.1

3 7

[alsa-devel] KASAN: null-ptr-deref Write in submit_audio_out_urb
by syzbot 27 May '19

27 May '19

Hello, syzbot found the following crash on: HEAD commit: 69bbe8c7 usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=118b0aa2a00000 kernel config: https://syzkaller.appspot.com/x/.config?x=c309d28e15db39c5 dashboard link: https://syzkaller.appspot.com/bug?extid=5255458d5e0a2b10bbb9 compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=11f4064ca00000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=14d51982a00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+5255458d5e0a2b10bbb9(a)syzkaller.appspotmail.com snd_usb_toneport 1-1:0.0: read request failed (error -110) snd_usb_toneport 1-1:0.0: read request failed (error -110) snd_usb_toneport 1-1:0.0: write request failed (error -110) usb 1-1: send failed (error -110) usb 1-1: send failed (error -110) snd_usb_toneport 1-1:0.0: Line 6 POD Studio UX2 now attached ================================================================== BUG: KASAN: null-ptr-deref in memset include/linux/string.h:344 [inline] BUG: KASAN: null-ptr-deref in submit_audio_out_urb+0x919/0x1780 sound/usb/line6/playback.c:246 Write of size 20 at addr 0000000000000010 by task kworker/1:2/2466 CPU: 1 PID: 2466 Comm: kworker/1:2 Not tainted 5.2.0-rc1+ #9 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events toneport_start_pcm Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 __kasan_report.cold+0x5/0x32 mm/kasan/report.c:321 kasan_report+0xe/0x20 mm/kasan/common.c:614 memset+0x20/0x40 mm/kasan/common.c:107 memset include/linux/string.h:344 [inline] submit_audio_out_urb+0x919/0x1780 sound/usb/line6/playback.c:246 line6_submit_audio_out_all_urbs+0xc9/0x120 sound/usb/line6/playback.c:295 line6_stream_start+0x156/0x1f0 sound/usb/line6/pcm.c:199 line6_pcm_acquire+0x134/0x210 sound/usb/line6/pcm.c:322 process_one_work+0x90a/0x1580 kernel/workqueue.c:2268 worker_thread+0x96/0xe20 kernel/workqueue.c:2414 kthread+0x30e/0x420 kernel/kthread.c:254 ret_from_fork+0x3a/0x50 arch/x86/entry/entry_64.S:352 ================================================================== Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 2466 Comm: kworker/1:2 Tainted: G B 5.2.0-rc1+ #9 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: events toneport_start_pcm Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 panic+0x292/0x6df kernel/panic.c:218 end_report+0x43/0x49 mm/kasan/report.c:95 __kasan_report.cold+0xd/0x32 mm/kasan/report.c:324 kasan_report+0xe/0x20 mm/kasan/common.c:614 memset+0x20/0x40 mm/kasan/common.c:107 memset include/linux/string.h:344 [inline] submit_audio_out_urb+0x919/0x1780 sound/usb/line6/playback.c:246 line6_submit_audio_out_all_urbs+0xc9/0x120 sound/usb/line6/playback.c:295 line6_stream_start+0x156/0x1f0 sound/usb/line6/pcm.c:199 line6_pcm_acquire+0x134/0x210 sound/usb/line6/pcm.c:322 process_one_work+0x90a/0x1580 kernel/workqueue.c:2268 worker_thread+0x96/0xe20 kernel/workqueue.c:2414 kthread+0x30e/0x420 kernel/kthread.c:254 ret_from_fork+0x3a/0x50 arch/x86/entry/entry_64.S:352 Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

1 0

[alsa-devel] [PATCH v3 0/7] Allwinner H6 SPDIF support
by Clément Péron 27 May '19

27 May '19

*H6 DMA support IS REQUIRED* Allwinner H6 SoC has a SPDIF controller called One Wire Audio (OWA) which is different from the previous H3 generation and not compatible. Difference are an increase of fifo sizes, some memory mapping are different and there is now the possibility to output the master clock on a pin. Actually all these features are unused and only a bit for flushing the TX fifo is required. Also this series requires the DMA working on H6, a first version has been submitted by Jernej Škrabec[1] but has not been accepted yet. [1] https://patchwork.kernel.org/project/linux-arm-kernel/list/?series=89011 Changes since v2: - Split quirks and H6 support patch - Add specific section for quirks comment Changes since v1: - Remove H3 compatible - Add TX fifo bit flush quirks - Add H6 bindings in SPDIF driver Clément Péron (7): dt-bindings: sound: sun4i-spdif: Add Allwinner H6 compatible ASoC: sun4i-spdif: Move quirks to the top ASoC: sun4i-spdif: Add TX fifo bit flush quirks ASoC: sun4i-spdif: Add support for H6 SoC arm64: dts: allwinner: Add SPDIF node for Allwinner H6 arm64: dts: allwinner: h6: Enable SPDIF for Beelink GS1 arm64: defconfig: Enable Sun4i SPDIF module .../bindings/sound/sunxi,sun4i-spdif.txt | 3 +- .../dts/allwinner/sun50i-h6-beelink-gs1.dts | 4 ++ arch/arm64/boot/dts/allwinner/sun50i-h6.dtsi | 38 ++++++++++++++ arch/arm64/configs/defconfig | 1 + sound/soc/sunxi/sun4i-spdif.c | 49 ++++++++++++++++--- 5 files changed, 88 insertions(+), 7 deletions(-) -- 2.20.1

2 14

[alsa-devel] WARNING in line6_pcm_acquire
by syzbot 27 May '19

27 May '19

Hello, syzbot found the following crash on: HEAD commit: 43151d6c usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=169fe1f8a00000 kernel config: https://syzkaller.appspot.com/x/.config?x=95aff7278e7ff25e dashboard link: https://syzkaller.appspot.com/bug?extid=192a537b5c634febc6cf compiler: gcc (GCC) 9.0.0 20181231 (experimental) Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+192a537b5c634febc6cf(a)syzkaller.appspotmail.com ------------[ cut here ]------------ do not call blocking ops when !TASK_RUNNING; state=1 set at [<000000009424b595>] do_nanosleep+0x107/0x6a0 kernel/time/hrtimer.c:1675 WARNING: CPU: 0 PID: 4661 at kernel/sched/core.c:6136 __might_sleep+0x135/0x190 kernel/sched/core.c:6136 Kernel panic - not syncing: panic_on_warn set ... CPU: 0 PID: 4661 Comm: syz-executor.4 Not tainted 5.1.0-rc3+ #8 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: <IRQ> __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 panic+0x292/0x5e1 kernel/panic.c:214 __warn.cold+0x20/0x53 kernel/panic.c:571 report_bug+0x262/0x2a0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:179 [inline] fixup_bug arch/x86/kernel/traps.c:174 [inline] do_error_trap+0x12b/0x1e0 arch/x86/kernel/traps.c:272 do_invalid_op+0x32/0x40 arch/x86/kernel/traps.c:291 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:973 RIP: 0010:__might_sleep+0x135/0x190 kernel/sched/core.c:6136 Code: 65 48 8b 1c 25 c0 de 01 00 48 8d 7b 10 48 89 fe 48 c1 ee 03 80 3c 06 00 75 2b 48 8b 73 10 48 c7 c7 c0 ec c5 85 e8 c6 4b f6 ff <0f> 0b e9 46 ff ff ff e8 2f d0 45 00 e9 29 ff ff ff e8 25 d0 45 00 RSP: 0018:ffff8881db207b48 EFLAGS: 00010286 RAX: 0000000000000000 RBX: ffff8881c7d71800 RCX: 0000000000000000 RDX: 0000000000000100 RSI: ffffffff8127bbcd RDI: ffffed103b640f5b RBP: ffffffff85c622c0 R08: ffff8881c7d71800 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 000000000000038c R13: 0000000000000000 R14: 0000000000000000 R15: ffffffff849f6d40 __mutex_lock_common kernel/locking/mutex.c:908 [inline] __mutex_lock+0xc8/0x12b0 kernel/locking/mutex.c:1072 line6_pcm_acquire+0x30/0x210 sound/usb/line6/pcm.c:311 call_timer_fn+0x15c/0x5e0 kernel/time/timer.c:1325 expire_timers kernel/time/timer.c:1362 [inline] __run_timers kernel/time/timer.c:1681 [inline] __run_timers kernel/time/timer.c:1649 [inline] run_timer_softirq+0x586/0x1400 kernel/time/timer.c:1694 __do_softirq+0x21f/0x8bc kernel/softirq.c:293 invoke_softirq kernel/softirq.c:374 [inline] irq_exit+0x17c/0x1a0 kernel/softirq.c:414 exiting_irq arch/x86/include/asm/apic.h:536 [inline] smp_apic_timer_interrupt+0xf1/0x490 arch/x86/kernel/apic/apic.c:1062 apic_timer_interrupt+0xf/0x20 arch/x86/entry/entry_64.S:807 </IRQ> RIP: 0010:arch_local_irq_restore arch/x86/include/asm/irqflags.h:81 [inline] RIP: 0010:__raw_spin_unlock_irqrestore include/linux/spinlock_api_smp.h:160 [inline] RIP: 0010:_raw_spin_unlock_irqrestore+0x40/0x50 kernel/locking/spinlock.c:184 Code: e8 25 19 bc fb 48 89 ef e8 dd f7 bc fb f6 c7 02 75 11 53 9d e8 a1 fd d8 fb 65 ff 0d 0a 59 99 7a 5b 5d c3 e8 d2 fb d8 fb 53 9d <eb> ed 0f 1f 40 00 66 2e 0f 1f 84 00 00 00 00 00 55 48 89 fd 65 ff RSP: 0018:ffff8881b13ffba8 EFLAGS: 00000246 ORIG_RAX: ffffffffffffff13 RAX: 0000000000000007 RBX: 0000000000000246 RCX: 0000000000000000 RDX: 0000000000000000 RSI: 0000000000000006 RDI: ffff8881c7d72034 RBP: ffff8881db224a80 R08: ffff8881c7d71800 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: dffffc0000000000 R13: ffff8881db224b00 R14: ffff8881db224b00 R15: ffff8881db224a80 unlock_hrtimer_base kernel/time/hrtimer.c:878 [inline] hrtimer_start_range_ns+0x5b0/0xad0 kernel/time/hrtimer.c:1109 hrtimer_start_expires include/linux/hrtimer.h:409 [inline] do_nanosleep+0x19b/0x6a0 kernel/time/hrtimer.c:1676 hrtimer_nanosleep+0x258/0x510 kernel/time/hrtimer.c:1733 __do_sys_nanosleep kernel/time/hrtimer.c:1767 [inline] __se_sys_nanosleep kernel/time/hrtimer.c:1754 [inline] __x64_sys_nanosleep+0x19d/0x220 kernel/time/hrtimer.c:1754 do_syscall_64+0xbd/0x500 arch/x86/entry/common.c:290 entry_SYSCALL_64_after_hwframe+0x49/0xbe RIP: 0033:0x486570 Code: 00 00 48 c7 c0 d4 ff ff ff 64 c7 00 16 00 00 00 31 c0 eb be 66 0f 1f 44 00 00 83 3d e1 01 5d 00 00 75 14 b8 23 00 00 00 0f 05 <48> 3d 01 f0 ff ff 0f 83 b4 e0 f8 ff c3 48 83 ec 08 e8 ea 53 fd ff RSP: 002b:00007ffc588bb088 EFLAGS: 00000246 ORIG_RAX: 0000000000000023 RAX: ffffffffffffffda RBX: 00000000000a95fa RCX: 0000000000486570 RDX: 0000000000000000 RSI: 0000000000000000 RDI: 00007ffc588bb090 RBP: 00000000000002d5 R08: 0000000000000001 R09: 00000000027fc940 R10: 0000000000000000 R11: 0000000000000246 R12: 0000000000000008 R13: 00007ffc588bb0e0 R14: 00000000000a8e2a R15: 00007ffc588bb0f0 Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

1 0

Re: [alsa-devel] [PATCH 2/5] ASoC: madera: Add common support for Cirrus Logic Madera codecs
by Charles Keepax 27 May '19

27 May '19

On Sun, May 26, 2019 at 01:18:46PM +0100, Mark Brown wrote: > On Fri, May 24, 2019 at 04:24:10PM +0100, Richard Fitzgerald wrote: > > On 24/05/19 16:21, Richard Fitzgerald wrote: > > > On 24/05/19 15:56, Mark Brown wrote: > > > > This will delay both before and after every power up and power down. > > > > Are you sure that makes sense? > > > > I think that's correct but we can re-check with hardware people. > > > It's not just a delay, it needs to ensure there are always a > > > certain number of SYSCLK cycles in the hardware to avoid leaving > > > certain state machines in limbo. > > That sounds like you might want both _POST_PMU and _POST_PMD but do you > really need the _PREs as well? Yeah the requirement from the hardware guys was that we needed a guard band between changing the SYSCLK state and other things happening. So it has to happen both before and after the state of SYSCLK changes. It is certainly far from ideal but it is what it is. Thanks, Charles

1 0

[alsa-devel] [PATCH 3/3] ALSA: jack: add switch event for unsupported jack types
by bgoswami＠codeaurora.org 27 May '19

27 May '19

From: Banajit Goswami <bgoswami(a)codeaurora.org> Add a jack switch event to report unsupported plug type. This event can be used to report a headset or an extension cable with GND/MIC swap etc., which may not be supported by the device. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- sound/core/jack.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/core/jack.c b/sound/core/jack.c index 36b047b..4c21e48 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -40,6 +40,7 @@ struct snd_jack_kctl { SW_JACK_PHYSICAL_INSERT, SW_VIDEOOUT_INSERT, SW_LINEIN_INSERT, + SW_UNSUPPORT_INSERT, }; #endif /* CONFIG_SND_JACK_INPUT_DEV */ -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH 1/3] ALSA: jack: Remove hard coding of jack_switch_types array size
by bgoswami＠codeaurora.org 27 May '19

27 May '19

From: Banajit Goswami <bgoswami(a)codeaurora.org> The size for jack_switch_types array is currently controlled by a MACRO 'SND_JACK_SWITCH_TYPES', whose value needs to be updated everytime a new jack switch type is added. Remove this MACRO and use ARRAY_SIZE instead to get size of the array. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- include/sound/jack.h | 3 --- sound/core/jack.c | 4 ++-- 2 files changed, 2 insertions(+), 5 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 1e84bfb..b0791c5 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -68,9 +68,6 @@ enum snd_jack_types { SND_JACK_BTN_5 = 0x0200, }; -/* Keep in sync with definitions above */ -#define SND_JACK_SWITCH_TYPES 6 - struct snd_jack { struct list_head kctl_list; struct snd_card *card; diff --git a/sound/core/jack.c b/sound/core/jack.c index 84c2a17..36b047b 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -33,7 +33,7 @@ struct snd_jack_kctl { }; #ifdef CONFIG_SND_JACK_INPUT_DEV -static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { +static int jack_switch_types[] = { SW_HEADPHONE_INSERT, SW_MICROPHONE_INSERT, SW_LINEOUT_INSERT, @@ -250,7 +250,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) + for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) if (type & (1 << i)) input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH RFC] ASoC: hdac_hdmi: refine hdmi jacks
by libin.yang＠intel.com 27 May '19

27 May '19

From: Libin Yang <libin.yang(a)intel.com> Sorry for a big patch. I will split it if the patch is acceptable. This patch does the following actions: 1) jack is bound to pcm in origal code. The patch binds jack to pin-port. This mean each pin-port (3x3) has a jack. pin-port is used for DP-MST mode. The port means device entry for DP-MST in hdac_hdmi As hdac_hdmi allows user to map the pin-port to a pcm manually, the jack is bound to pcm is not accurate. However, PA assumes jack is bound to PCM for the legacy mode. For example, for ubuntu, in /usr/share/pulseaudio/alsa-mixer/paths/hdmi-output-x.conf uses "Jack HDMI/DP,pcm=n" to find which jack event is used. This main issue is: pcm number is fixed (3, 7, 8, 9, 10) for hdmi in legacy mode. But it is not always the same for ASoC hdmi audio. 2) move snd_soc_card_jack_new() from machine driver to codec driver. This is because the jack information is more lated to the codec Also this patch calls snd_jack_add_new_kctl() to export the jack kctl to userspace. 3) Don't call hdac_hdmi_jack_report() in hdac_hdmi_create_pin_port_muxs() Jack event means the jack pin hotplug event. It should not related to user setting. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 141 +++++++++++---------- sound/soc/intel/boards/bxt_da7219_max98357a.c | 10 -- sound/soc/intel/boards/bxt_rt298.c | 10 -- sound/soc/intel/boards/glk_rt5682_max98357a.c | 10 -- sound/soc/intel/boards/kbl_da7219_max98357a.c | 10 -- sound/soc/intel/boards/kbl_da7219_max98927.c | 10 -- sound/soc/intel/boards/kbl_rt5660.c | 10 -- sound/soc/intel/boards/kbl_rt5663_max98927.c | 10 -- .../soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 9 -- sound/soc/intel/boards/skl_hda_dsp_common.c | 9 -- sound/soc/intel/boards/skl_nau88l25_max98357a.c | 11 -- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 11 -- sound/soc/intel/boards/skl_rt286.c | 10 -- sound/soc/intel/boards/sof_rt5682.c | 11 -- 14 files changed, 72 insertions(+), 200 deletions(-) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 660e058..49f02b1 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -4,6 +4,7 @@ * Copyright (C) 2014-2015 Intel Corp * Author: Samreen Nilofer <samreen.nilofer(a)intel.com> * Subhransu S. Prusty <subhransu.s.prusty(a)intel.com> + * Libin Yang <libin.yang(a)intel.com> * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ * * This program is free software; you can redistribute it and/or modify @@ -30,6 +31,7 @@ #include <sound/hda_i915.h> #include <sound/pcm_drm_eld.h> #include <sound/hda_chmap.h> +#include <sound/hda_codec.h> #include "../../hda/local.h" #include "hdac_hdmi.h" @@ -95,6 +97,8 @@ struct hdac_hdmi_port { int num_mux_nids; hda_nid_t mux_nids[HDA_MAX_CONNECTIONS]; struct hdac_hdmi_eld eld; + struct snd_soc_jack jack; + int jack_event; const char *jack_pin; struct snd_soc_dapm_context *dapm; const char *output_pin; @@ -105,14 +109,12 @@ struct hdac_hdmi_pcm { int pcm_id; struct list_head port_list; struct hdac_hdmi_cvt *cvt; - struct snd_soc_jack *jack; int stream_tag; int channels; int format; bool chmap_set; unsigned char chmap[8]; /* ALSA API channel-map */ struct mutex lock; - int jack_event; }; struct hdac_hdmi_dai_port_map { @@ -166,8 +168,7 @@ hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, return pcm; } -static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, - struct hdac_hdmi_port *port, bool is_connect) +static void hdac_hdmi_jack_report(struct hdac_hdmi_port *port, bool is_connect) { struct hdac_device *hdev = port->pin->hdev; @@ -176,33 +177,24 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, else snd_soc_dapm_disable_pin(port->dapm, port->jack_pin); + snd_soc_dapm_sync(port->dapm); + + if (!port->jack.jack) + return; if (is_connect) { - /* - * Report Jack connect event when a device is connected - * for the first time where same PCM is attached to multiple - * ports. - */ - if (pcm->jack_event == 0) { + if (port->jack_event == 0) { dev_dbg(&hdev->dev, - "jack report for pcm=%d\n", - pcm->pcm_id); - snd_soc_jack_report(pcm->jack, SND_JACK_AVOUT, - SND_JACK_AVOUT); + "jack report for pin:port = %d:%d\n", + port->id, port->pin->nid); + snd_soc_jack_report(&port->jack, SND_JACK_AVOUT, + SND_JACK_AVOUT); } - pcm->jack_event++; + port->jack_event = 1; } else { - /* - * Report Jack disconnect event when a device is disconnected - * is the only last connected device when same PCM is attached - * to multiple ports. - */ - if (pcm->jack_event == 1) - snd_soc_jack_report(pcm->jack, 0, SND_JACK_AVOUT); - if (pcm->jack_event > 0) - pcm->jack_event--; + if (port->jack_event == 1) + snd_soc_jack_report(&port->jack, 0, SND_JACK_AVOUT); + port->jack_event = 0; } - - snd_soc_dapm_sync(port->dapm); } /* MST supported verbs */ @@ -855,7 +847,6 @@ static int hdac_hdmi_pin_mux_widget_event(struct snd_soc_dapm_widget *w, static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - int ret; struct hdac_hdmi_port *p, *p_next; struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; struct snd_soc_dapm_widget *w = snd_soc_dapm_kcontrol_widget(kcontrol); @@ -865,6 +856,7 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, struct hdac_hdmi_priv *hdmi = hdev_to_hdmi_priv(hdev); struct hdac_hdmi_pcm *pcm = NULL; const char *cvt_name = e->texts[ucontrol->value.enumerated.item[0]]; + int ret; ret = snd_soc_dapm_put_enum_double(kcontrol, ucontrol); if (ret < 0) @@ -880,13 +872,12 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, list_for_each_entry_safe(p, p_next, &pcm->port_list, head) { if (p == port && p->id == port->id && - p->pin == port->pin) { - hdac_hdmi_jack_report(pcm, port, false); + p->pin == port->pin) list_del(&p->head); - } } } + /* * Jack status is not reported during device probe as the * PCMs are not registered by then. So report it here. @@ -895,7 +886,6 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, if (!strcmp(cvt_name, pcm->cvt->name)) { list_add_tail(&port->head, &pcm->port_list); if (port->eld.monitor_present && port->eld.eld_valid) { - hdac_hdmi_jack_report(pcm, port, true); mutex_unlock(&hdmi->pin_mutex); return ret; } @@ -1247,27 +1237,18 @@ static void hdac_hdmi_present_sense(struct hdac_hdmi_pin *pin, } pcm = hdac_hdmi_get_pcm(hdev, port); - if (!port->eld.monitor_present || !port->eld.eld_valid) { - dev_err(&hdev->dev, "%s: disconnect for pin:port %d:%d\n", __func__, pin->nid, port->id); - /* - * PCMs are not registered during device probe, so don't - * report jack here. It will be done in usermode mux - * control select. - */ - if (pcm) - hdac_hdmi_jack_report(pcm, port, false); + hdac_hdmi_jack_report(port, false); mutex_unlock(&hdmi->pin_mutex); return; } if (port->eld.monitor_present && port->eld.eld_valid) { - if (pcm) - hdac_hdmi_jack_report(pcm, port, true); + hdac_hdmi_jack_report(port, true); print_hex_dump_debug("ELD: ", DUMP_PREFIX_OFFSET, 16, 1, port->eld.eld_buffer, port->eld.eld_size, false); @@ -1299,6 +1280,7 @@ static int hdac_hdmi_add_ports(struct hdac_device *hdev, } pin->ports = ports; pin->num_ports = max_ports; + return 0; } @@ -1664,6 +1646,30 @@ static int create_fill_jack_kcontrols(struct snd_soc_card *card, return snd_soc_add_card_controls(card, kc, i); } +static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, + struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) +{ + int i; + struct hdac_hdmi_pin *pin; + + list_for_each_entry(pin, &hdmi->pin_list, head) { + if (detect_pin_caps) { + + if (hdac_hdmi_get_port_len(hdev, pin->nid) == 0) + pin->mst_capable = false; + else + pin->mst_capable = true; + } + + for (i = 0; i < pin->num_ports; i++) { + if (!pin->mst_capable && i > 0) + continue; + + hdac_hdmi_present_sense(pin, &pin->ports[i]); + } + } +} + int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm) { @@ -1673,6 +1679,7 @@ int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_widget *widgets; struct snd_soc_dapm_route *route; char w_name[NAME_SIZE]; + char jack_name[NAME_SIZE]; int i = 0, j, ret; widgets = devm_kcalloc(dapm->dev, hdmi->num_ports, @@ -1701,6 +1708,23 @@ int hdac_hdmi_jack_port_init(struct snd_soc_component *component, pin->ports[j].jack_pin = widgets[i].name; pin->ports[j].dapm = dapm; + /* create pin-port jack */ + snprintf(jack_name, sizeof(jack_name), + "HDMI/DP, pin:port=%d:%d Jack", + pin->nid, pin->ports[j].id); + ret = snd_soc_card_jack_new(dapm->card, jack_name, + SND_JACK_AVOUT, + &(pin->ports[j].jack), + NULL, 0); + if (ret) + return ret; + + ret = snd_jack_add_new_kctl(pin->ports[j].jack.jack, + jack_name, SND_JACK_AVOUT); + if (ret) + dev_warn(&hdev->dev, "failed creating Jack kctl\n"); + + /* add to route from Jack widget to output */ hdac_hdmi_fill_route(&route[i], pin->ports[j].jack_pin, NULL, pin->ports[j].output_pin, NULL); @@ -1708,6 +1732,8 @@ int hdac_hdmi_jack_port_init(struct snd_soc_component *component, i++; } } + /* now jack is ready, let's update the status */ + hdac_hdmi_present_sense_all_pins(hdev, hdmi, true); /* Add Route from Jack widget to the output widget */ ret = snd_soc_dapm_new_controls(dapm, widgets, hdmi->num_ports); @@ -1739,6 +1765,10 @@ int hdac_hdmi_jack_port_init(struct snd_soc_component *component, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_port_init); +/* + * TODO: maybe we should rename this function name + * as it is not related to jack init any more. + */ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, struct snd_soc_jack *jack) { @@ -1758,8 +1788,6 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, return -ENOMEM; pcm->pcm_id = device; pcm->cvt = hdmi->dai_map[dai->id].cvt; - pcm->jack_event = 0; - pcm->jack = jack; mutex_init(&pcm->lock); INIT_LIST_HEAD(&pcm->port_list); snd_pcm = hdac_hdmi_get_pcm_from_id(dai->component->card, device); @@ -1779,30 +1807,6 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); -static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, - struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) -{ - int i; - struct hdac_hdmi_pin *pin; - - list_for_each_entry(pin, &hdmi->pin_list, head) { - if (detect_pin_caps) { - - if (hdac_hdmi_get_port_len(hdev, pin->nid) == 0) - pin->mst_capable = false; - else - pin->mst_capable = true; - } - - for (i = 0; i < pin->num_ports; i++) { - if (!pin->mst_capable && i > 0) - continue; - - hdac_hdmi_present_sense(pin, &pin->ports[i]); - } - } -} - static int hdmi_codec_probe(struct snd_soc_component *component) { struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); @@ -1837,7 +1841,6 @@ static int hdmi_codec_probe(struct snd_soc_component *component) return ret; } - hdac_hdmi_present_sense_all_pins(hdev, hdmi, true); /* Imp: Store the card pointer in hda_codec */ hdmi->card = dapm->card->snd_card; diff --git a/sound/soc/intel/boards/bxt_da7219_max98357a.c b/sound/soc/intel/boards/bxt_da7219_max98357a.c index 5cadb7f..c69cfa9 100644 --- a/sound/soc/intel/boards/bxt_da7219_max98357a.c +++ b/sound/soc/intel/boards/bxt_da7219_max98357a.c @@ -573,14 +573,12 @@ static const struct x86_cpu_id glk_ids[] = { {} }; -#define NAME_SIZE 32 static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_card_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; if (x86_match_cpu(glk_ids)) snd_soc_dapm_add_routes(&card->dapm, gemini_map, @@ -591,14 +589,6 @@ static int bxt_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &broxton_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &broxton_hdmi[i]); diff --git a/sound/soc/intel/boards/bxt_rt298.c b/sound/soc/intel/boards/bxt_rt298.c index e91057f..d1372891 100644 --- a/sound/soc/intel/boards/bxt_rt298.c +++ b/sound/soc/intel/boards/bxt_rt298.c @@ -507,25 +507,15 @@ static struct snd_soc_dai_link broxton_rt298_dais[] = { }, }; -#define NAME_SIZE 32 static int bxt_card_late_probe(struct snd_soc_card *card) { struct bxt_rt286_private *ctx = snd_soc_card_get_drvdata(card); struct bxt_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &broxton_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &broxton_hdmi[i]); diff --git a/sound/soc/intel/boards/glk_rt5682_max98357a.c b/sound/soc/intel/boards/glk_rt5682_max98357a.c index d17126f..2a35974 100644 --- a/sound/soc/intel/boards/glk_rt5682_max98357a.c +++ b/sound/soc/intel/boards/glk_rt5682_max98357a.c @@ -29,7 +29,6 @@ #define MAXIM_DEV0_NAME "MX98357A:00" #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 -#define NAME_SIZE 32 static struct snd_soc_jack geminilake_hdmi[3]; @@ -523,21 +522,12 @@ static int glk_card_late_probe(struct snd_soc_card *card) { struct glk_card_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_component *component = NULL; - char jack_name[NAME_SIZE]; struct glk_hdmi_pcm *pcm; int err = 0; int i = 0; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &geminilake_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &geminilake_hdmi[i]); diff --git a/sound/soc/intel/boards/kbl_da7219_max98357a.c b/sound/soc/intel/boards/kbl_da7219_max98357a.c index 07491a0..f293307 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98357a.c +++ b/sound/soc/intel/boards/kbl_da7219_max98357a.c @@ -519,25 +519,15 @@ static struct snd_soc_dai_link kabylake_dais[] = { }, }; -#define NAME_SIZE 32 static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); struct kbl_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &skylake_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &skylake_hdmi[i]); diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index f72a7bf..84d3609 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -34,7 +34,6 @@ #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 -#define NAME_SIZE 32 static struct snd_soc_card *kabylake_audio_card; static struct snd_soc_jack kabylake_hdmi[3]; @@ -952,18 +951,9 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) struct kbl_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &kabylake_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &kabylake_hdmi[i]); diff --git a/sound/soc/intel/boards/kbl_rt5660.c b/sound/soc/intel/boards/kbl_rt5660.c index 3255e00..06411bf 100644 --- a/sound/soc/intel/boards/kbl_rt5660.c +++ b/sound/soc/intel/boards/kbl_rt5660.c @@ -447,25 +447,15 @@ static struct snd_soc_dai_link kabylake_rt5660_dais[] = { }; -#define NAME_SIZE 32 static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); struct kbl_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &skylake_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &skylake_hdmi[i]); diff --git a/sound/soc/intel/boards/kbl_rt5663_max98927.c b/sound/soc/intel/boards/kbl_rt5663_max98927.c index d714752..9c43c6c 100644 --- a/sound/soc/intel/boards/kbl_rt5663_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_max98927.c @@ -898,25 +898,15 @@ static struct snd_soc_dai_link kabylake_5663_dais[] = { }, }; -#define NAME_SIZE 32 static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_rt5663_private *ctx = snd_soc_card_get_drvdata(card); struct kbl_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &skylake_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &skylake_hdmi[i]); diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 879f142..8fe4007 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -40,7 +40,6 @@ #define RT5663_DEV_NAME "i2c-10EC5663:00" #define RT5514_AIF1_BCLK_FREQ (48000 * 8 * 16) #define RT5514_AIF1_SYSCLK_FREQ 12288000 -#define NAME_SIZE 32 #define DMIC_CH(p) p->list[p->count-1] @@ -600,18 +599,10 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) struct kbl_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP,pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &ctx->kabylake_hdmi[i], - NULL, 0); - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &ctx->kabylake_hdmi[i]); if (err < 0) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index 8b68f41..0f57fc2 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -118,19 +118,10 @@ int skl_hda_hdmi_jack_init(struct snd_soc_card *card) struct skl_hda_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_component *component = NULL; struct skl_hda_hdmi_pcm *pcm; - char jack_name[NAME_SIZE]; int err; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &pcm->hdmi_jack, - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &pcm->hdmi_jack); diff --git a/sound/soc/intel/boards/skl_nau88l25_max98357a.c b/sound/soc/intel/boards/skl_nau88l25_max98357a.c index 0922106..aca426f 100644 --- a/sound/soc/intel/boards/skl_nau88l25_max98357a.c +++ b/sound/soc/intel/boards/skl_nau88l25_max98357a.c @@ -587,26 +587,15 @@ static struct snd_soc_dai_link skylake_dais[] = { }, }; -#define NAME_SIZE 32 static int skylake_card_late_probe(struct snd_soc_card *card) { struct skl_nau8825_private *ctx = snd_soc_card_get_drvdata(card); struct skl_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, - &skylake_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &skylake_hdmi[i]); diff --git a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c index 8433c52..788a837 100644 --- a/sound/soc/intel/boards/skl_nau88l25_ssm4567.c +++ b/sound/soc/intel/boards/skl_nau88l25_ssm4567.c @@ -638,26 +638,15 @@ static struct snd_soc_dai_link skylake_dais[] = { }, }; -#define NAME_SIZE 32 static int skylake_card_late_probe(struct snd_soc_card *card) { struct skl_nau88125_private *ctx = snd_soc_card_get_drvdata(card); struct skl_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, - &skylake_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &skylake_hdmi[i]); diff --git a/sound/soc/intel/boards/skl_rt286.c b/sound/soc/intel/boards/skl_rt286.c index 0e1818d..5d245f5 100644 --- a/sound/soc/intel/boards/skl_rt286.c +++ b/sound/soc/intel/boards/skl_rt286.c @@ -473,25 +473,15 @@ static struct snd_soc_dai_link skylake_rt286_dais[] = { }, }; -#define NAME_SIZE 32 static int skylake_card_late_probe(struct snd_soc_card *card) { struct skl_rt286_private *ctx = snd_soc_card_get_drvdata(card); struct skl_hdmi_pcm *pcm; struct snd_soc_component *component = NULL; int err, i = 0; - char jack_name[NAME_SIZE]; list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &skylake_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &skylake_hdmi[i]); diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index f28fb98..660e4a6 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -22,8 +22,6 @@ #include "../../codecs/rt5682.h" #include "../../codecs/hdac_hdmi.h" -#define NAME_SIZE 32 - #define SOF_RT5682_SSP_CODEC(quirk) ((quirk) & GENMASK(2, 0)) #define SOF_RT5682_SSP_CODEC_MASK (GENMASK(2, 0)) #define SOF_RT5682_MCLK_EN BIT(3) @@ -216,7 +214,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) { struct sof_card_private *ctx = snd_soc_card_get_drvdata(card); struct snd_soc_component *component = NULL; - char jack_name[NAME_SIZE]; struct sof_hdmi_pcm *pcm; int err = 0; int i = 0; @@ -227,14 +224,6 @@ static int sof_card_late_probe(struct snd_soc_card *card) list_for_each_entry(pcm, &ctx->hdmi_pcm_list, head) { component = pcm->codec_dai->component; - snprintf(jack_name, sizeof(jack_name), - "HDMI/DP, pcm=%d Jack", pcm->device); - err = snd_soc_card_jack_new(card, jack_name, - SND_JACK_AVOUT, &sof_hdmi[i], - NULL, 0); - - if (err) - return err; err = hdac_hdmi_jack_init(pcm->codec_dai, pcm->device, &sof_hdmi[i]); -- 2.7.4

2 1

[alsa-devel] [PATCH v2 00/15] soundwire: corrections to ACPI/DisCo/Intel support
by Pierre-Louis Bossart 27 May '19

27 May '19

Now that we are done with cleanups, we can start fixing the code with actual semantic or functional changes. This patchset corrects issues with Intel BIOS and hardware properties that prevented a successful init, aligns the code with the MIPI DisCo spec, adds rate-limiting for frequent errors and adds checks on number of links and PDIs. With all these changes, the hardware can be initialized correctly and modules can be added/removed without issues on WhiskyLake and IceLake. Parts of this code was initially written by my Intel colleagues Vinod Koul, Sanyog Kale, Shreyas Nc and Hardik Shah, who are either no longer with Intel or no longer involved in SoundWire development. When relevant, I explictly added a note in commit messages to give them credit for their hard work, but I removed their signed-off-by tags to avoid email bounces and avoid spamming them forever with SoundWire patches. Changes since v2: Feedback from Vinod: 1. improve the SoundWire controller search without magic values 2. split patches as needed Other additions rate-limiting to avoid flooding dmesg logs provide better Slave status on errors more checks on links and PDIs Pierre-Louis Bossart (15): soundwire: intel: filter SoundWire controller device search soundwire: mipi_disco: fix master/link error soundwire: add port-related definitions soundwire: remove master data port properties soundwire: mipi-disco: remove master_count property for masters soundwire: rename 'freq' fields soundwire: mipi-disco: fix clock stop modes soundwire: clarify comment soundwire: rename/clarify MIPI DisCo properties soundwire: cadence_master: use rate_limited dynamic debug soundwire: cadence_master: log Slave status mask on errors soundwire: cadence_master: check the number of bidir PDIs soundwire: Intel: add log for number of PCM and PDM PDIs soundwire: fix typo in comments soundwire: intel_init: add checks on link numbers drivers/soundwire/bus.c | 6 +- drivers/soundwire/cadence_master.c | 29 +++++----- drivers/soundwire/intel.c | 17 ++++-- drivers/soundwire/intel.h | 2 +- drivers/soundwire/intel_init.c | 25 ++++++++- drivers/soundwire/mipi_disco.c | 35 ++++++------ drivers/soundwire/stream.c | 8 +-- include/linux/soundwire/sdw.h | 88 +++++++++++++++++++++++------- 8 files changed, 147 insertions(+), 63 deletions(-) -- 2.20.1

2 16

[alsa-devel] [PATCH] soundwire: cdns: Fix compilation error on arm64
by Jan Kotas 27 May '19

27 May '19

On arm64 the cadence_master.c file doesn't compile. readl and writel are undefined. This patch fixes that by including io.h. Signed-off-by: Jan Kotas <jank(a)cadence.com> --- drivers/soundwire/cadence_master.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/cadence_master.c b/drivers/soundwire/cadence_master.c index cb6a331f4..0b5bcc209 100644 --- a/drivers/soundwire/cadence_master.c +++ b/drivers/soundwire/cadence_master.c @@ -9,6 +9,7 @@ #include <linux/delay.h> #include <linux/device.h> #include <linux/interrupt.h> +#include <linux/io.h> #include <linux/module.h> #include <linux/mod_devicetable.h> #include <linux/soundwire/sdw_registers.h> -- 2.15.0

4 6

[alsa-devel] [PATCH] soundwire: stream: fix out of boundary access on port properties
by Srinivas Kandagatla 27 May '19

27 May '19

Assigning local iterator to array element and using it again for indexing would cross the array boundary. Fix this by directly referring array element without using the local variable. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/stream.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index d01060dbee96..544925ff0b40 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -1406,9 +1406,7 @@ struct sdw_dpn_prop *sdw_get_slave_dpn_prop(struct sdw_slave *slave, } for (i = 0; i < num_ports; i++) { - dpn_prop = &dpn_prop[i]; - - if (dpn_prop->num == port_num) + if (dpn_prop[i].num == port_num) return &dpn_prop[i]; } -- 2.21.0

3 2

[alsa-devel] [PATCH][next] ALSA: firewire-lib: remove redundant assignment to cip_header
by Colin King 25 May '19

25 May '19

From: Colin Ian King <colin.king(a)canonical.com> The assignement to cip_header is redundant as the value never read and it is being re-assigned in the if and else paths of the following if statement. Clean up the code by removing it. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/firewire/amdtp-stream.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 2d9c764061d1..4236955bbf57 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -675,7 +675,6 @@ static int handle_in_packet(struct amdtp_stream *s, unsigned int cycle, return -EIO; } - cip_header = ctx_header + 2; if (!(s->flags & CIP_NO_HEADER)) { cip_header = &ctx_header[2]; err = check_cip_header(s, cip_header, payload_length, -- 2.20.1

2 1

[alsa-devel] arecord doesnt record into file
by Aditya Paluri 25 May '19

25 May '19

Hello all, I use the below command: arecord --device plughw:0,2 -r 48000 -f S16_LE -c 2 recorded_sound.wav And as I see the audio stream getting dumped onto the DRAM/memory, I donot see recorded_sound.wav contain any data on the file system. I have tried to use strace and I could only see "write" system call happen on the file only for writing the wav format header. ioctl(4, _IOC(_IOC_READ|_IOC_WRITE, 0x41, 0x23, 0x88), 0x1ec27190) = 0 newfstatat(AT_FDCWD, "/home/root/recorded_wave.wav", 0x7fe8bfeb50, AT_SYMLINK_NOFOLLOW) = -1 ENOENT (No such file or directory) openat(AT_FDCWD, "/home/root/recorded_wave.wav", O_WRONLY|O_CREAT, 0644) = 3 write(3, "RIFF$\0\0\200WAVE", 12) = 12 write(3, "fmt \20\0\0\0", 8) = 8 write(3, "\1\0\2\0\200\273\0\0\0\356\2\0\4\0\20\0", 16) = 16 write(3, "data\0\0\0\200", 8) = 8 ioctl(4, _IOC(_IOC_READ|_IOC_WRITE, 0x41, 0x23, 0x88), 0x1ec27190) = 0 ioctl(4, _IOC(_IOC_READ, 0x41, 0x51, 0x18)[ 24.145583] [ 24.587069] [0 I also see EPIPE error due to overrun. , 0x7fe8bfeaa8) = -1 EPIPE (Broken pipe) ioctl(4, _IOC(_IOC_READ|_IOC_WRITE, 0x41, 0x24, 0x98), 0x7fe8bfe9b0) = 0 write(2, "overrun!!! (at least 0.717 ms lo"..., 36overrun!!! (at least 0.717 ms long) ) = 36 Could you please help me in understanding why the recorded_sound.wav doesn't contain any data ? BR\Thanks. Aditya Paluri.

2 1

[alsa-devel] [PATCH] tegra_wm9712: Fix a memory leaking bug in tegra_wm9712_driver_probe()
by Gen Zhang 24 May '19

24 May '19

In tegra_wm9712_driver_probe(), 'machine->codec' is allocated by platform_device_alloc(). When it is NULL, function returns ENOMEM. However, 'machine' is allocated by devm_kzalloc() before this site. Thus we should free 'machine' before function ends to prevent memory leaking. Further, we should free 'machine->util_data', 'machine->codec' and 'machine' before this function normally ends to prevent memory leaking. Signed-off-by: Gen Zhang <blackgod016574(a)gmail.com> --- diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 864a334..295c41d 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -86,7 +86,8 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) machine->codec = platform_device_alloc("wm9712-codec", -1); if (!machine->codec) { dev_err(&pdev->dev, "Can't allocate wm9712 platform device\n"); - return -ENOMEM; + ret = -ENOMEM; + goto codec_free; } ret = platform_device_add(machine->codec); @@ -127,6 +128,10 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) goto asoc_utils_fini; } + tegra_asoc_utils_fini(&machine->util_data); + platform_device_del(machine->codec); + platform_device_put(machine->codec); + devm_kfree(&pdev->dev, machine); return 0; asoc_utils_fini: @@ -135,6 +140,8 @@ static int tegra_wm9712_driver_probe(struct platform_device *pdev) platform_device_del(machine->codec); codec_put: platform_device_put(machine->codec); +codec_free: + devm_kfree(&pdev->dev, machine); return ret; } ---

2 6

Re: [alsa-devel] [PATCH 2/5] ASoC: madera: Add common support for Cirrus Logic Madera codecs
by Charles Keepax 24 May '19

24 May '19

On Fri, May 24, 2019 at 03:56:03PM +0100, Mark Brown wrote: > On Fri, May 24, 2019 at 11:41:55AM +0100, Charles Keepax wrote: > > > + /* > > + * Just read a register a few times to ensure the internal > > + * oscillator sends out a few clocks. > > + */ > > + for (i = 0; i < 4; i++) { > > + ret = regmap_read(madera->regmap, MADERA_SOFTWARE_RESET, &val); > > + if (ret) > > + dev_err(madera->dev, > > + "%s Failed to read register: %d (%d)\n", > > + __func__, ret, i); > > Why use %s to format the __func__ rather than just concatenate? > Really the error message is quite out of character in general, we don't normally print the func and it doesn't really say what failed. I will refactor it. > > + } > > + > > + udelay(300); > > So we read the register a few times then add a few hundred us of delay > after? Surely that delay is going to be negligable compared to the time > spent on I/O? > This is one of those technically situations. We have to do the reads and we have to have at least 300uS after those. Normally there will be enough slack in the system but better to guarantee it. Maybe a comment? > > +int madera_sysclk_ev(struct snd_soc_dapm_widget *w, > > + struct snd_kcontrol *kcontrol, int event) > > +{ > > + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); > > + struct madera_priv *priv = snd_soc_component_get_drvdata(component); > > + > > + madera_spin_sysclk(priv); > > + > > + return 0; > > +} > > +EXPORT_SYMBOL_GPL(madera_sysclk_ev); > > This will delay both before and after every power up and power down. > Are you sure that makes sense? > Regrettably yes, or well at least I am sure that despite a lot of complaining this is what the hardware guys insisted needed done. > > + > > + ret = madera_check_speaker_overheat(madera, &warn, &shutdown); > > + if (ret) > > + shutdown = true; /* for safety attempt to shutdown on error */ > > + > > + if (shutdown) { > > + dev_crit(madera->dev, "Thermal shutdown\n"); > > + ret = regmap_update_bits(madera->regmap, > > + MADERA_OUTPUT_ENABLES_1, > > + MADERA_OUT4L_ENA | > > + MADERA_OUT4R_ENA, 0); > > + if (ret != 0) > > + dev_crit(madera->dev, > > + "Failed to disable speaker outputs: %d\n", > > + ret); > > + } else if (warn) { > > + dev_crit(madera->dev, "Thermal warning\n"); > > + } > > + > > + return IRQ_HANDLED; > > We will flag the interrupt as handled if there was neither a warning nor > a critical overheat? I'd expect some warning about a spurious interrupt > at least. > Yeah I will update that one. > > +static int madera_get_variable_u32_array(struct madera_priv *priv, > > + const char *propname, > > + u32 *dest, > > + int n_max, > > + int multiple) > > +{ > > + struct madera *madera = priv->madera; > > + int n, ret; > > + > > + n = device_property_read_u32_array(madera->dev, propname, NULL, 0); > > + if (n == -EINVAL) { > > + return 0; /* missing, ignore */ > > + } else if (n < 0) { > > + dev_warn(madera->dev, "%s malformed (%d)\n", > > + propname, n); > > + return n; > > + } else if ((n % multiple) != 0) { > > + dev_warn(madera->dev, "%s not a multiple of %d entries\n", > > + propname, multiple); > > + return -EINVAL; > > + } > > + > > + if (n > n_max) > > + n = n_max; > > + > > + ret = device_property_read_u32_array(madera->dev, propname, dest, n); > > + > > + if (ret < 0) > > + return ret; > > + else > > + return n; > > +} > > This feels like it should perhaps be a generic OF helper function - > there's nothing driver specific I'm seeing here and arrays that need to > be a multiple of N entries aren't that uncommon I think. > I will have a look and see if we might do a generic helper function for this one. > > + mutex_lock(&priv->rate_lock); > > + cached_rate = priv->adsp_rate_cache[adsp_num]; > > + mutex_unlock(&priv->rate_lock); > > What's this lock protecting? The value can we read can change as soon > as the lock is released and we're just reading a single word here rather > than traversing a data structure that might change under us or > something. > Overall the lock protects anything that changes a rate on the chip, since we have to do that weird spin_sysclk thing before/after anything that does that. It might normally be safe without the lock here I guess, since you are only really protecting against the reads of adsp_rate_cache. But suppose that might depend on the architecture if the read/writes arn't effectively atomic (although that is probably unlikely on most modern architectures). > > +void madera_destroy_bus_error_irq(struct madera_priv *priv, int dsp_num) > > +{ > > + struct madera *madera = priv->madera; > > + > > + madera_free_irq(madera, > > + madera_dsp_bus_error_irqs[dsp_num], > > + &priv->adsp[dsp_num]); > > +} > > +EXPORT_SYMBOL_GPL(madera_destroy_bus_error_irq); > > We use free rather than destroy normally? > Yup will fix that. > > +static const char * const madera_dfc_width_text[MADERA_DFC_WIDTH_ENUM_SIZE] = { > > + "8bit", "16bit", "20bit", "24bit", "32bit", > > +}; > > Spaces might make these more readable. > No problem to fix that too. > > +static void madera_sleep(unsigned int delay) > > +{ > > + if (delay < 20) { > > + delay *= 1000; > > + usleep_range(delay, delay + 500); > > + } else { > > + msleep(delay); > > + } > > +} > > This feels like it might make sense as a helper function as well - I > could've sworn there was one already but I can't immediately find it. I do vaguely remember some discussion that it was preferred that people had to think about which delay function to call, rather than having a generic helper. TBF I could probably just switch it all to use msleep, for the most part this is a quite small optimisation of path bring up (although it was asked for by one of our customers). Thanks, Charles

1 0

Re: [alsa-devel] [PATCH 2/5] ASoC: madera: Add common support for Cirrus Logic Madera codecs
by Richard Fitzgerald 24 May '19

24 May '19

On 24/05/19 15:56, Mark Brown wrote: > On Fri, May 24, 2019 at 11:41:55AM +0100, Charles Keepax wrote: > >> + /* >> + * Just read a register a few times to ensure the internal >> + * oscillator sends out a few clocks. >> + */ >> + for (i = 0; i < 4; i++) { >> + ret = regmap_read(madera->regmap, MADERA_SOFTWARE_RESET, &val); >> + if (ret) >> + dev_err(madera->dev, >> + "%s Failed to read register: %d (%d)\n", >> + __func__, ret, i); > > Why use %s to format the __func__ rather than just concatenate? GCC docs say that it's a magic variable so cannot be concatenated with string literals. Though I never tried concatenation to see if it works. > >> + } >> + >> + udelay(300); > > So we read the register a few times then add a few hundred us of delay > after? Surely that delay is going to be negligable compared to the time > spent on I/O? The register reads are to create clock cycles in the silicon, not to generate delay. > >> +int madera_sysclk_ev(struct snd_soc_dapm_widget *w, >> + struct snd_kcontrol *kcontrol, int event) >> +{ >> + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); >> + struct madera_priv *priv = snd_soc_component_get_drvdata(component); >> + >> + madera_spin_sysclk(priv); >> + >> + return 0; >> +} >> +EXPORT_SYMBOL_GPL(madera_sysclk_ev); > > This will delay both before and after every power up and power down. > Are you sure that makes sense? > I think that's correct but we can re-check with hardware people. It's not just a delay, it needs to ensure there are always a certain number of SYSCLK cycles in the hardware to avoid leaving certain state machines in limbo. >> + >> + ret = madera_check_speaker_overheat(madera, &warn, &shutdown); >> + if (ret) >> + shutdown = true; /* for safety attempt to shutdown on error */ >> + >> + if (shutdown) { >> + dev_crit(madera->dev, "Thermal shutdown\n"); >> + ret = regmap_update_bits(madera->regmap, >> + MADERA_OUTPUT_ENABLES_1, >> + MADERA_OUT4L_ENA | >> + MADERA_OUT4R_ENA, 0); >> + if (ret != 0) >> + dev_crit(madera->dev, >> + "Failed to disable speaker outputs: %d\n", >> + ret); >> + } else if (warn) { >> + dev_crit(madera->dev, "Thermal warning\n"); >> + } >> + >> + return IRQ_HANDLED; > > We will flag the interrupt as handled if there was neither a warning nor > a critical overheat? I'd expect some warning about a spurious interrupt > at least. > >> +static int madera_get_variable_u32_array(struct madera_priv *priv, >> + const char *propname, >> + u32 *dest, >> + int n_max, >> + int multiple) >> +{ >> + struct madera *madera = priv->madera; >> + int n, ret; >> + >> + n = device_property_read_u32_array(madera->dev, propname, NULL, 0); >> + if (n == -EINVAL) { >> + return 0; /* missing, ignore */ >> + } else if (n < 0) { >> + dev_warn(madera->dev, "%s malformed (%d)\n", >> + propname, n); >> + return n; >> + } else if ((n % multiple) != 0) { >> + dev_warn(madera->dev, "%s not a multiple of %d entries\n", >> + propname, multiple); >> + return -EINVAL; >> + } >> + >> + if (n > n_max) >> + n = n_max; >> + >> + ret = device_property_read_u32_array(madera->dev, propname, dest, n); >> + >> + if (ret < 0) >> + return ret; >> + else >> + return n; >> +} > > This feels like it should perhaps be a generic OF helper function - > there's nothing driver specific I'm seeing here and arrays that need to > be a multiple of N entries aren't that uncommon I think. > >> + mutex_lock(&priv->rate_lock); >> + cached_rate = priv->adsp_rate_cache[adsp_num]; >> + mutex_unlock(&priv->rate_lock); > > What's this lock protecting? The value can we read can change as soon > as the lock is released and we're just reading a single word here rather > than traversing a data structure that might change under us or > something. > >> +void madera_destroy_bus_error_irq(struct madera_priv *priv, int dsp_num) >> +{ >> + struct madera *madera = priv->madera; >> + >> + madera_free_irq(madera, >> + madera_dsp_bus_error_irqs[dsp_num], >> + &priv->adsp[dsp_num]); >> +} >> +EXPORT_SYMBOL_GPL(madera_destroy_bus_error_irq); > > We use free rather than destroy normally? > >> +static const char * const madera_dfc_width_text[MADERA_DFC_WIDTH_ENUM_SIZE] = { >> + "8bit", "16bit", "20bit", "24bit", "32bit", >> +}; > > Spaces might make these more readable. > >> +static void madera_sleep(unsigned int delay) >> +{ >> + if (delay < 20) { >> + delay *= 1000; >> + usleep_range(delay, delay + 500); >> + } else { >> + msleep(delay); >> + } >> +} > > This feels like it might make sense as a helper function as well - I > could've sworn there was one already but I can't immediately find it. >

1 1

[alsa-devel] [PATCH v2] ASoC: drop COMPILE_TEST for sound/soc/intel/boards/ that use X86_INTEL_LPSS
by Randy Dunlap 24 May '19

24 May '19

From: Randy Dunlap <rdunlap(a)infradead.org> This is a partial revert of 164a263bf8d0, which causes build errors on non-X86 platforms (specifically seen on IA64) when COMPILE_TEST is set/enabled. Fixes these build errors (on ia64): i../sound/soc/intel/boards/bxt_da7219_max98357a.c:19:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcr_rt5640.c:31:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcr_rt5651.c:33:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/cht_bsw_rt5645.c:29:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcht_es8316.c:33:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcht_da7213.c:26:10: fatal error: asm/platform_sst_audio.h: No such file or directory #include <asm/platform_sst_audio.h> Fixes: 164a263bf8d0 ("ASoC: Intel: Make boards more available for compile test") Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/boards/Kconfig | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) --- lnx-52-rc1.orig/sound/soc/intel/boards/Kconfig +++ lnx-52-rc1/sound/soc/intel/boards/Kconfig @@ -18,7 +18,7 @@ config SND_SOC_INTEL_HASWELL_MACH tristate "Haswell Lynxpoint" depends on I2C depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_RT5640 help This adds support for the Lynxpoint Audio DSP on Intel(R) Haswell @@ -35,7 +35,7 @@ config SND_SOC_INTEL_BDW_RT5677_MACH depends on I2C depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST depends on GPIOLIB || COMPILE_TEST - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_RT5677 help This adds support for Intel Broadwell platform based boards with @@ -47,7 +47,7 @@ config SND_SOC_INTEL_BROADWELL_MACH tristate "Broadwell Wildcatpoint" depends on I2C depends on I2C_DESIGNWARE_PLATFORM || COMPILE_TEST - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_RT286 help This adds support for the Wilcatpoint Audio DSP on Intel(R) Broadwell @@ -61,7 +61,7 @@ if SND_SOC_INTEL_BAYTRAIL config SND_SOC_INTEL_BYT_MAX98090_MACH tristate "Baytrail with MAX98090 codec" depends on I2C - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_MAX98090 help This adds audio driver for Intel Baytrail platform based boards @@ -72,7 +72,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BYT_RT5640_MACH tristate "Baytrail with RT5640 codec" depends on I2C - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_RT5640 help This adds audio driver for Intel Baytrail platform based boards @@ -86,7 +86,7 @@ if SND_SST_ATOM_HIFI2_PLATFORM || SND_SO config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "Baytrail and Baytrail-CR with RT5640 codec" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_ACPI select SND_SOC_RT5640 help @@ -98,7 +98,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "Baytrail and Baytrail-CR with RT5651 codec" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_ACPI select SND_SOC_RT5651 help @@ -110,7 +110,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH config SND_SOC_INTEL_CHT_BSW_RT5672_MACH tristate "Cherrytrail & Braswell with RT5672 codec" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_ACPI select SND_SOC_RT5670 help @@ -122,7 +122,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5672_MACH config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_ACPI select SND_SOC_RT5645 help @@ -134,7 +134,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH config SND_SOC_INTEL_CHT_BSW_MAX98090_TI_MACH tristate "Cherrytrail & Braswell with MAX98090 & TI codec" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_MAX98090 select SND_SOC_TS3A227E help @@ -146,7 +146,7 @@ config SND_SOC_INTEL_CHT_BSW_MAX98090_TI config SND_SOC_INTEL_CHT_BSW_NAU8824_MACH tristate "Cherrytrail & Braswell with NAU88L24 codec" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_ACPI select SND_SOC_NAU8824 help @@ -158,7 +158,7 @@ config SND_SOC_INTEL_CHT_BSW_NAU8824_MAC config SND_SOC_INTEL_BYT_CHT_DA7213_MACH tristate "Baytrail & Cherrytrail with DA7212/7213 codec" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_ACPI select SND_SOC_DA7213 help @@ -170,7 +170,7 @@ config SND_SOC_INTEL_BYT_CHT_DA7213_MACH config SND_SOC_INTEL_BYT_CHT_ES8316_MACH tristate "Baytrail & Cherrytrail with ES8316 codec" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS select SND_SOC_ACPI select SND_SOC_ES8316 help @@ -186,7 +186,7 @@ if SND_SST_ATOM_HIFI2_PLATFORM config SND_SOC_INTEL_BYT_CHT_NOCODEC_MACH tristate "Baytrail & Cherrytrail platform with no codec (MinnowBoard MAX, Up)" depends on I2C && ACPI - depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86_INTEL_LPSS help This adds support for ASoC machine driver for the MinnowBoard Max or Up boards and provides access to I2S signals on the Low-Speed @@ -248,7 +248,7 @@ if SND_SOC_INTEL_APL config SND_SOC_INTEL_BXT_DA7219_MAX98357A_MACH tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI - depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on MFD_INTEL_LPSS select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC

2 1

[alsa-devel] [PATCH] ASoc: fix sound/soc/intel/skylake/slk-ssp-clk.c build error on IA64
by Randy Dunlap 24 May '19

24 May '19

From: Randy Dunlap <rdunlap(a)infradead.org> skl-ssp-clk.c does not build on IA64 because the driver uses the common clock interface, so make the driver depend on COMMON_CLK. Fixes this build error: ../sound/soc/intel/skylake/skl-ssp-clk.c:26:16: error: field 'hw' has incomplete type struct clk_hw hw; ^~ Reported-by: kbuild test robot <lkp(a)intel.com> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/Kconfig | 3 ++- sound/soc/intel/boards/Kconfig | 2 +- 2 files changed, 3 insertions(+), 2 deletions(-) --- lnx-52-rc1.orig/sound/soc/intel/Kconfig +++ lnx-52-rc1/sound/soc/intel/Kconfig @@ -104,7 +104,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_ACPI config SND_SOC_INTEL_SKYLAKE tristate "All Skylake/SST Platforms" depends on PCI && ACPI - select SND_SOC_INTEL_SKL + select SND_SOC_INTEL_SKL if COMMON_CLK select SND_SOC_INTEL_APL select SND_SOC_INTEL_KBL select SND_SOC_INTEL_GLK @@ -120,6 +120,7 @@ config SND_SOC_INTEL_SKYLAKE config SND_SOC_INTEL_SKL tristate "Skylake Platforms" depends on PCI && ACPI + depends on COMMON_CLK select SND_SOC_INTEL_SKYLAKE_FAMILY help If you have a Intel Skylake platform with the DSP enabled --- lnx-52-rc1.orig/sound/soc/intel/boards/Kconfig +++ lnx-52-rc1/sound/soc/intel/boards/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_INTEL_KBL_RT5663_MAX98927 select SND_SOC_MAX98927 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI - select SND_SOC_INTEL_SKYLAKE_SSP_CLK + select SND_SOC_INTEL_SKYLAKE_SSP_CLK if COMMON_CLK help This adds support for ASoC Onboard Codec I2S machine driver. This will create an alsa sound card for RT5663 + MAX98927.

2 3

[alsa-devel] [PATCH] Revert "ASoC: simple-card: Fix configuration of DAI format"
by Jon Hunter 24 May '19

24 May '19

Revert commit 069d037aea98 ("ASoC: simple-card: Fix configuration of DAI format"). During further review, it turns out that the actual issue was caused by an incorrectly formatted device-tree node describing the soundcard. The following is incorrect because the simple-audio-card 'bitclock-master' and 'frame-master' properties should not reference the actual codec phandle ... sound { compatible = "simple-audio-card"; ... => simple-audio-card,bitclock-master = <&codec>; => simple-audio-card,frame-master = <&codec>; ... simple-audio-card,cpu { sound-dai = <&xxx>; }; simple-audio-card,codec { => sound-dai = <&codec>; }; }; Rather, these properties should reference the phandle to the 'simple-audio-card,codec' property as shown below ... sound { compatible = "simple-audio-card"; ... => simple-audio-card,bitclock-master = <&codec>; => simple-audio-card,frame-master = <&codec>; ... simple-audio-card,cpu { sound-dai = <&xxx>; }; => codec: simple-audio-card,codec { /* simple-card wants here */ sound-dai = <&xxx>; /* not here */ }; }; Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/generic/simple-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index c2a2c5fd0801..d16e894fce2b 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -283,6 +283,11 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, codec_dai = dai_props->codec_dai = &priv->dais[li->dais++]; + ret = asoc_simple_parse_daifmt(dev, node, codec, + prefix, &dai_link->dai_fmt); + if (ret < 0) + goto dai_link_of_err; + simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu); @@ -293,11 +298,6 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_parse_daifmt(dev, node, dai_link->codecs->of_node, - prefix, &dai_link->dai_fmt); - if (ret < 0) - goto dai_link_of_err; - ret = asoc_simple_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; -- 2.7.4

2 1

[alsa-devel] [PATCH 1/5] ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs
by Charles Keepax 24 May '19

24 May '19

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> The Cirrus Logic Madera codecs are a family of related codecs with extensive digital and analogue I/O, digital mixing and routing, signal processing and programmable DSPs. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Documentation/devicetree/bindings/sound/madera.txt | 67 ++++++++++++++++++++++ MAINTAINERS | 1 + include/dt-bindings/sound/madera.h | 29 ++++++++++ 3 files changed, 97 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 include/dt-bindings/sound/madera.h diff --git a/Documentation/devicetree/bindings/sound/madera.txt b/Documentation/devicetree/bindings/sound/madera.txt new file mode 100644 index 0000000000000..1114fcf1aa4c2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/madera.txt @@ -0,0 +1,67 @@ +Cirrus Logic Madera class audio codecs + +This describes audio configuration bindings for these codecs. + +See also the core bindings for the parent MFD driver: +See Documentation/devicetree/bindings/mfd/madera.txt + +and defines for values used in these bindings: +include/dt-bindings/sound/madera.h + +These properties are all contained in the parent MFD node. + +Optional properties: + - cirrus,dmic-ref : Indicates how the MICBIAS pins have been externally + connected to DMICs on each input, one cell per input. + <IN1 IN2 IN3 ...> + A value of 0 indicates MICVDD and is the default, other values depend on the + codec: + For CS47L35 one of the CS47L35_DMIC_REF_xxx values + For all other codecs one of the MADERA_DMIC_REF_xxx values + Also see the datasheet for a description of the INn_DMIC_SUP field. + + - cirrus,inmode : A list of input mode settings for each input. A maximum of + 16 cells, with four cells per input in the order INnAL, INnAR INnBL INnBR. + For non-muxed inputs the first two cells for that input set the mode for + the left and right channel and the second two cells must be 0. + For muxed inputs the first two cells for that input set the mode of the + left and right A inputs and the second two cells set the mode of the left + and right B inputs. + Valid mode values are one of the MADERA_INMODE_xxx. If the array is shorter + than the number of inputs the unspecified inputs default to + MADERA_INMODE_DIFF. + + - cirrus,out-mono : Mono bit for each output, must contain six cells if + specified. A non-zero value indicates the corresponding output is mono. + + - cirrus,max-channels-clocked : Maximum number of channels that I2S clocks + will be generated for. Useful when clock master for systems where the I2S + bus has multiple data lines. + One cell for each AIF, use a value of zero for AIFs that should be handled + normally. + + - cirrus,pdm-fmt : PDM speaker data format, must contain 2 cells + (OUT5 and OUT6). See the PDM_SPKn_FMT field in the datasheet for a + description of this value. + The second cell is ignored for codecs that do not have OUT6. + + - cirrus,pdm-mute : PDM mute format, must contain 2 cells + (OUT5 and OUT6). See the PDM_SPKn_CTRL_1 register in the datasheet for a + description of this value. + The second cell is ignored for codecs that do not have OUT6. + +Example: + +cs47l35@0 { + compatible = "cirrus,cs47l35"; + + cirrus,dmic-ref = <0 0 CS47L35_DMIC_REF_MICBIAS1B 0>; + cirrus,inmode = < + MADERA_INMODE_DMIC MADERA_INMODE_DMIC /* IN1A digital */ + MADERA_INMODE_SE MADERA_INMODE_SE /* IN1B single-ended */ + MADERA_INMODE_DIFF MADERA_INMODE_DIFF /* IN2 differential */ + 0 0 /* not used on this codec */ + >; + cirrus,out-mono = <0 0 0 0 0 0>; + cirrus,max-channels-clocked = <2 0 0>; +}; diff --git a/MAINTAINERS b/MAINTAINERS index 5cfbea4ce5750..642cb5610dd50 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3915,6 +3915,7 @@ W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: Documentation/devicetree/bindings/mfd/madera.txt F: Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt +F: include/dt-bindings/sound/madera* F: include/linux/irqchip/irq-madera* F: include/linux/mfd/madera/* F: drivers/gpio/gpio-madera* diff --git a/include/dt-bindings/sound/madera.h b/include/dt-bindings/sound/madera.h new file mode 100644 index 0000000000000..9ff4eae5259b0 --- /dev/null +++ b/include/dt-bindings/sound/madera.h @@ -0,0 +1,29 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Device Tree defines for Madera codecs + * + * Copyright (C) 2016-2017 Cirrus Logic, Inc. and + * Cirrus Logic International Semiconductor Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef DT_BINDINGS_SOUND_MADERA_H +#define DT_BINDINGS_SOUND_MADERA_H + +#define MADERA_INMODE_DIFF 0 +#define MADERA_INMODE_SE 1 +#define MADERA_INMODE_DMIC 2 + +#define MADERA_DMIC_REF_MICVDD 0 +#define MADERA_DMIC_REF_MICBIAS1 1 +#define MADERA_DMIC_REF_MICBIAS2 2 +#define MADERA_DMIC_REF_MICBIAS3 3 + +#define CS47L35_DMIC_REF_MICBIAS1B 1 +#define CS47L35_DMIC_REF_MICBIAS2A 2 +#define CS47L35_DMIC_REF_MICBIAS2B 3 + +#endif -- 2.11.0

1 4

[alsa-devel] [PATCH] pcm030-audio-fabric: Fix a memory leaking bug in pcm030_fabric_probe()
by Gen Zhang 24 May '19

24 May '19

In pcm030_fabric_probe(), 'pdata->codec_device' is allocated by platform_device_alloc(). When this allocation fails, ENOMEM is returned. However, 'pdata' is allocated by devm_kzalloc() before this site. We should free 'pdata' before function ends to prevent memory leaking. Similarly, we should free 'pdata' when 'pdata->codec_device' is NULL. And we should free 'pdata->codec_device' and 'pdata' when 'ret' is error to prevent memory leaking. Signed-off-by: Gen Zhang <blackgod016574(a)gmail.com> --- diff --git a/sound/soc/fsl/pcm030-audio-fabric.c b/sound/soc/fsl/pcm030-audio-fabric.c index a7fe4ad..d2e6eae 100644 --- a/sound/soc/fsl/pcm030-audio-fabric.c +++ b/sound/soc/fsl/pcm030-audio-fabric.c @@ -72,29 +72,43 @@ static int pcm030_fabric_probe(struct platform_device *op) platform_np = of_parse_phandle(np, "asoc-platform", 0); if (!platform_np) { dev_err(&op->dev, "ac97 not registered\n"); - return -ENODEV; + ret = -ENODEV; + goto out_free1; } for_each_card_prelinks(card, i, dai_link) dai_link->platform_of_node = platform_np; ret = request_module("snd-soc-wm9712"); - if (ret) + if (ret) { dev_err(&op->dev, "request_module returned: %d\n", ret); + goto out_free1; + } pdata->codec_device = platform_device_alloc("wm9712-codec", -1); - if (!pdata->codec_device) + if (!pdata->codec_device) { dev_err(&op->dev, "platform_device_alloc() failed\n"); + ret = -ENOMEM; + goto out_free1; + } ret = platform_device_add(pdata->codec_device); - if (ret) + if (ret) { dev_err(&op->dev, "platform_device_add() failed: %d\n", ret); + goto out_free2; + } ret = snd_soc_register_card(card); - if (ret) + if (ret) { dev_err(&op->dev, "snd_soc_register_card() failed: %d\n", ret); + goto out_free2; + } platform_set_drvdata(op, pdata); +out_free2: + platform_device_put(pdata->codec_device); +out_free1: + devm_kfree(&op->dev, pdata); return ret; } ---

2 1

[alsa-devel] [PATCH 0/4] ALSA: firewire-lib: bug fix and code cleanup for packet streaming engine
by Takashi Sakamoto 24 May '19

24 May '19

Hi, This patchset is the last part of refactoring for ALSA IEC 61883-1/6 packet streaming engine. It includes two fixes and two refactoring. Takashi Sakamoto (4): ALSA: firewire-lib: fix data block counter for incoming packet without CIP header ALSA: firewire-lib: fix inverted node IDs for amdtp_packet events ALSA: firewire-lib: refactoring to obsolete IT packet handler ALSA: firewire-lib: refactoring to obsolete IR packet handler sound/firewire/amdtp-stream-trace.h | 6 +- sound/firewire/amdtp-stream.c | 138 ++++++++++++++-------------- 2 files changed, 74 insertions(+), 70 deletions(-) -- 2.20.1

2 5

[alsa-devel] [PATCH 0/2] add pcm_file support in MMAP read access mode
by Adam Miartus 24 May '19

24 May '19

adds support for overwritting input buffer in MMAP access mode for in pcm_file plugin new api pcm_fast_ops callback mmap_begin is introduced to ease the operation needed to overwrite mmaped buffer. this allows to modify only the next continuous part of data in ring buffer. Adam Miartus (2): pcm: add mmap_begin callback to snd_pcm_fast_ops_t api pcm_file: add infile read support for mmap mode src/pcm/pcm.c | 6 ++++++ src/pcm/pcm_file.c | 31 +++++++++++++++++++++++++++++++ src/pcm/pcm_local.h | 1 + 3 files changed, 38 insertions(+) -- 2.7.4

2 3

[alsa-devel] [PATCH 0/4] ALSA: firewire-lib: unify handlers for outgoing packet
by Takashi Sakamoto 24 May '19

24 May '19

Hi, In IT context of Linux firewire subsystem, some quadlets of packet payload can be put as a part of packet header. This enables to use context payload for data blocks only. This patchset uses the mechanism to unify handlers of outgoing packet for with-CIP and without-CIP headers. Takashi Sakamoto (4): ALSA: firewire-lib: split helper function to generate CIP header ALSA: firewire-lib: unify packet handler for IT context ALSA: firewire-lib: code refactoring to queueing packets ALSA: firewire-lib: use 8 byte packet header for IT context to separate CIP header from CIP payload sound/firewire/amdtp-stream.c | 150 ++++++++++++++++------------------ sound/firewire/amdtp-stream.h | 3 - 2 files changed, 69 insertions(+), 84 deletions(-) -- 2.20.1

2 7

[alsa-devel] [PATCH 0/6] ALSA: firewire-lib: unify handlers for incoming packet
by Takashi Sakamoto 24 May '19

24 May '19

Hi, In IR context of Linux FireWire subsystem, some quadlets of packet payload can be handled as a part of context header. As a result context payload can just include the rest of packet payload. This patchset uses the mechanism to unify handlers of incoming packet for with-CIP and without-CIP headers. Takashi Sakamoto (6): ALSA: firewire-lib: use clear name for variable of CIP header ALSA: firewire-lib: calculate the length of packet payload in packet handler ALSA: firewire-lib: compute pointer to payload buffer in context handler ALSA: firewire-lib: split helper function to check incoming CIP header ALSA: firewire-lib: use 16 bytes IR context header to separate CIP header ALSA: firewire-lib: unify packet handler for IR context sound/firewire/amdtp-stream.c | 180 ++++++++++++++++++---------------- sound/firewire/amdtp-stream.h | 8 +- 2 files changed, 97 insertions(+), 91 deletions(-) -- 2.20.1

2 10

[alsa-devel] [PATCH] Fix single speed ADAT capture and playback with RME HDSPe AIO
by Philippe Bekaert 24 May '19

24 May '19

By taking into account the mapping from logical to DMA channels when enabling or disabling audio channels, ADAT channels 3 to 8 on the RME HDSPe AIO card are now correctly captured and played back in single speed mode. Since the mapping is an identity mapping for all cards except AIO and RayDAT, only those cards should be affected by this patch. It was tested on an AIO card. The patch needs testing on other cards, in particular RayDAT. Note: this patch does not solve ADAT capture and playback issues in double or triple speed mode. That seems to be another problem. Signed-off-by: Philippe Bekaert <Philippe.Bekaert(a)panokkel.be> --- sound/pci/rme9652/hdspm.c | 61 +++++++++++++++++++++++---------------- 1 file changed, 36 insertions(+), 25 deletions(-) diff --git a/sound/pci/rme9652/hdspm.c b/sound/pci/rme9652/hdspm.c index 1209cf0b05e0..982b297b3d0a 100644 --- a/sound/pci/rme9652/hdspm.c +++ b/sound/pci/rme9652/hdspm.c @@ -23,6 +23,9 @@ * * Modified 2011-01-25 variable period sizes on RayDAT/AIO by Adrian Knoth * + * Modified 2019-05-23 fix AIO single speed ADAT capture and playback + * by Philippe.Bekaert(a)uhasselt.be + * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or @@ -1105,9 +1108,9 @@ static int hdspm_autosync_ref(struct hdspm *hdspm); static int hdspm_set_toggle_setting(struct hdspm *hdspm, u32 regmask, int out); static int snd_hdspm_set_defaults(struct hdspm *hdspm); static int hdspm_system_clock_mode(struct hdspm *hdspm); -static void hdspm_set_sgbuf(struct hdspm *hdspm, - struct snd_pcm_substream *substream, - unsigned int reg, int channels); +static void hdspm_set_channel_dma_addr(struct hdspm *hdspm, + struct snd_pcm_substream *substream, + unsigned int reg, int channels); static int hdspm_aes_sync_check(struct hdspm *hdspm, int idx); static int hdspm_wc_sync_check(struct hdspm *hdspm); @@ -5588,11 +5591,16 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream, if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - hdspm_set_sgbuf(hdspm, substream, HDSPM_pageAddressBufferOut, - params_channels(params)); + for (i = 0; i < params_channels(params); ++i) { + int c = hdspm->channel_map_out[i]; - for (i = 0; i < params_channels(params); ++i) - snd_hdspm_enable_out(hdspm, i, 1); + if (c < 0) + continue; /* just make sure */ + hdspm_set_channel_dma_addr(hdspm, substream, + HDSPM_pageAddressBufferOut, + c); + snd_hdspm_enable_out(hdspm, c, 1); + } hdspm->playback_buffer = (unsigned char *) substream->runtime->dma_area; @@ -5600,11 +5608,16 @@ static int snd_hdspm_hw_params(struct snd_pcm_substream *substream, "Allocated sample buffer for playback at %p\n", hdspm->playback_buffer); } else { - hdspm_set_sgbuf(hdspm, substream, HDSPM_pageAddressBufferIn, - params_channels(params)); - - for (i = 0; i < params_channels(params); ++i) - snd_hdspm_enable_in(hdspm, i, 1); + for (i = 0; i < params_channels(params); ++i) { + int c = hdspm->channel_map_in[i]; + + if (c < 0) + continue; + hdspm_set_channel_dma_addr(hdspm, substream, + HDSPM_pageAddressBufferIn, + c); + snd_hdspm_enable_in(hdspm, c, 1); + } hdspm->capture_buffer = (unsigned char *) substream->runtime->dma_area; @@ -5665,19 +5678,17 @@ static int snd_hdspm_hw_free(struct snd_pcm_substream *substream) struct hdspm *hdspm = snd_pcm_substream_chip(substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - - /* params_channels(params) should be enough, - but to get sure in case of error */ - for (i = 0; i < hdspm->max_channels_out; ++i) + /* Just disable all channels. The saving when disabling a */ + /* smaller set is not worth the trouble. */ + for (i = 0; i < HDSPM_MAX_CHANNELS; ++i) snd_hdspm_enable_out(hdspm, i, 0); hdspm->playback_buffer = NULL; } else { - for (i = 0; i < hdspm->max_channels_in; ++i) + for (i = 0; i < HDSPM_MAX_CHANNELS; ++i) snd_hdspm_enable_in(hdspm, i, 0); hdspm->capture_buffer = NULL; - } snd_pcm_lib_free_pages(substream); @@ -6416,17 +6427,17 @@ static int snd_hdspm_preallocate_memory(struct hdspm *hdspm) return 0; } - -static void hdspm_set_sgbuf(struct hdspm *hdspm, - struct snd_pcm_substream *substream, - unsigned int reg, int channels) +/* Inform the card what DMA addresses to use for the indicated channel. */ +/* Each channel got 16 4K pages allocated for DMA transfers. */ +static void hdspm_set_channel_dma_addr(struct hdspm *hdspm, + struct snd_pcm_substream *substream, + unsigned int reg, int channel) { int i; - /* continuous memory segment */ - for (i = 0; i < (channels * 16); i++) + for (i = channel * 16; i < channel * 16 + 16; i++) hdspm_write(hdspm, reg + 4 * i, - snd_pcm_sgbuf_get_addr(substream, 4096 * i)); + snd_pcm_sgbuf_get_addr(substream, 4096 * i)); } -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: Use struct_size() helper
by Gustavo A. R. Silva 24 May '19

24 May '19

Make use of the struct_size() helper instead of an open-coded version in order to avoid any potential type mistakes, in particular in the context in which this code is being used. So, replace the following form: sizeof(struct rate_priv) + src_format->channels * sizeof(struct rate_channel) with: struct_size(data, channels, src_format->channels) This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/core/oss/rate.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/oss/rate.c b/sound/core/oss/rate.c index 2fa9299a440d..7cd09cef6961 100644 --- a/sound/core/oss/rate.c +++ b/sound/core/oss/rate.c @@ -323,8 +323,8 @@ int snd_pcm_plugin_build_rate(struct snd_pcm_substream *plug, err = snd_pcm_plugin_build(plug, "rate conversion", src_format, dst_format, - sizeof(struct rate_priv) + - src_format->channels * sizeof(struct rate_channel), + struct_size(data, channels, + src_format->channels), &plugin); if (err < 0) return err; -- 2.21.0

2 1

Re: [alsa-devel] [PATCH] ASoC: fsl_esai: fix the channel swap issue after xrun
by S.j. Wang 24 May '19

24 May '19

Hi > On Thu, May 23, 2019 at 09:53:42AM +0000, S.j. Wang wrote: > > > > + /* > > > > + * Add fifo reset here, because the regcache_sync will > > > > + * write one more data to ETDR. > > > > + * Which will cause channel shift. > > > > > > Sounds like a bug to me...should fix it first by marking the data > > > registers as volatile. > > > > > The ETDR is a writable register, it is not volatile. Even we change it > > to Volatile, I don't think we can't avoid this issue. for the > > regcache_sync Just to write this register, it is correct behavior. > > Is that so? Quoting the comments of regcache_sync(): > "* regcache_sync - Sync the register cache with the hardware. > * > * @map: map to configure. > * > * Any registers that should not be synced should be marked as > * volatile." > > If regcache_sync() does sync volatile registers too as you said, I don't mind > having this FIFO reset WAR for now, though I think this mismatch between > the comments and the actual behavior then should get people's attention. > > Thank you ETDR is not volatile, if we mark it is volatile, is it correct? Bets regards Wang shengjiu

2 1

[alsa-devel] [PATCH] ASoC: Intel: Skylake: Use struct_size() helper
by Gustavo A. R. Silva 24 May '19

24 May '19

Make use of the struct_size() helper instead of an open-coded version in order to avoid any potential type mistakes, in particular in the context in which this code is being used. So, replace the following form: uuid_params->num_modules * sizeof(struct skl_mod_inst_map) + sizeof(uuid_params->num_modules) with: struct_size(params, u.map, uuid_params->num_modules) and so on... This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/intel/skylake/skl-topology.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 389f1862bc43..c69d999d7bf1 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -941,9 +941,7 @@ static int skl_tplg_find_moduleid_from_uuid(struct skl *skl, if (bc->set_params == SKL_PARAM_BIND && bc->max) { uuid_params = (struct skl_kpb_params *)bc->params; - size = uuid_params->num_modules * - sizeof(struct skl_mod_inst_map) + - sizeof(uuid_params->num_modules); + size = struct_size(params, u.map, uuid_params->num_modules); params = devm_kzalloc(bus->dev, size, GFP_KERNEL); if (!params) @@ -3315,7 +3313,7 @@ static int skl_tplg_get_int_tkn(struct device *dev, struct snd_soc_tplg_vendor_value_elem *tkn_elem, struct skl *skl) { - int tkn_count = 0, ret, size; + int tkn_count = 0, ret; static int mod_idx, res_val_idx, intf_val_idx, dir, pin_idx; struct skl_module_res *res = NULL; struct skl_module_iface *fmt = NULL; @@ -3323,6 +3321,7 @@ static int skl_tplg_get_int_tkn(struct device *dev, static struct skl_astate_param *astate_table; static int astate_cfg_idx, count; int i; + size_t size; if (skl->modules) { mod = skl->modules[mod_idx]; @@ -3366,8 +3365,8 @@ static int skl_tplg_get_int_tkn(struct device *dev, return -EINVAL; } - size = tkn_elem->value * sizeof(struct skl_astate_param) + - sizeof(count); + size = struct_size(skl->cfg.astate_cfg, astate_table, + tkn_elem->value); skl->cfg.astate_cfg = devm_kzalloc(dev, size, GFP_KERNEL); if (!skl->cfg.astate_cfg) return -ENOMEM; -- 2.21.0

1 0

[alsa-devel] [PATCH] ASoC: dapm: Use struct_size() in krealloc()
by Gustavo A. R. Silva 23 May '19

23 May '19

One of the more common cases of allocation size calculations is finding the size of a structure that has a zero-sized array at the end, along with memory for some number of elements for that array. For example: struct foo { int stuff; struct boo entry[]; }; instance = krealloc(instance, sizeof(struct foo) + count * sizeof(struct boo), GFP_KERNEL); Instead of leaving these open-coded and prone to type mistakes, use the new struct_size() helper: instance = krealloc(instance, struct_size(instance, entry, count), GFP_KERNEL); This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/soc-dapm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b71ddaca70e5..511ed84b147b 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -487,7 +487,8 @@ static int dapm_kcontrol_add_widget(struct snd_kcontrol *kcontrol, n = 1; new_wlist = krealloc(data->wlist, - sizeof(*new_wlist) + sizeof(widget) * n, GFP_KERNEL); + struct_size(new_wlist, widgets, n), + GFP_KERNEL); if (!new_wlist) return -ENOMEM; -- 2.21.0

1 0

[alsa-devel] No sound on Asrock Z68 Extreme4
by Marcin Juszkiewicz 23 May '19

23 May '19

I recently moved my home machine to Asrock Z68 Extreme4 mainboard. Booted into Fedora 30 and found out that my speakers are not making any sound. alsa-info.sh output: http://alsa-project.org/db/?f=9c079d91b5f2278d3c3bd98518a6a717c86d31df Pulseaudio says that are no speakers connected and 'hdajacksensetest' says the same: 21:08 (0s) hrw@puchatek:~$ sudo hdajacksensetest Pin 0x14 (Green Line Out, Rear side): present = No Pin 0x15 (Black Line Out, Rear side): present = No Pin 0x16 (Orange Line Out, Rear side): present = No Pin 0x18 (Pink Mic, Rear side): present = No Pin 0x1a (Blue Line In, Rear side): present = No I noticed 'hdajackretask' tool but no idea how to use it properly. Any ideas?

1 0

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Fix UAF decrement if card has no live interfaces in card.c
by Suren Baghdasaryan 23 May '19

23 May '19

Commit 5f8cf712582617d523120df67d392059eaf2fc4b upstream. This is a backport to stable 3.18.y. Implementation in 3.18 differs using chip->probing flag instead of chip->active atomic but it still has the UAF issue. If a USB sound card reports 0 interfaces, an error condition is triggered and the function usb_audio_probe errors out. In the error path, there was a use-after-free vulnerability where the memory object of the card was first freed, followed by a decrement of the number of active chips. Moving the decrement above the atomic_dec fixes the UAF. [ The original problem was introduced in 3.1 kernel, while it was developed in a different form. The Fixes tag below indicates the original commit but it doesn't mean that the patch is applicable cleanly. -- tiwai ] Fixes: 362e4e49abe5 ("ALSA: usb-audio - clear chip->probing on error exit") Reported-by: Hui Peng <benquike(a)gmail.com> Reported-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Signed-off-by: Hui Peng <benquike(a)gmail.com> Signed-off-by: Mathias Payer <mathias.payer(a)nebelwelt.net> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> [surenb(a)google.com: resolve 3.18 differences] Signed-off-by: Suren Baghdasaryan <surenb(a)google.com> --- Analysis for 3.18 codebase: snd_usb_audio_create() sets card->device_data = chip snd_usb_audio_probe() calls snd_card_free() and then resets chip->probing snd_card_free() results in the following call chain: snd_card_free_when_closed() which waits on release_completion snd_card_do_free() calls snd_device_free_all() and signals release_completion snd_card_do_free() calls __snd_device_free() __snd_device_free() calls dev->ops->dev_free() == snd_usb_audio_dev_free() snd_usb_audio_dev_free() calls snd_usb_audio_free(chip) and frees "chip" chip->probing = 0 results in UAF sound/usb/card.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/usb/card.c b/sound/usb/card.c index f7dbdc10bf77..59fb1ef3cd55 100644 --- a/sound/usb/card.c +++ b/sound/usb/card.c @@ -593,9 +593,12 @@ snd_usb_audio_probe(struct usb_device *dev, __error: if (chip) { + /* chip->probing is inside the chip->card object, + * reset before memory is possibly returned. + */ + chip->probing = 0; if (!chip->num_interfaces) snd_card_free(chip->card); - chip->probing = 0; } mutex_unlock(&register_mutex); __err_val: -- 2.21.0.1020.gf2820cf01a-goog

2 1

[alsa-devel] [PATCH] ASoC: core: lock client_mutex while removing link components
by Ranjani Sridharan 23 May '19

23 May '19

Removing link components results in topology unloading. So, acquire the client_mutex before removing components in soc_remove_link_components. This will prevent the lockdep warning seen when dai links are removed during topology removal. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ce8c057bcd5b..bbb52fbd195e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1010,12 +1010,14 @@ static void soc_remove_link_components(struct snd_soc_card *card, struct snd_soc_component *component; struct snd_soc_rtdcom_list *rtdcom; + mutex_lock(&client_mutex); for_each_rtdcom(rtd, rtdcom) { component = rtdcom->component; if (component->driver->remove_order == order) soc_remove_component(component); } + mutex_unlock(&client_mutex); } static void soc_remove_dai_links(struct snd_soc_card *card) -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: qdsp6: q6core: Use struct_size() in kmemdup()
by Gustavo A. R. Silva 23 May '19

23 May '19

Make use of the struct_size() helper instead of an open-coded version in order to avoid any potential type mistakes, in particular in the context in which this code is being used. So, replace code of the following form: sizeof(*fwk) + fwk->num_services * sizeof(fwk->svc_api_info[0]); with: struct_size(fwk, svc_api_info, fwk->num_services) and so on... Notice that variables bytes and len are unnecessary, hence they are removed. This code was detected with the help of Coccinelle. Signed-off-by: Gustavo A. R. Silva <gustavo(a)embeddedor.com> --- sound/soc/qcom/qdsp6/q6core.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/qcom/qdsp6/q6core.c b/sound/soc/qcom/qdsp6/q6core.c index cdfc8ab6cfc0..ae314a652efe 100644 --- a/sound/soc/qcom/qdsp6/q6core.c +++ b/sound/soc/qcom/qdsp6/q6core.c @@ -98,13 +98,13 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) } case AVCS_CMDRSP_GET_FWK_VERSION: { struct avcs_cmdrsp_get_fwk_version *fwk; - int bytes; fwk = data->payload; - bytes = sizeof(*fwk) + fwk->num_services * - sizeof(fwk->svc_api_info[0]); - core->fwk_version = kmemdup(data->payload, bytes, GFP_ATOMIC); + core->fwk_version = kmemdup(data->payload, + struct_size(fwk, svc_api_info, + fwk->num_services), + GFP_ATOMIC); if (!core->fwk_version) return -ENOMEM; @@ -115,13 +115,13 @@ static int q6core_callback(struct apr_device *adev, struct apr_resp_pkt *data) } case AVCS_GET_VERSIONS_RSP: { struct avcs_cmdrsp_get_version *v; - int len; v = data->payload; - len = sizeof(*v) + v->num_services * sizeof(v->svc_api_info[0]); - - core->svc_version = kmemdup(data->payload, len, GFP_ATOMIC); + core->svc_version = kmemdup(data->payload, + struct_size(v, svc_api_info, + v->num_services), + GFP_ATOMIC); if (!core->svc_version) return -ENOMEM; -- 2.21.0

1 0

[alsa-devel] [PATCH 1/5] ASoC: madera: Add DT bindings for Cirrus Logic Madera codecs
by Charles Keepax 23 May '19

23 May '19

From: Richard Fitzgerald <rf(a)opensource.cirrus.com> The Cirrus Logic Madera codecs are a family of related codecs with extensive digital and analogue I/O, digital mixing and routing, signal processing and programmable DSPs. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- Documentation/devicetree/bindings/sound/madera.txt | 67 ++++++++++++++++++++++ MAINTAINERS | 1 + include/dt-bindings/sound/madera.h | 29 ++++++++++ 3 files changed, 97 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/madera.txt create mode 100644 include/dt-bindings/sound/madera.h diff --git a/Documentation/devicetree/bindings/sound/madera.txt b/Documentation/devicetree/bindings/sound/madera.txt new file mode 100644 index 0000000000000..1114fcf1aa4c2 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/madera.txt @@ -0,0 +1,67 @@ +Cirrus Logic Madera class audio codecs + +This describes audio configuration bindings for these codecs. + +See also the core bindings for the parent MFD driver: +See Documentation/devicetree/bindings/mfd/madera.txt + +and defines for values used in these bindings: +include/dt-bindings/sound/madera.h + +These properties are all contained in the parent MFD node. + +Optional properties: + - cirrus,dmic-ref : Indicates how the MICBIAS pins have been externally + connected to DMICs on each input, one cell per input. + <IN1 IN2 IN3 ...> + A value of 0 indicates MICVDD and is the default, other values depend on the + codec: + For CS47L35 one of the CS47L35_DMIC_REF_xxx values + For all other codecs one of the MADERA_DMIC_REF_xxx values + Also see the datasheet for a description of the INn_DMIC_SUP field. + + - cirrus,inmode : A list of input mode settings for each input. A maximum of + 16 cells, with four cells per input in the order INnAL, INnAR INnBL INnBR. + For non-muxed inputs the first two cells for that input set the mode for + the left and right channel and the second two cells must be 0. + For muxed inputs the first two cells for that input set the mode of the + left and right A inputs and the second two cells set the mode of the left + and right B inputs. + Valid mode values are one of the MADERA_INMODE_xxx. If the array is shorter + than the number of inputs the unspecified inputs default to + MADERA_INMODE_DIFF. + + - cirrus,out-mono : Mono bit for each output, must contain six cells if + specified. A non-zero value indicates the corresponding output is mono. + + - cirrus,max-channels-clocked : Maximum number of channels that I2S clocks + will be generated for. Useful when clock master for systems where the I2S + bus has multiple data lines. + One cell for each AIF, use a value of zero for AIFs that should be handled + normally. + + - cirrus,pdm-fmt : PDM speaker data format, must contain 2 cells + (OUT5 and OUT6). See the PDM_SPKn_FMT field in the datasheet for a + description of this value. + The second cell is ignored for codecs that do not have OUT6. + + - cirrus,pdm-mute : PDM mute format, must contain 2 cells + (OUT5 and OUT6). See the PDM_SPKn_CTRL_1 register in the datasheet for a + description of this value. + The second cell is ignored for codecs that do not have OUT6. + +Example: + +cs47l35@0 { + compatible = "cirrus,cs47l35"; + + cirrus,dmic-ref = <0 0 CS47L35_DMIC_REF_MICBIAS1B 0>; + cirrus,inmode = < + MADERA_INMODE_DMIC MADERA_INMODE_DMIC /* IN1A digital */ + MADERA_INMODE_SE MADERA_INMODE_SE /* IN1B single-ended */ + MADERA_INMODE_DIFF MADERA_INMODE_DIFF /* IN2 differential */ + 0 0 /* not used on this codec */ + >; + cirrus,out-mono = <0 0 0 0 0 0>; + cirrus,max-channels-clocked = <2 0 0>; +}; diff --git a/MAINTAINERS b/MAINTAINERS index 5cfbea4ce5750..642cb5610dd50 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -3915,6 +3915,7 @@ W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: Documentation/devicetree/bindings/mfd/madera.txt F: Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt +F: include/dt-bindings/sound/madera* F: include/linux/irqchip/irq-madera* F: include/linux/mfd/madera/* F: drivers/gpio/gpio-madera* diff --git a/include/dt-bindings/sound/madera.h b/include/dt-bindings/sound/madera.h new file mode 100644 index 0000000000000..9ff4eae5259b0 --- /dev/null +++ b/include/dt-bindings/sound/madera.h @@ -0,0 +1,29 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * Device Tree defines for Madera codecs + * + * Copyright (C) 2016-2017 Cirrus Logic, Inc. and + * Cirrus Logic International Semiconductor Ltd. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef DT_BINDINGS_SOUND_MADERA_H +#define DT_BINDINGS_SOUND_MADERA_H + +#define MADERA_INMODE_DIFF 0 +#define MADERA_INMODE_SE 1 +#define MADERA_INMODE_DMIC 2 + +#define MADERA_DMIC_REF_MICVDD 0 +#define MADERA_DMIC_REF_MICBIAS1 1 +#define MADERA_DMIC_REF_MICBIAS2 2 +#define MADERA_DMIC_REF_MICBIAS3 3 + +#define CS47L35_DMIC_REF_MICBIAS1B 1 +#define CS47L35_DMIC_REF_MICBIAS2A 2 +#define CS47L35_DMIC_REF_MICBIAS2B 3 + +#endif -- 2.11.0

1 2

[alsa-devel] [PATCH v6 0/2] ASoC: CX2072X codec support (revised, v6)
by Takashi Iwai 23 May '19

23 May '19

Hi, the merge window was closed, and now it's time for fun again. This is a v6 revision of ASoC CX2072X codec support patchset for Intel platforms. Takashi === v5->v6: Drop buggy&untested EQ and DRC codes Lots of code reduction/cleanup in codec driver Add more comments about platform-specific stuff v4->v5: Move clk management to runtime PM Sparse warning fixes Some more code simplification Drop tricky regcache fiddling Apply mutex locks around possible racy sequences Move exported jack detection stuff into set_jack callback v3->v4: Fix the wrong endianess conversion in reg write Minor code cleanups v2->v3: CX2072X codec driver: * Move register tables to appropriate place * Remove some confusing codes * Set snd_ctl_boolean_* helpers directly * Fix EQ put callback * Rename to "DAC1 Switch" from "DAC1 Mute Switch" * Drop superfluous regmap calls at shutdown * Avoid regmap_register_patch() * Add missing register definitions * Fix register access on big-endian machine * Remove regcache messes CX2072X machine driver: * Add Pierre's ack * Drop superfluous ssp0 routes v1->v2: Uncomment SOF entries in ACPI binding Move snd_soc_dai_set_bclk_ratio() call into init callback The original cover letter is below: === It's been long time ago the first version of CX2072X codec patch was submitted, and I hoped that it'd be resubmitted, but never happened, as the original author seems to have left, unfortunately. Since I have such a laptop (ASUS E200HA), I took some time to cook and brush up the patches. The patches are for 5.2, should be applicable to ASoC tree for-next branch. Will post the corresponding UCM profile patch once when this gets accepted. thanks, Takashi === Simon Ho (1): ASoC: Add support for Conexant CX2072X CODEC Takashi Iwai (1): ASoC: Intel: Add machine driver for CX2072X on BYT/CHT platforms sound/soc/codecs/Kconfig | 7 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cx2072x.c | 1727 +++++++++++++++++++++ sound/soc/codecs/cx2072x.h | 314 ++++ sound/soc/intel/boards/Kconfig | 11 + sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/bytcht_cx2072x.c | 262 ++++ sound/soc/intel/common/soc-acpi-intel-byt-match.c | 8 + sound/soc/intel/common/soc-acpi-intel-cht-match.c | 8 + 9 files changed, 2341 insertions(+) create mode 100644 sound/soc/codecs/cx2072x.c create mode 100644 sound/soc/codecs/cx2072x.h create mode 100644 sound/soc/intel/boards/bytcht_cx2072x.c -- 2.16.4

2 5

[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: enable dev runtime suspend and resume
by Jiaxin Yu 23 May '19

23 May '19

System suspend will power off audio, so we need regcache sync when system resume. Remove pm_runtime_get_sync in dev probe and pm_runtime_put_sync in dev remove.So that audio dev can triggle runitme suspend and resume. Signed-off-by: Jiaxin Yu <jiaxin.yu(a)mediatek.com> --- Hi, This patch is based on for-next and test passed. --- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 5 +---- 1 file changed, 1 insertion(+), 4 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c index 56c3732..4a31106 100644 --- a/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c +++ b/sound/soc/mediatek/mt8183/mt8183-afe-pcm.c @@ -1142,8 +1142,6 @@ static int mt8183_afe_pcm_dev_probe(struct platform_device *pdev) regcache_cache_only(afe->regmap, true); regcache_mark_dirty(afe->regmap); - pm_runtime_get_sync(&pdev->dev); - /* init memif */ afe->memif_size = MT8183_MEMIF_NUM; afe->memif = devm_kcalloc(dev, afe->memif_size, sizeof(*afe->memif), @@ -1235,11 +1233,10 @@ static int mt8183_afe_pcm_dev_probe(struct platform_device *pdev) static int mt8183_afe_pcm_dev_remove(struct platform_device *pdev) { - pm_runtime_put_sync(&pdev->dev); - pm_runtime_disable(&pdev->dev); if (!pm_runtime_status_suspended(&pdev->dev)) mt8183_afe_runtime_suspend(&pdev->dev); + return 0; } -- 1.8.1.1.dirty

2 2

[alsa-devel] [PATCH] ASoC: mediatek: mt8183: move card registration to the end of probe
by Tzung-Bi Shih 23 May '19

23 May '19

mt8183_mt6358_ts3a227_max98357_dev_probe() does not return immediately if devm_snd_soc_register_card() returns fail. As a result, the rest of code still get executed. Two options to fix: 1. add an extra return statement if devm_snd_soc_register_card() returns fail 2. move devm_snd_soc_register_card() to the end of probe Adopt option 2 which is more clear and common. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c | 9 ++------- 1 file changed, 2 insertions(+), 7 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 4e44e5689d6f..1efc7f5d04e4 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -345,7 +345,7 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) struct device_node *platform_node; struct snd_soc_dai_link *dai_link; struct pinctrl *default_pins; - int ret, i; + int i; card->dev = &pdev->dev; @@ -380,11 +380,6 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) return -EINVAL; } - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", - __func__, ret); - default_pins = devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); if (IS_ERR(default_pins)) { @@ -393,7 +388,7 @@ mt8183_mt6358_ts3a227_max98357_dev_probe(struct platform_device *pdev) return PTR_ERR(default_pins); } - return ret; + return devm_snd_soc_register_card(&pdev->dev, card); } #ifdef CONFIG_OF -- 2.21.0.1020.gf2820cf01a-goog

2 1

[alsa-devel] [PATCH] ASoC: mediatek: mt8183: remove unused DAPM pins
by Tzung-Bi Shih 23 May '19

23 May '19

Kernel emits the following messages when booting. ASoC: DAPM unknown pin Headset Mic ASoC: DAPM unknown pin Headphone Remove the unused DAPM pins. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- .../soc/mediatek/mt8183/mt8183-da7219-max98357.c | 15 +-------------- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 16 +--------------- 2 files changed, 2 insertions(+), 29 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c index 31ea8632c397..231fb49d79b9 100644 --- a/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-da7219-max98357.c @@ -18,18 +18,6 @@ static struct snd_soc_jack headset_jack; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin headset_jack_pins[] = { - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, -}; - static struct snd_soc_dai_link_component mt8183_da7219_max98357_external_codecs[] = { { @@ -375,8 +363,7 @@ mt8183_da7219_max98357_headset_init(struct snd_soc_component *component) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &headset_jack, - headset_jack_pins, - ARRAY_SIZE(headset_jack_pins)); + NULL, 0); if (ret) return ret; diff --git a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c index 4e44e5689d6f..1ba7aa58e4b6 100644 --- a/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c +++ b/sound/soc/mediatek/mt8183/mt8183-mt6358-ts3a227-max98357.c @@ -17,19 +17,6 @@ static struct snd_soc_jack headset_jack; -/* Headset jack detection DAPM pins */ -static struct snd_soc_jack_pin headset_jack_pins[] = { - { - .pin = "Headphone", - .mask = SND_JACK_HEADPHONE, - }, - { - .pin = "Headset Mic", - .mask = SND_JACK_MICROPHONE, - }, - -}; - static int mt8183_mt6358_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -328,8 +315,7 @@ mt8183_mt6358_ts3a227_max98357_headset_init(struct snd_soc_component *component) SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3, &headset_jack, - headset_jack_pins, - ARRAY_SIZE(headset_jack_pins)); + NULL, 0); if (ret) return ret; -- 2.21.0.1020.gf2820cf01a-goog

2 1

[alsa-devel] [RFC PATCH] ASoC: simple-card: Fix configuration of DAI format
by Jon Hunter 23 May '19

23 May '19

When configuring a codec to be both bit-clock and frame-master, it was found that the codec was always configured as bit-clock and frame-slave. Looking at the simple_dai_link_of() function there appears to be two problems with the configuration of the DAI format, which are ... 1. The function asoc_simple_parse_daifmt() is called before the function asoc_simple_parse_codec() and this means that the device-tree node for the codec has not been parsed yet, which is needed by the function asoc_simple_parse_daifmt() to determine who is the codec. 2. The phandle passed to asoc_simple_parse_daifmt() is the phandle to the 'codec' node and not the phandle of the actual codec defined by the 'sound-dai' property under the 'codec' node. Fix the above by moving the call to asoc_simple_parse_daifmt() after the the call to asoc_simple_parse_codec() and pass the phandle for the codec to asoc_simple_parse_daifmt(). Signed-off-by: Jon Hunter <jonathanh(a)nvidia.com> --- sound/soc/generic/simple-card.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9b568f578bcd..c2c8dcbcf795 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -283,11 +283,6 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, codec_dai = dai_props->codec_dai = &priv->dais[li->dais++]; - ret = asoc_simple_parse_daifmt(dev, node, codec, - prefix, &dai_link->dai_fmt); - if (ret < 0) - goto dai_link_of_err; - simple_parse_mclk_fs(top, cpu, codec, dai_props, prefix); ret = asoc_simple_parse_cpu(cpu, dai_link, &single_cpu); @@ -298,6 +293,11 @@ static int simple_dai_link_of(struct asoc_simple_priv *priv, if (ret < 0) goto dai_link_of_err; + ret = asoc_simple_parse_daifmt(dev, node, dai_link->codecs->of_node, + prefix, &dai_link->dai_fmt); + if (ret < 0) + goto dai_link_of_err; + ret = asoc_simple_parse_platform(plat, dai_link); if (ret < 0) goto dai_link_of_err; -- 2.7.4

2 3

[alsa-devel] [PATCH] ASoC: sound/soc/intel/boards: limit some drivers to X86 since headers are only in arch/x86/
by Randy Dunlap 23 May '19

23 May '19

From: Randy Dunlap <rdunlap(a)infradead.org> Several drivers in sound/soc/intel/boards/ #include header files that only exist in arch/x86/include/asm. This causes build errors, so make these drivers depend on X86. Fixes these build errors (on ia64): ../sound/soc/intel/boards/bxt_da7219_max98357a.c:19:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcr_rt5640.c:31:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcr_rt5651.c:33:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/cht_bsw_rt5645.c:29:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcht_es8316.c:33:10: fatal error: asm/cpu_device_id.h: No such file or directory #include <asm/cpu_device_id.h> ../sound/soc/intel/boards/bytcht_da7213.c:26:10: fatal error: asm/platform_sst_audio.h: No such file or directory #include <asm/platform_sst_audio.h> And more drivers determined by: > grep "include.*asm.cpu_device_id.h" *.c bxt_da7219_max98357a.c:#include <asm/cpu_device_id.h> bytcht_es8316.c:#include <asm/cpu_device_id.h> bytcr_rt5640.c:#include <asm/cpu_device_id.h> bytcr_rt5651.c:#include <asm/cpu_device_id.h> cht_bsw_rt5645.c:#include <asm/cpu_device_id.h> sof_rt5682.c:#include <asm/cpu_device_id.h> and > grep "include.*asm.platform_sst_audio.h" *.c bytcht_da7213.c:#include <asm/platform_sst_audio.h> bytcht_es8316.c:#include <asm/platform_sst_audio.h> Signed-off-by: Randy Dunlap <rdunlap(a)infradead.org> Cc: Mark Brown <broonie(a)kernel.org> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: Jie Yang <yang.jie(a)linux.intel.com> Cc: alsa-devel(a)alsa-project.org --- sound/soc/intel/boards/Kconfig | 6 ++++++ 1 file changed, 6 insertions(+) --- lnx-52-rc1.orig/sound/soc/intel/boards/Kconfig +++ lnx-52-rc1/sound/soc/intel/boards/Kconfig @@ -87,6 +87,7 @@ config SND_SOC_INTEL_BYTCR_RT5640_MACH tristate "Baytrail and Baytrail-CR with RT5640 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_RT5640 help @@ -99,6 +100,7 @@ config SND_SOC_INTEL_BYTCR_RT5651_MACH tristate "Baytrail and Baytrail-CR with RT5651 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_RT5651 help @@ -123,6 +125,7 @@ config SND_SOC_INTEL_CHT_BSW_RT5645_MACH tristate "Cherrytrail & Braswell with RT5645/5650 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_RT5645 help @@ -159,6 +162,7 @@ config SND_SOC_INTEL_BYT_CHT_DA7213_MACH tristate "Baytrail & Cherrytrail with DA7212/7213 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_DA7213 help @@ -171,6 +175,7 @@ config SND_SOC_INTEL_BYT_CHT_ES8316_MACH tristate "Baytrail & Cherrytrail with ES8316 codec" depends on I2C && ACPI depends on X86_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_ACPI select SND_SOC_ES8316 help @@ -249,6 +254,7 @@ config SND_SOC_INTEL_BXT_DA7219_MAX98357 tristate "Broxton with DA7219 and MAX98357A in I2S Mode" depends on I2C && ACPI depends on MFD_INTEL_LPSS || COMPILE_TEST + depends on X86 select SND_SOC_DA7219 select SND_SOC_MAX98357A select SND_SOC_DMIC

3 2

[alsa-devel] [PATCH] ALSA: Enable micmute LED for Huawei laptops
by Ayman Bagabas 23 May '19

23 May '19

Since this LED is found on all Huawei laptops, we can hook it to huawei-wmi platform driver to control it. Also, some renames have been made to use product name instead of common name to avoid confusions. Signed-off-by: Ayman Bagabas <ayman.bagabas(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f83f21d64dd4..ab6a856ec614 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5747,7 +5747,7 @@ enum { ALC298_FIXUP_TPT470_DOCK, ALC255_FIXUP_DUMMY_LINEOUT_VERB, ALC255_FIXUP_DELL_HEADSET_MIC, - ALC256_FIXUP_HUAWEI_MBXP_PINS, + ALC256_FIXUP_HUAWEI_MACH_WX9_PINS, ALC295_FIXUP_HP_X360, ALC221_FIXUP_HP_HEADSET_MIC, ALC285_FIXUP_LENOVO_HEADPHONE_NOISE, @@ -6038,7 +6038,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_HEADSET_MIC }, - [ALC256_FIXUP_HUAWEI_MBXP_PINS] = { + [ALC256_FIXUP_HUAWEI_MACH_WX9_PINS] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { {0x12, 0x90a60130}, @@ -7063,9 +7063,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x511f, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x3bf8, "Quanta FL1", ALC269_FIXUP_PCM_44K), SND_PCI_QUIRK(0x17aa, 0x9e54, "LENOVO NB", ALC269_FIXUP_LENOVO_EAPD), - SND_PCI_QUIRK(0x19e5, 0x3200, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), - SND_PCI_QUIRK(0x19e5, 0x3201, "Huawei MBX", ALC255_FIXUP_MIC_MUTE_LED), - SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MBXP", ALC256_FIXUP_HUAWEI_MBXP_PINS), + SND_PCI_QUIRK(0x19e5, 0x3204, "Huawei MACH-WX9", ALC256_FIXUP_HUAWEI_MACH_WX9_PINS), SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */ #if 0 @@ -7124,6 +7122,7 @@ static const struct snd_pci_quirk alc269_fixup_vendor_tbl[] = { SND_PCI_QUIRK_VENDOR(0x103c, "HP", ALC269_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK_VENDOR(0x104d, "Sony VAIO", ALC269_FIXUP_SONY_VAIO), SND_PCI_QUIRK_VENDOR(0x17aa, "Thinkpad", ALC269_FIXUP_THINKPAD_ACPI), + SND_PCI_QUIRK_VENDOR(0x19e5, "Huawei Matebook", ALC255_FIXUP_MIC_MUTE_LED), {} }; -- 2.20.1

2 1

Re: [alsa-devel] [PATCH] ASoC: fsl_esai: fix the channel swap issue after xrun
by S.j. Wang 23 May '19

23 May '19

Hi > > + /* > > + * Add fifo reset here, because the regcache_sync will > > + * write one more data to ETDR. > > + * Which will cause channel shift. > > Sounds like a bug to me...should fix it first by marking the data registers as > volatile. > The ETDR is a writable register, it is not volatile. Even we change it to Volatile, I don't think we can't avoid this issue. for the regcache_sync Just to write this register, it is correct behavior. Best regards Wang shengjiu

2 1

[alsa-devel] [PATCH - alsa-lib 0/1] pcm_file add missing unlock
by Adam Miartus 23 May '19

23 May '19

previous patches: pcm: file: add support for infile reading in non interleaved mode pcm: file: use snd_pcm_file_areas_read_infile for readi introduced an issue where pcm would not be unlocked on early return Adam Miartus (1): pcm_file: add missing unlock on early return src/pcm/pcm_file.c | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) -- 2.7.4

2 2

[alsa-devel] [PATCH] soundwire: stream: fix bad unlock balance
by Srinivas Kandagatla 23 May '19

23 May '19

This patch fixes below warning due to unlocking without locking. ===================================== WARNING: bad unlock balance detected! 5.1.0-16506-gc1c383a6f0a2-dirty #1523 Tainted: G W ------------------------------------- aplay/2954 is trying to release lock (&bus->msg_lock) at: do_bank_switch+0x21c/0x480 but there are no more locks to release! Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- drivers/soundwire/stream.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/stream.c b/drivers/soundwire/stream.c index 544925ff0b40..d16268f30e4f 100644 --- a/drivers/soundwire/stream.c +++ b/drivers/soundwire/stream.c @@ -814,7 +814,8 @@ static int do_bank_switch(struct sdw_stream_runtime *stream) goto error; } - mutex_unlock(&bus->msg_lock); + if (mutex_is_locked(&bus->msg_lock)) + utex_unlock(&bus->msg_lock); } return ret; -- 2.21.0

3 6

[alsa-devel] set power_save_node=0 as default
by Kailang 23 May '19

23 May '19

Hi Takashi, I measured power consumption between power_save_node=1 and power_save_node=0. It's almost the same. Codec will enter to runtime suspend and suspend. That pin also will enter to D3. Don't need to enter to D3 by single pin. So, Disable power_save_node as default. It will avoid more issues. Windows Driver also has not this option at runtime PM. Many Thanks. BR, Kailang

2 1

[alsa-devel] Question about soc_bind_aux_dev()
by Kuninori Morimoto 23 May '19

23 May '19

Hi ALSA ML Do you know why snd_soc_aux_dev has both name and codec_name ?? struct snd_soc_aux_dev { => const char *name; /* Codec name */ ... => const char *codec_name; ... }; For example, samsung's speyside.c has both name and codec_name static struct snd_soc_aux_dev speyside_aux_dev[] = { { => .name = "wm9081", => .codec_name = "wm9081.1-006c", ... }, }; In my research, there is no driver which is using .name only. All drivers are having codec_of_node, or .codec_name Now, in the soc_bind_aux_dev(), it checks .codec_name or codec_of_node first. This means, there is zero chance to use .name ? static int soc_bind_aux_dev(struct snd_soc_card *card, int num) { ... => if (aux_dev->codec_of_node || aux_dev->codec_name) { ... component = soc_find_component(...); ... => } else if (aux_dev->name) { ... component = soc_find_component(...); ... } ... } Why .name is needed ?? Thank you for your help !! Best regards --- Kuninori Morimoto

2 4

[alsa-devel] Add polling mode support on snd_hdac_bus_get_response(),
by Liao, Bard 23 May '19

23 May '19

Hi Takashi, We would like to add polling mode support on snd_hdac_bus_get_response(), and use it as a workaround for some specific platforms. Just like azx_rirb_get_response() and azx_create() if (chip->polling_mode || do_poll) snd_hdac_bus_update_rirb(bus); and if (IS_CFL(pci)) chip->polling_mode = 1; However struct azx is a local variable and other driver may have their own chip structure. Like struct skl and struct sof_intel_hda_dev. So I am thinking does it make sense to add the polling_mode in struct hdac_bus? So we can get the polling_mode directly from bus->polling_mode. Or it makes more sense to create a global chip struct similar to struct azx for snd_hdac_bus_get_response()? Thanks, Bard

2 1

[alsa-devel] [PATCH - alsa-lib 0/4] Add pcm_file input file support for MMAP
by Adam Miartus 22 May '19

22 May '19

implement overwriting read buffer with input file in case of RW_NONINTERLEAVED and MMAP access modes for alsa file plugin first two patches introduce a helper function for copying data from input file to readi and readn buffers for user. to facilitate this snd_pcm_areas_from_buf and snd_pcm_areas_from_bufs with snd_pcm_areas_copy is used. Previous readi behaviour in corner case of input file EOF is changed (more in patch comment). Patch 3,4 introduce new api callback mmap_begin to ease overwritting mmaped buffer in case of MMAP access. Adam Miartus (4): pcm_file: add support for infile reading in non interleaved mode pcm_file: use snd_pcm_file_areas_read_infile for readi pcm: add mmap_begin callback to snd_pcm_fast_ops_t api pcm_file: add infile read support for mmap mode src/pcm/pcm.c | 6 +++ src/pcm/pcm_dmix.c | 1 + src/pcm/pcm_dshare.c | 1 + src/pcm/pcm_dsnoop.c | 1 + src/pcm/pcm_file.c | 125 ++++++++++++++++++++++++++++++++++++++---------- src/pcm/pcm_generic.h | 1 - src/pcm/pcm_hooks.c | 1 + src/pcm/pcm_hw.c | 2 + src/pcm/pcm_ioplug.c | 1 + src/pcm/pcm_local.h | 1 + src/pcm/pcm_meter.c | 1 + src/pcm/pcm_mmap_emul.c | 1 + src/pcm/pcm_multi.c | 1 + src/pcm/pcm_null.c | 1 + src/pcm/pcm_plugin.c | 1 + src/pcm/pcm_rate.c | 1 + src/pcm/pcm_share.c | 1 + src/pcm/pcm_shm.c | 1 + 18 files changed, 122 insertions(+), 26 deletions(-) -- 2.7.4

2 1

[alsa-devel] Change Dell headset mode rule
by Kailang 22 May '19

22 May '19

Hi Takashi, In Dell headset mode, it will recheck during runtime resume when headset was plugged. This patch will move check headset type on unplug and system resume. Many Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH 0/4] ALSA: firewire-lib: use IT context header to compute cycle count
by Takashi Sakamoto 22 May '19

22 May '19

Hi, In Linux firewire subsystem, the series of timestamp for each packets is passed as argument to callback of IT context. The timestamp comes from descriptor of 1394 OHCI IT context and represents the cycle to process queued packet. In current implementation of ALSA IEC 61883-1/6 packet streaming engine, the timestamp comes from calculation with the last packet to generate the callback, thus it's not per-packet information. For simplicity, this patchset uses the header to compute the cycle for packet to be going to be queued. Additionally, this patchset adds some minor refactoring. Takashi Sakamoto (4): ALSA: firewire-lib: use union for directional parameters ALSA: firewire-lib: add helper function to cancel context inner callback handler ALSA: firewire-lib: obsolete macro for header of IT context ALSA: firewire-lib: use IT context header to compute cycle count for scheduling packet sound/firewire/amdtp-am824.c | 2 +- sound/firewire/amdtp-stream.c | 181 +++++++++++--------- sound/firewire/amdtp-stream.h | 35 ++-- sound/firewire/digi00x/amdtp-dot.c | 2 +- sound/firewire/fireworks/fireworks_stream.c | 4 +- sound/firewire/motu/amdtp-motu.c | 2 +- sound/firewire/tascam/amdtp-tascam.c | 2 +- 7 files changed, 135 insertions(+), 93 deletions(-) -- 2.20.1

2 5

[alsa-devel] [PATCH RESEND V6 0/3] Support more sample rate in asrc
by S.j. Wang 22 May '19

22 May '19

Support more sample rate in asrc Shengjiu Wang (3): ASoC: fsl_asrc: Fix the issue about unsupported rate ASoC: fsl_asrc: replace the process_option table with function ASoC: fsl_asrc: Unify the supported input and output rate Changes in RESEND V6 - change the Content-Transfer-Encoding to "quoted-printable", for - "base64" can't be applied Changes in v6 - add acked-by - fixed minor issue according to comments in v5 Changes in v5 - fix the [1/24, 8] - move fsl_asrc_sel_proc before setting Changes in v4 - add patch to Fix the [8kHz, 30kHz] open set issue. Changes in v3 - remove FSL_ASRC_RATES - refine fsl_asrc_sel_proc according to comments Changes in v2 - add more comments in code - add commit "Unify the supported input and output rate" sound/soc/fsl/fsl_asrc.c | 105 ++++++++++++++++++++++++++------------- 1 file changed, 71 insertions(+), 34 deletions(-) -- 2.21.0

2 6

[alsa-devel] [PATCH v2] ASoC: soc-pcm: fixup try_module_get()/module_put() timing
by Kuninori Morimoto 22 May '19

22 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_pcm_components_open/close() try to call try_module_get()/module_put() based on component->driver->module_get_upon_open. Here, the purpose why we need to call these functions are to checking module reference. Thus, we need to call try_module_open() even though it doesn't have .open callback. The same reason, we need to call module_put() even though it doesn't have .close This patch calls try_module_get()/module_put() regardless of .open/.close Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - merge both try_module_get()/module_put() patch into 1 sound/soc/soc-pcm.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7737e00..e24eab3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -440,10 +440,6 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, component = rtdcom->component; *last = component; - if (!component->driver->ops || - !component->driver->ops->open) - continue; - if (component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) { dev_err(component->dev, @@ -452,6 +448,10 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, return -ENODEV; } + if (!component->driver->ops || + !component->driver->ops->open) + continue; + ret = component->driver->ops->open(substream); if (ret < 0) { dev_err(component->dev, @@ -477,11 +477,9 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream, if (component == last) break; - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); + if (component->driver->ops && + component->driver->ops->close) + component->driver->ops->close(substream); if (component->driver->module_get_upon_open) module_put(component->dev->driver->owner); -- 2.7.4

3 2

[alsa-devel] [PATCH][RFC] ASoC: soc-dpm: fixup DAI active unbalance
by Kuninori Morimoto 21 May '19

21 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dai_link_event() is updating snd_soc_dai :: active, but it is unbalance. It counts up if it has startup callback. case SND_SOC_DAPM_PRE_PMU: ... snd_soc_dapm_widget_for_each_source_path(w, path) { ... if (source->driver->ops->startup) { ... => source->active++; } ... } ... But, always counts down case SND_SOC_DAPM_PRE_PMD: ... snd_soc_dapm_widget_for_each_source_path(w, path) { ... => source->active--; ... } This patch always counts up when SND_SOC_DAPM_PRE_PMD. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Liam I think this is bug, but I can't confirm it, because my driver need to have .startup. Thus, I added [RFC] on this patch. I'm happy if someone can confirm it. sound/soc/soc-dapm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index c864502..147ad9d 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3828,8 +3828,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret); goto out; } - source->active++; } + source->active++; ret = soc_dai_hw_params(&substream, params, source); if (ret < 0) goto out; @@ -3850,8 +3850,8 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret); goto out; } - sink->active++; } + sink->active++; ret = soc_dai_hw_params(&substream, params, sink); if (ret < 0) goto out; -- 2.7.4

4 3

[alsa-devel] [PATCH] ASoC: rt5677-spi: Handle over reading when flipping bytes
by Curtis Malainey 21 May '19

21 May '19

There is a case when a we want to read a large number of bytes that require a burst but is not a multiple of the word size (8). When this happens rt5677_spi_reverse will run off the end of the buffer. The solution is to tell spi_reverse the actual size of the destination and stop if we reach it even if we have data left that we read. Cc: Ben Zhang <benzh(a)chromium.org> Signed-off-by: Curtis Malainey <cujomalainey(a)chromium.org> --- sound/soc/codecs/rt5677-spi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5677-spi.c b/sound/soc/codecs/rt5677-spi.c index 84b6bd8b50e1..a4dfa0345c6e 100644 --- a/sound/soc/codecs/rt5677-spi.c +++ b/sound/soc/codecs/rt5677-spi.c @@ -101,7 +101,7 @@ static void rt5677_spi_reverse(u8 *dst, u32 dstlen, const u8 *src, u32 srclen) u32 word_size = min_t(u32, dstlen, 8); for (w = 0; w < dstlen; w += word_size) { - for (i = 0; i < word_size; i++) { + for (i = 0; i < word_size && i + w < dstlen; i++) { si = w + word_size - i - 1; dst[w + i] = si < srclen ? src[si] : 0; } @@ -152,8 +152,9 @@ int rt5677_spi_read(u32 addr, void *rxbuf, size_t len) status |= spi_sync(g_spi, &m); mutex_unlock(&spi_mutex); + /* Copy data back to caller buffer */ - rt5677_spi_reverse(cb + offset, t[1].len, body, t[1].len); + rt5677_spi_reverse(cb + offset, len - offset, body, t[1].len); } return status; } -- 2.21.0.1020.gf2820cf01a-goog

2 1

[alsa-devel] Applied "ASoC: rename functions that pollute the simple_xxx namespace" to the asoc tree
by Mark Brown 21 May '19

21 May '19

The patch ASoC: rename functions that pollute the simple_xxx namespace has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 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 b0a821daf0d04e5a8ae99829e24f2fe538f25763 Mon Sep 17 00:00:00 2001 From: Amir Goldstein <amir73il(a)gmail.com> Date: Thu, 16 May 2019 13:26:28 +0300 Subject: [PATCH] ASoC: rename functions that pollute the simple_xxx namespace include/linux/fs.h defines a bunch of simple fs helpers, (e.g. simple_rename) and we intend to add an fs helper named simple_remove. Rename the ASoC driver static functions, so they will not collide with the upcoming fs helper function name. Cc: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Amir Goldstein <amir73il(a)gmail.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/generic/simple-card.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 9b568f578bcd..d16e894fce2b 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -607,7 +607,7 @@ static int simple_soc_probe(struct snd_soc_card *card) return 0; } -static int simple_probe(struct platform_device *pdev) +static int asoc_simple_probe(struct platform_device *pdev) { struct asoc_simple_priv *priv; struct device *dev = &pdev->dev; @@ -705,7 +705,7 @@ static int simple_probe(struct platform_device *pdev) return ret; } -static int simple_remove(struct platform_device *pdev) +static int asoc_simple_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); @@ -726,8 +726,8 @@ static struct platform_driver asoc_simple_card = { .pm = &snd_soc_pm_ops, .of_match_table = simple_of_match, }, - .probe = simple_probe, - .remove = simple_remove, + .probe = asoc_simple_probe, + .remove = asoc_simple_remove, }; module_platform_driver(asoc_simple_card); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: rsnd: move pcm_new from snd_soc_component_driver to snd_soc_dai_driver
by Kuninori Morimoto 21 May '19

21 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> snd_soc_dai_driver :: pcm_new has snd_soc_dai as parameter, but snd_soc_component_driver :: pcm_new doesn't have it. rsnd driver needs snd_soc_dai at pcm_new. This patch moves .pcm_new from snd_soc_component_driver to snd_soc_dai_driver, and don't use rtd->cpu_dai anymore. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 120 +++++++++++++++++++++++------------------------ 1 file changed, 60 insertions(+), 60 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 37cb615..56e8dae 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1176,6 +1176,65 @@ static struct device_node *rsnd_dai_of_node(struct rsnd_priv *priv, return ret; } + +#define PREALLOC_BUFFER (32 * 1024) +#define PREALLOC_BUFFER_MAX (32 * 1024) + +static int rsnd_preallocate_pages(struct snd_soc_pcm_runtime *rtd, + struct rsnd_dai_stream *io, + int stream) +{ + struct rsnd_priv *priv = rsnd_io_to_priv(io); + struct device *dev = rsnd_priv_to_dev(priv); + struct snd_pcm_substream *substream; + + /* + * use Audio-DMAC dev if we can use IPMMU + * see + * rsnd_dmaen_attach() + */ + if (io->dmac_dev) + dev = io->dmac_dev; + + for (substream = rtd->pcm->streams[stream].substream; + substream; + substream = substream->next) { + snd_pcm_lib_preallocate_pages(substream, + SNDRV_DMA_TYPE_DEV, + dev, + PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); + } + + return 0; +} + +static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai) +{ + struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); + int ret; + + ret = rsnd_dai_call(pcm_new, &rdai->playback, rtd); + if (ret) + return ret; + + ret = rsnd_dai_call(pcm_new, &rdai->capture, rtd); + if (ret) + return ret; + + ret = rsnd_preallocate_pages(rtd, &rdai->playback, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + return ret; + + ret = rsnd_preallocate_pages(rtd, &rdai->capture, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + return ret; + + return 0; +} + static void __rsnd_dai_probe(struct rsnd_priv *priv, struct device_node *dai_np, int dai_i) @@ -1198,6 +1257,7 @@ static void __rsnd_dai_probe(struct rsnd_priv *priv, rdai->priv = priv; drv->name = rdai->name; drv->ops = &rsnd_soc_dai_ops; + drv->pcm_new = rsnd_pcm_new; snprintf(io_playback->name, RSND_DAI_NAME_SIZE, "DAI%d Playback", dai_i); @@ -1572,68 +1632,8 @@ int rsnd_kctrl_new(struct rsnd_mod *mod, /* * snd_soc_component */ - -#define PREALLOC_BUFFER (32 * 1024) -#define PREALLOC_BUFFER_MAX (32 * 1024) - -static int rsnd_preallocate_pages(struct snd_soc_pcm_runtime *rtd, - struct rsnd_dai_stream *io, - int stream) -{ - struct rsnd_priv *priv = rsnd_io_to_priv(io); - struct device *dev = rsnd_priv_to_dev(priv); - struct snd_pcm_substream *substream; - - /* - * use Audio-DMAC dev if we can use IPMMU - * see - * rsnd_dmaen_attach() - */ - if (io->dmac_dev) - dev = io->dmac_dev; - - for (substream = rtd->pcm->streams[stream].substream; - substream; - substream = substream->next) { - snd_pcm_lib_preallocate_pages(substream, - SNDRV_DMA_TYPE_DEV, - dev, - PREALLOC_BUFFER, PREALLOC_BUFFER_MAX); - } - - return 0; -} - -static int rsnd_pcm_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *dai = rtd->cpu_dai; - struct rsnd_dai *rdai = rsnd_dai_to_rdai(dai); - int ret; - - ret = rsnd_dai_call(pcm_new, &rdai->playback, rtd); - if (ret) - return ret; - - ret = rsnd_dai_call(pcm_new, &rdai->capture, rtd); - if (ret) - return ret; - - ret = rsnd_preallocate_pages(rtd, &rdai->playback, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - return ret; - - ret = rsnd_preallocate_pages(rtd, &rdai->capture, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - return ret; - - return 0; -} - static const struct snd_soc_component_driver rsnd_soc_component = { .ops = &rsnd_pcm_ops, - .pcm_new = rsnd_pcm_new, .name = "rsnd", }; -- 2.7.4

2 1

[alsa-devel] [PATCH][RFC] ASoC: soc-core: WARN() is not related to component->driver->probe
by Kuninori Morimoto 21 May '19

21 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_probe_component() has WARN() under if (component->driver->probe), but, this WARN() check is not related to .probe callback. So, it should be called at (B) instead of (A). This patch moves it out of if(). if (component->driver->probe) { ret = component->driver->probe(component); ... (A) WARN(...) } (B) WARN(...) Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Pierre-Louis, Vinod, Liam I think this patch is correct, but I'm not sure. I'm happy someone can confirm it. sound/soc/soc-core.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index cab30ae..7157d67 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1420,12 +1420,11 @@ static int soc_probe_component(struct snd_soc_card *card, "ASoC: failed to probe component %d\n", ret); goto err_probe; } - - WARN(dapm->idle_bias_off && - dapm->bias_level != SND_SOC_BIAS_OFF, - "codec %s can not start from non-off bias with idle_bias_off==1\n", - component->name); } + WARN(dapm->idle_bias_off && + dapm->bias_level != SND_SOC_BIAS_OFF, + "codec %s can not start from non-off bias with idle_bias_off==1\n", + component->name); /* machine specific init */ if (component->init) { -- 2.7.4

3 7

[alsa-devel] [PATCH 0/3] ASoC: Intel: fix rt5682-based machine drivers
by Pierre-Louis Bossart 21 May '19

21 May '19

We have two machine drivers based on rt5682 and they both need a correction for the mapping of the buttons, the initial mapping taken from the DA7219 is incorrect. Also fix ACPI tables, quirks as needed. Sathya Prakash M R (2): ASoC: Intel: sof-rt5682: Few minor fixes for AMP SSP and codec button ASoC: Intel: soc-acpi: Fix machine selection order Yong Zhi (1): ASoC: Intel: glk_rt5682_max98357a: Remap button control-function sound/soc/intel/boards/glk_rt5682_max98357a.c | 7 ++++--- sound/soc/intel/boards/sof_rt5682.c | 12 +++++++----- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 10 +++++----- 3 files changed, 16 insertions(+), 13 deletions(-) base-commit: d0723c9d845f9a6ff1069e5786ccda8e81566763 -- 2.20.1

2 7

[alsa-devel] [PATCH v2] ASoC: Mediatek: MT8183: enable IIR filter
by Shunli Wang 21 May '19

21 May '19

IIR fileter can remove DC offset. It must be enabled when dmic or amic connected to pmic is used. Signed-off-by: Shunli Wang <shunli.wang(a)mediatek.com> --- v2: correct the commit message --- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c index 017d7d1d9148..2b758a18c2ea 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c @@ -176,9 +176,6 @@ static int mtk_adda_ul_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ usleep_range(125, 135); - - /* reset dmic */ - afe_priv->mtkaif_dmic = 0; break; default: break; @@ -426,6 +423,17 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); + /* enable iir */ + ul_src_con0 |= (1 << UL_IIR_ON_TMP_CTL_SFT) & + UL_IIR_ON_TMP_CTL_MASK_SFT; + + /* 35Hz @ 48k */ + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_02_01, 0x00000000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_04_03, 0x00003FB8); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_06_05, 0x3FB80000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_08_07, 0x3FB80000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_10_09, 0x0000C048); + regmap_write(afe->regmap, AFE_ADDA_UL_SRC_CON0, ul_src_con0); /* mtkaif_rxif_data_mode = 0, amic */ -- 2.18.0

2 1

[alsa-devel] [PATCH V5] ASoC: cs42xx8: add reset-gpios in binding document
by S.j. Wang 21 May '19

21 May '19

Add reset-gpios property, which is optional. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in V5 - use GPIO_ACTIVE_LOW Changes in V4 - use gpios instead of gpio Changes in RESEND v3 - send updated binding document only Changes in v3 - update binding document. Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio Documentation/devicetree/bindings/sound/cs42xx8.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt index 8619a156d038..bbfe39347c20 100644 --- a/Documentation/devicetree/bindings/sound/cs42xx8.txt +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -14,6 +14,11 @@ Required properties: - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt +Optional properties: + + - reset-gpios : a GPIO spec to define which pin is connected to the chip's + !RESET pin + Example: cs42888: codec@48 { @@ -25,4 +30,5 @@ cs42888: codec@48 { VD-supply = <&reg_audio>; VLS-supply = <&reg_audio>; VLC-supply = <&reg_audio>; + reset-gpios = <&pca9557_b 1 GPIO_ACTIVE_LOW>; }; -- 2.21.0

3 2

[alsa-devel] [PATCH v2 0/5] ASoC: Intel: fix rt5682-based machine drivers
by Pierre-Louis Bossart 21 May '19

21 May '19

We have two machine drivers based on rt5682 and they both need a correction for the mapping of the buttons, the initial mapping taken from the DA7219 is incorrect. Also fix ACPI tables, quirks as needed, and add newlines for clarity Changes since v1: Split first patch in 3 (feedback from Mark). No code changes. Sathya Prakash M R (4): ASoC: Intel: sof-rt5682: fix for codec button mapping ASoC: Intel: sof-rt5682: fix AMP quirk support ASoC: Intel: sof-rt5682: add newline for clarity ASoC: Intel: soc-acpi: Fix machine selection order Yong Zhi (1): ASoC: Intel: glk_rt5682_max98357a: Remap button control-function sound/soc/intel/boards/glk_rt5682_max98357a.c | 7 ++++--- sound/soc/intel/boards/sof_rt5682.c | 12 +++++++----- sound/soc/intel/common/soc-acpi-intel-cnl-match.c | 10 +++++----- 3 files changed, 16 insertions(+), 13 deletions(-) -- 2.20.1

2 8

[alsa-devel] [PATCH 0/2] ASoC: Mediatek: MT8183: fix compile error
by Jiaxin Yu 21 May '19

21 May '19

This patch fixed the compile error which patches for support S32_LE and S24_LE. The patch is based on broonie "for-5.3" branch. Jiaxin Yu (2): ASoC: Mediatek: MT8183: fix compile error ASoC: Mediatek: MT8183: fix compile error sound/soc/mediatek/common/mtk-afe-fe-dai.c | 3 +-- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 18 +++++++++--------- 2 files changed, 10 insertions(+), 11 deletions(-) -- 1.8.1.1.dirty

2 4

[alsa-devel] [RFC] ASoC: Intel: skl_hda_dsp_common: set long name for skl_hda_card
by Rander Wang 21 May '19

21 May '19

skl_hda_generic machine driver is used by many different devices and userspace uses the card long name to differentiate devices. The card long name is figured out by DMI info and is in format of "vendor-product-version-board". Ucm file is searched by this card long name and one problem is different devices can't share one ucm file based on this type of long name. We have three different product devices with the same codecs and audio settings, and these devices can share the same ucm setting, but now we need to provide three ucm files with different long names, because we can't get a same long name from DMI info. The solution is to provide card long name in machine driver like bytcr_rt5640 and ASoC will use this long name to generate final long name. The card long name is composed of codec name, input and output enabled by devices. The long name should be initialized after hda codec is initialized and before sound card checking long name which is done after dai link initialization, so the long name is set in codec dai link initialization function. Possible card long names may be: skl-hda-ALC233-config135440 skl-hda-ALC700-config69649 skl-hda-ALC3204-config135185 Tested on intel hda platform whiskylake with SOF driver and gemilake with intel SST driver Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> --- sound/soc/intel/boards/skl_hda_dsp_common.c | 54 +++++++++++++++++++++++++++++ 1 file changed, 54 insertions(+) diff --git a/sound/soc/intel/boards/skl_hda_dsp_common.c b/sound/soc/intel/boards/skl_hda_dsp_common.c index 8b68f41a5b88..0193d2138e16 100644 --- a/sound/soc/intel/boards/skl_hda_dsp_common.c +++ b/sound/soc/intel/boards/skl_hda_dsp_common.c @@ -11,11 +11,18 @@ #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <sound/hda_codec.h> +#include "../../../pci/hda/hda_jack.h" +#include "../../../pci/hda/hda_local.h" +#include "../../../pci/hda/hda_auto_parser.h" +#include "../../../pci/hda/hda_generic.h" #include "../../codecs/hdac_hdmi.h" +#include "../../codecs/hdac_hda.h" #include "../skylake/skl.h" #include "skl_hda_dsp_common.h" #define NAME_SIZE 32 +static char skl_hda_long_name[NAME_SIZE]; int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) { @@ -39,6 +46,52 @@ int skl_hda_hdmi_add_pcm(struct snd_soc_card *card, int device) return 0; } +int skl_hda_long_name_init(struct snd_soc_pcm_runtime *rtd) +{ + struct snd_soc_rtdcom_list *new_rtdcom; + struct snd_soc_component *component; + struct hdac_hda_priv *hda_pvt; + struct auto_pin_cfg *autocfg; + struct hda_gen_spec *spec; + struct snd_soc_card *card; + const char *name; + int config; + + name = rtd->dai_link->codecs->name; + card = rtd->card; + + list_for_each_entry(new_rtdcom, &rtd->component_list, list) { + component = new_rtdcom->component; + if (name && strcmp(component->name, name) == 0) { + hda_pvt = snd_soc_component_get_drvdata(component); + + if (!hda_pvt) + return -EINVAL; + + spec = hda_pvt->codec.spec; + autocfg = &spec->autocfg; + + /* + * config is figured out by combining the number of + * enabled input and output. + */ + config = autocfg->speaker_outs | (autocfg->hp_outs << 4) + | (autocfg->dig_outs << 8) + | (autocfg->line_outs << 12) + | (autocfg->num_inputs << 16); + snprintf(skl_hda_long_name, sizeof(skl_hda_long_name), + "skl-hda-%s-config%d", + hda_pvt->codec.core.chip_name, + config); + + card->long_name = skl_hda_long_name; + break; + } + } + + return 0; +} + /* skl_hda_digital audio interface glue - connects codec <--> CPU */ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { /* Back End DAI links */ @@ -79,6 +132,7 @@ struct snd_soc_dai_link skl_hda_be_dai_links[HDA_DSP_MAX_BE_DAI_LINKS] = { .dpcm_playback = 1, .dpcm_capture = 1, .no_pcm = 1, + .init = skl_hda_long_name_init, }, { .name = "Digital Playback and Capture", -- 2.14.1

2 1

[alsa-devel] [PATCH - alsa-lib 4/4] pcm_file: add infile read support for mmap mode
by Adam Miartus 21 May '19

21 May '19

mmap_begin callback is used to copy data from input file to mmaped buffer guard for corner use of api (multiple mmap_begin calls by user) is introduced to check if next continuous buffer was already overwritten buffer is overwritten with input file data only in case of stream capture Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 10d227b..7d3439a 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -88,6 +88,7 @@ typedef struct { size_t buffer_bytes; struct wav_fmt wav_header; size_t filelen; + char ifmmap_overwritten; } snd_pcm_file_t; #if __BYTE_ORDER == __LITTLE_ENDIAN @@ -626,6 +627,8 @@ static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, const snd_pcm_channel_area_t *areas; snd_pcm_sframes_t result; + file->ifmmap_overwritten = 0; + result = snd_pcm_mmap_begin(file->gen.slave, &areas, &ofs, &siz); if (result >= 0) { assert(ofs == offset && siz == size); @@ -636,6 +639,32 @@ static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, return result; } +static int snd_pcm_file_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, + snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames) +{ + snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas_if[pcm->channels]; + snd_pcm_uframes_t frames_if; + void *buffer = NULL; + int result; + + result = snd_pcm_mmap_begin(file->gen.slave, areas, offset, frames); + if (result < 0) + return result; + + if (pcm->stream != SND_PCM_STREAM_CAPTURE) + return result; + + /* user may run mmap_begin without mmap_commit multiple times in row */ + if (file->ifmmap_overwritten) + return result; + file->ifmmap_overwritten = 1; + + snd_pcm_file_areas_read_infile(pcm, *areas, *offset, *frames); + + return result; +} + static int snd_pcm_file_hw_free(snd_pcm_t *pcm) { snd_pcm_file_t *file = pcm->private_data; @@ -662,6 +691,7 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) file->wbuf_size = slave->buffer_size * 2; file->wbuf_size_bytes = snd_pcm_frames_to_bytes(slave, file->wbuf_size); file->wbuf_used_bytes = 0; + file->ifmmap_overwritten = 0; assert(!file->wbuf); file->wbuf = malloc(file->wbuf_size_bytes); if (file->wbuf == NULL) { @@ -773,7 +803,7 @@ static const snd_pcm_fast_ops_t snd_pcm_file_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .htimestamp = snd_pcm_generic_htimestamp, - .mmap_begin = NULL, + .mmap_begin = snd_pcm_file_mmap_begin, }; /** -- 2.7.4

1 0

[alsa-devel] [PATCH - alsa-lib 3/4] pcm: add mmap_begin callback to snd_pcm_fast_ops_t api
by Adam Miartus 21 May '19

21 May '19

main motivation for adding the callback is to use it to enable operation on mmaped buffer before user access for pcm_file plugin support for MMAP read access with masking by data from input file is not implemented for pcm_file plugin, by adding this callback implementing such feature can be done by rewriting next continuous portion of buffer on each mmap_begin call plugins like softvol use pcm_plugin interface and overwrite the buffer by looping around it in avail_update callback, this patch hopes to simplify the task by adding new api callback, removing the need for rewriting pcm_file (to use pcm_plugin callbacks) and careful checking when looping around whole mmaped buffer Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index 3a71d79..323926e 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -7129,7 +7129,13 @@ int __snd_pcm_mmap_begin(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t f; snd_pcm_uframes_t avail; const snd_pcm_channel_area_t *xareas; + assert(pcm && areas && offset && frames); + + if (pcm->fast_ops->mmap_begin) + return pcm->fast_ops->mmap_begin(pcm->fast_op_arg, areas, offset, frames); + + /* fallback for plugins that do not specify new callback */ xareas = snd_pcm_mmap_areas(pcm); if (xareas == NULL) return -EBADFD; diff --git a/src/pcm/pcm_dmix.c b/src/pcm/pcm_dmix.c index 274726e..96049e0 100644 --- a/src/pcm/pcm_dmix.c +++ b/src/pcm/pcm_dmix.c @@ -990,6 +990,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dmix_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_dmix_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_dshare.c b/src/pcm/pcm_dshare.c index b75809c..328ef83 100644 --- a/src/pcm/pcm_dshare.c +++ b/src/pcm/pcm_dshare.c @@ -667,6 +667,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dshare_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_dsnoop.c b/src/pcm/pcm_dsnoop.c index 58b1e53..5b9d9f5 100644 --- a/src/pcm/pcm_dsnoop.c +++ b/src/pcm/pcm_dsnoop.c @@ -544,6 +544,7 @@ static const snd_pcm_fast_ops_t snd_pcm_dsnoop_fast_ops = { .poll_descriptors = snd_pcm_direct_poll_descriptors, .poll_descriptors_count = NULL, .poll_revents = snd_pcm_direct_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index dcaa41d..10d227b 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -773,6 +773,7 @@ static const snd_pcm_fast_ops_t snd_pcm_file_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .htimestamp = snd_pcm_generic_htimestamp, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_generic.h b/src/pcm/pcm_generic.h index dfbe21a..683e60b 100644 --- a/src/pcm/pcm_generic.h +++ b/src/pcm/pcm_generic.h @@ -161,4 +161,3 @@ snd_pcm_chmap_query_t **snd_pcm_generic_query_chmaps(snd_pcm_t *pcm); snd_pcm_chmap_t *snd_pcm_generic_get_chmap(snd_pcm_t *pcm); int snd_pcm_generic_set_chmap(snd_pcm_t *pcm, const snd_pcm_chmap_t *map); int snd_pcm_generic_may_wait_for_avail_min(snd_pcm_t *pcm, snd_pcm_uframes_t avail); - diff --git a/src/pcm/pcm_hooks.c b/src/pcm/pcm_hooks.c index 4416d36..14396fa 100644 --- a/src/pcm/pcm_hooks.c +++ b/src/pcm/pcm_hooks.c @@ -200,6 +200,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hooks_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index 91370a8..8ee1d69 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1481,6 +1481,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops = { .poll_descriptors = NULL, .poll_descriptors_count = NULL, .poll_revents = NULL, + .mmap_begin = NULL, }; static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops_timer = { @@ -1512,6 +1513,7 @@ static const snd_pcm_fast_ops_t snd_pcm_hw_fast_ops_timer = { .poll_descriptors = snd_pcm_hw_poll_descriptors, .poll_descriptors_count = snd_pcm_hw_poll_descriptors_count, .poll_revents = snd_pcm_hw_poll_revents, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_ioplug.c b/src/pcm/pcm_ioplug.c index a437ca3..3e19499 100644 --- a/src/pcm/pcm_ioplug.c +++ b/src/pcm/pcm_ioplug.c @@ -940,6 +940,7 @@ static const snd_pcm_fast_ops_t snd_pcm_ioplug_fast_ops = { .poll_descriptors_count = snd_pcm_ioplug_poll_descriptors_count, .poll_descriptors = snd_pcm_ioplug_poll_descriptors, .poll_revents = snd_pcm_ioplug_poll_revents, + .mmap_begin = NULL, }; #endif /* !DOC_HIDDEN */ diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index d52229d..d5726eb 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -184,6 +184,7 @@ typedef struct { int (*poll_descriptors)(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int space); /* locked */ int (*poll_revents)(snd_pcm_t *pcm, struct pollfd *pfds, unsigned int nfds, unsigned short *revents); /* locked */ int (*may_wait_for_avail_min)(snd_pcm_t *pcm, snd_pcm_uframes_t avail); + int (*mmap_begin)(snd_pcm_t *pcm, const snd_pcm_channel_area_t **areas, snd_pcm_uframes_t *offset, snd_pcm_uframes_t *frames); /* locked */ } snd_pcm_fast_ops_t; struct _snd_pcm { diff --git a/src/pcm/pcm_meter.c b/src/pcm/pcm_meter.c index 20b4187..28bb578 100644 --- a/src/pcm/pcm_meter.c +++ b/src/pcm/pcm_meter.c @@ -548,6 +548,7 @@ static const snd_pcm_fast_ops_t snd_pcm_meter_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_mmap_emul.c b/src/pcm/pcm_mmap_emul.c index 009cebb..6fa5fe6 100644 --- a/src/pcm/pcm_mmap_emul.c +++ b/src/pcm/pcm_mmap_emul.c @@ -401,6 +401,7 @@ static const snd_pcm_fast_ops_t snd_pcm_mmap_emul_fast_ops = { .poll_descriptors_count = snd_pcm_generic_poll_descriptors_count, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_generic_may_wait_for_avail_min, + .mmap_begin = NULL, }; #ifndef DOC_HIDDEN diff --git a/src/pcm/pcm_multi.c b/src/pcm/pcm_multi.c index 53c414d..069d224 100644 --- a/src/pcm/pcm_multi.c +++ b/src/pcm/pcm_multi.c @@ -1042,6 +1042,7 @@ static const snd_pcm_fast_ops_t snd_pcm_multi_fast_ops = { .poll_descriptors = snd_pcm_multi_poll_descriptors, .poll_revents = snd_pcm_multi_poll_revents, .may_wait_for_avail_min = snd_pcm_multi_may_wait_for_avail_min, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_null.c b/src/pcm/pcm_null.c index 1d81548..e6fa049 100644 --- a/src/pcm/pcm_null.c +++ b/src/pcm/pcm_null.c @@ -361,6 +361,7 @@ static const snd_pcm_fast_ops_t snd_pcm_null_fast_ops = { .avail_update = snd_pcm_null_avail_update, .mmap_commit = snd_pcm_null_mmap_commit, .htimestamp = snd_pcm_generic_real_htimestamp, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_plugin.c b/src/pcm/pcm_plugin.c index ea60eb9..0714d2c 100644 --- a/src/pcm/pcm_plugin.c +++ b/src/pcm/pcm_plugin.c @@ -646,6 +646,7 @@ const snd_pcm_fast_ops_t snd_pcm_plugin_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_generic_poll_revents, .may_wait_for_avail_min = snd_pcm_plugin_may_wait_for_avail_min, + .mmap_begin = NULL, }; #endif diff --git a/src/pcm/pcm_rate.c b/src/pcm/pcm_rate.c index 5dc8a92..f29739b 100644 --- a/src/pcm/pcm_rate.c +++ b/src/pcm/pcm_rate.c @@ -1217,6 +1217,7 @@ static const snd_pcm_fast_ops_t snd_pcm_rate_fast_ops = { .poll_descriptors = snd_pcm_generic_poll_descriptors, .poll_revents = snd_pcm_rate_poll_revents, .may_wait_for_avail_min = snd_pcm_plugin_may_wait_for_avail_min, + .mmap_begin = NULL, }; static const snd_pcm_ops_t snd_pcm_rate_ops = { diff --git a/src/pcm/pcm_share.c b/src/pcm/pcm_share.c index 5a540c4..b803a82 100644 --- a/src/pcm/pcm_share.c +++ b/src/pcm/pcm_share.c @@ -1346,6 +1346,7 @@ static const snd_pcm_fast_ops_t snd_pcm_share_fast_ops = { .avail_update = snd_pcm_share_avail_update, .htimestamp = snd_pcm_share_htimestamp, .mmap_commit = snd_pcm_share_mmap_commit, + .mmap_begin = NULL, }; /** diff --git a/src/pcm/pcm_shm.c b/src/pcm/pcm_shm.c index 26a27a5..a18720f 100644 --- a/src/pcm/pcm_shm.c +++ b/src/pcm/pcm_shm.c @@ -629,6 +629,7 @@ static const snd_pcm_fast_ops_t snd_pcm_shm_fast_ops = { .avail_update = snd_pcm_shm_avail_update, .mmap_commit = snd_pcm_shm_mmap_commit, .htimestamp = snd_pcm_shm_htimestamp, + .mmap_begin = NULL, }; static int make_local_socket(const char *filename) -- 2.7.4

1 0

[alsa-devel] [PATCH - alsa-lib 2/4] pcm_file: use snd_pcm_file_areas_read_infile for readi
by Adam Miartus 21 May '19

21 May '19

use previously introduced helper function, this commit unifies behavior of readi and readn corner case behavior of readi is changed by this commit, previously, in case 0 bytes were red from file (EOF), frames = 0 was returned, signaling api user as if no data was red from slave, after the patch, amount of frames red from slave with data red from slave stored in buffer is returned when EOF is reached Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 3c68265..dcaa41d 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -579,22 +579,21 @@ static snd_pcm_sframes_t snd_pcm_file_readi(snd_pcm_t *pcm, void *buffer, snd_pc { snd_pcm_file_t *file = pcm->private_data; snd_pcm_channel_area_t areas[pcm->channels]; - snd_pcm_sframes_t n; + snd_pcm_sframes_t frames; + + __snd_pcm_lock(pcm); + + frames = _snd_pcm_readi(file->gen.slave, buffer, size); + if (frames <= 0) + return frames; - n = _snd_pcm_readi(file->gen.slave, buffer, size); - if (n <= 0) - return n; - if (file->ifd >= 0) { - __snd_pcm_lock(pcm); - n = read(file->ifd, buffer, n * pcm->frame_bits / 8); - __snd_pcm_unlock(pcm); - if (n < 0) - return n; - n = n * 8 / pcm->frame_bits; - } snd_pcm_areas_from_buf(pcm, areas, buffer); - snd_pcm_file_add_frames(pcm, areas, 0, n); - return n; + snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); + snd_pcm_file_add_frames(pcm, areas, 0, frames); + + __snd_pcm_unlock(pcm); + + return frames; } /* locking */ -- 2.7.4

1 0

[alsa-devel] [PATCH - alsa-lib 1/4] pcm_file: add support for infile reading in non interleaved mode
by Adam Miartus 21 May '19

21 May '19

add helper function to copy input file data to buffer mapped by areas, in case of an error, do not fill the areas, allowing device read buffer to be provided to api caller previously unused rbuf variable is reused for this purpose Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 3a19cef..3c68265 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -77,6 +77,7 @@ typedef struct { snd_pcm_uframes_t appl_ptr; snd_pcm_uframes_t file_ptr_bytes; snd_pcm_uframes_t wbuf_size; + snd_pcm_uframes_t rbuf_size; size_t wbuf_size_bytes; size_t wbuf_used_bytes; char *wbuf; @@ -266,6 +267,39 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) return 0; } +/* fill areas with data from input file, return bytes red */ +static int snd_pcm_file_areas_read_infile(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t frames) +{ + snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas_if[pcm->channels]; + ssize_t bytes; + + if (file->ifd < 0) + return -EBADF; + + if (file->rbuf == NULL) + return -ENOMEM; + + if (file->rbuf_size < frames) { + SYSERR("requested more frames than pcm buffer"); + return -ENOMEM; + } + + bytes = read(file->ifd, file->rbuf, snd_pcm_frames_to_bytes(pcm, frames)); + if (bytes < 0) { + SYSERR("read from file failed, error: %d", bytes); + return bytes; + } + + snd_pcm_areas_from_buf(pcm, areas_if, file->rbuf); + snd_pcm_areas_copy(areas, offset, areas_if, 0, pcm->channels, snd_pcm_bytes_to_frames(pcm, bytes), pcm->format); + + return bytes; +} + static void setup_wav_header(snd_pcm_t *pcm, struct wav_fmt *fmt) { fmt->fmt = TO_LE16(0x01); @@ -568,19 +602,19 @@ static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm { snd_pcm_file_t *file = pcm->private_data; snd_pcm_channel_area_t areas[pcm->channels]; - snd_pcm_sframes_t n; + snd_pcm_sframes_t frames; - if (file->ifd >= 0) { - SNDERR("DEBUG: Noninterleaved read not yet implemented.\n"); - return 0; /* TODO: Noninterleaved read */ - } + __snd_pcm_lock(pcm); + frames = _snd_pcm_readn(file->gen.slave, bufs, size); + if (frames <= 0) + return frames; - n = _snd_pcm_readn(file->gen.slave, bufs, size); - if (n > 0) { - snd_pcm_areas_from_bufs(pcm, areas, bufs); - snd_pcm_file_add_frames(pcm, areas, 0, n); - } - return n; + snd_pcm_areas_from_bufs(pcm, areas, bufs); + snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); + snd_pcm_file_add_frames(pcm, areas, 0, frames); + + __snd_pcm_unlock(pcm); + return frames; } static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, @@ -609,9 +643,11 @@ static int snd_pcm_file_hw_free(snd_pcm_t *pcm) free(file->wbuf); free(file->wbuf_areas); free(file->final_fname); + free(file->rbuf); file->wbuf = NULL; file->wbuf_areas = NULL; file->final_fname = NULL; + file->rbuf = NULL; return snd_pcm_hw_free(file->gen.slave); } @@ -638,6 +674,15 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_file_hw_free(pcm); return -ENOMEM; } + assert(!file->rbuf); + file->rbuf_size = slave->buffer_size; + file->rbuf_size_bytes = snd_pcm_frames_to_bytes(slave, file->rbuf_size); + file->rbuf_used_bytes = 0; + file->rbuf = malloc(file->rbuf_size_bytes); + if (file->rbuf == NULL) { + snd_pcm_file_hw_free(pcm); + return -ENOMEM; + } file->appl_ptr = file->file_ptr_bytes = 0; for (channel = 0; channel < slave->channels; ++channel) { snd_pcm_channel_area_t *a = &file->wbuf_areas[channel]; -- 2.7.4

1 0

Re: [alsa-devel] [ALSA patch] FW: [PATCH - alsa-lib 1/4] pcm_file: add support for infile reading in non interleaved mode
by Takashi Iwai 21 May '19

21 May '19

On Mon, 20 May 2019 18:49:34 +0200, Miartus, Adam (Arion Recruitment; ADITG/ESM) wrote: > > From: Adam Miartus <amiartus(a)de.adit-jv.com> > > add helper function to copy input file data to buffer mapped by areas, in case of an error, do not fill the areas, allowing device read buffer to be provided to api caller > > previously unused rbuf variable is reused for this purpose > > Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> > Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> > > diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 3a19cef..7998b64 100644 > --- a/src/pcm/pcm_file.c > +++ b/src/pcm/pcm_file.c > @@ -77,6 +77,7 @@ typedef struct { > snd_pcm_uframes_t appl_ptr; > snd_pcm_uframes_t file_ptr_bytes; > snd_pcm_uframes_t wbuf_size; > + snd_pcm_uframes_t rbuf_size; > size_t wbuf_size_bytes; > size_t wbuf_used_bytes; > char *wbuf; > @@ -266,6 +267,37 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) > return 0; > } > > +/* fill areas with data from input file, return bytes red */ static int > +snd_pcm_file_areas_read_infile(snd_pcm_t *pcm, const snd_pcm_channel_area_t *areas, > + snd_pcm_uframes_t offset, snd_pcm_uframes_t frames) { Please follow the standard coding style. > + snd_pcm_file_t *file = pcm->private_data; > + snd_pcm_channel_area_t areas_if[pcm->channels]; > + ssize_t bytes; > + > + if (file->ifd < 0) > + return -EBADF; > + > + if (file->rbuf == NULL) > + return -ENOMEM; > + > + if (file->rbuf_size < frames) { > + SYSERR("requested more frames than pcm buffer"); > + return -ENOMEM; > + } > + > + bytes = read(file->ifd, file->rbuf, snd_pcm_frames_to_bytes(pcm, frames)); > + if (bytes < 0) { > + SYSERR("read from file failed, error: %d", bytes); > + return bytes; > + } > + > + snd_pcm_areas_from_buf(pcm, areas_if, file->rbuf); > + snd_pcm_areas_copy(areas, offset, areas_if, 0, pcm->channels, > +snd_pcm_bytes_to_frames(pcm, bytes), pcm->format); Wrong indentation. thanks, Takashi

2 2

[alsa-devel] [PATCH - alsa-lib 1/4] pcm_file: add support for infile reading in non interleaved mode
by Adam Miartus 21 May '19

21 May '19

add helper function to copy input file data to buffer mapped by areas, in case of an error, do not fill the areas, allowing device read buffer to be provided to api caller previously unused rbuf variable is reused for this purpose Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 3a19cef..3c68265 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -77,6 +77,7 @@ typedef struct { snd_pcm_uframes_t appl_ptr; snd_pcm_uframes_t file_ptr_bytes; snd_pcm_uframes_t wbuf_size; + snd_pcm_uframes_t rbuf_size; size_t wbuf_size_bytes; size_t wbuf_used_bytes; char *wbuf; @@ -266,6 +267,39 @@ static int snd_pcm_file_open_output_file(snd_pcm_file_t *file) return 0; } +/* fill areas with data from input file, return bytes red */ +static int snd_pcm_file_areas_read_infile(snd_pcm_t *pcm, + const snd_pcm_channel_area_t *areas, + snd_pcm_uframes_t offset, + snd_pcm_uframes_t frames) +{ + snd_pcm_file_t *file = pcm->private_data; + snd_pcm_channel_area_t areas_if[pcm->channels]; + ssize_t bytes; + + if (file->ifd < 0) + return -EBADF; + + if (file->rbuf == NULL) + return -ENOMEM; + + if (file->rbuf_size < frames) { + SYSERR("requested more frames than pcm buffer"); + return -ENOMEM; + } + + bytes = read(file->ifd, file->rbuf, snd_pcm_frames_to_bytes(pcm, frames)); + if (bytes < 0) { + SYSERR("read from file failed, error: %d", bytes); + return bytes; + } + + snd_pcm_areas_from_buf(pcm, areas_if, file->rbuf); + snd_pcm_areas_copy(areas, offset, areas_if, 0, pcm->channels, snd_pcm_bytes_to_frames(pcm, bytes), pcm->format); + + return bytes; +} + static void setup_wav_header(snd_pcm_t *pcm, struct wav_fmt *fmt) { fmt->fmt = TO_LE16(0x01); @@ -568,19 +602,19 @@ static snd_pcm_sframes_t snd_pcm_file_readn(snd_pcm_t *pcm, void **bufs, snd_pcm { snd_pcm_file_t *file = pcm->private_data; snd_pcm_channel_area_t areas[pcm->channels]; - snd_pcm_sframes_t n; + snd_pcm_sframes_t frames; - if (file->ifd >= 0) { - SNDERR("DEBUG: Noninterleaved read not yet implemented.\n"); - return 0; /* TODO: Noninterleaved read */ - } + __snd_pcm_lock(pcm); + frames = _snd_pcm_readn(file->gen.slave, bufs, size); + if (frames <= 0) + return frames; - n = _snd_pcm_readn(file->gen.slave, bufs, size); - if (n > 0) { - snd_pcm_areas_from_bufs(pcm, areas, bufs); - snd_pcm_file_add_frames(pcm, areas, 0, n); - } - return n; + snd_pcm_areas_from_bufs(pcm, areas, bufs); + snd_pcm_file_areas_read_infile(pcm, areas, 0, frames); + snd_pcm_file_add_frames(pcm, areas, 0, frames); + + __snd_pcm_unlock(pcm); + return frames; } static snd_pcm_sframes_t snd_pcm_file_mmap_commit(snd_pcm_t *pcm, @@ -609,9 +643,11 @@ static int snd_pcm_file_hw_free(snd_pcm_t *pcm) free(file->wbuf); free(file->wbuf_areas); free(file->final_fname); + free(file->rbuf); file->wbuf = NULL; file->wbuf_areas = NULL; file->final_fname = NULL; + file->rbuf = NULL; return snd_pcm_hw_free(file->gen.slave); } @@ -638,6 +674,15 @@ static int snd_pcm_file_hw_params(snd_pcm_t *pcm, snd_pcm_hw_params_t * params) snd_pcm_file_hw_free(pcm); return -ENOMEM; } + assert(!file->rbuf); + file->rbuf_size = slave->buffer_size; + file->rbuf_size_bytes = snd_pcm_frames_to_bytes(slave, file->rbuf_size); + file->rbuf_used_bytes = 0; + file->rbuf = malloc(file->rbuf_size_bytes); + if (file->rbuf == NULL) { + snd_pcm_file_hw_free(pcm); + return -ENOMEM; + } file->appl_ptr = file->file_ptr_bytes = 0; for (channel = 0; channel < slave->channels; ++channel) { snd_pcm_channel_area_t *a = &file->wbuf_areas[channel]; -- 2.7.4

1 0

Re: [alsa-devel] [ALSA patch] [PATCH - alsa-lib 4/4] pcm_file: add infile read support for mmap mode
by Takashi Iwai 21 May '19

21 May '19

On Mon, 20 May 2019 18:51:06 +0200, Miartus, Adam (Arion Recruitment; ADITG/ESM) wrote: > > From: Adam Miartus <amiartus(a)de.adit-jv.com> > > mmap_begin callback is used to copy data from input file to mmaped buffer > > guard for corner use of api (multiple mmap_begin calls by user) is introduced to check if next continuous buffer was already overwritten > > buffer is overwritten with input file data only in case of stream capture > > Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> > Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> Can't we copy the data in snd_pcm_file_mmap_commit() just like the playback case? thanks, Takashi

3 4

[alsa-devel] [PATCH 0/5] ALSA: firewire-lib: unify tracing events for isoc packet
by Takashi Sakamoto 21 May '19

21 May '19

Hi, This patchset is for Linux kernel v5.3 development. ALSA IEC 61883-1/6 packet streaming engine has four types of tracing events to probe data of handled isochronous packets on IEEE 1394 bus. The most parameters of data are common independent of type of isochronous payload and inbound/outbound direction. This commit unifies the tracing events as 'amdtp_packet'. Instead of named tracing events, filtering functionality of Linux tracing framework is available to filter out target events, like: $ echo 'src == 0xffc0' > /sys/kernel/debug/tracing/events/snd_firewire_lib/amdtp_packet/filter $ echo 1 > /sys/kernel/debug/tracing/events/snd_firewire_lib/amdtp_packet/enable $ cat /sys/kernel/debug/tracing/trace ... ...: amdtp_packet: 00 0594 ffc0 ffc1 00 002 00 240 41 1 09 {0x0,0x2,0x0,0xf0,0x90,0x1,0xff,0xff} ...: amdtp_packet: 00 0595 ffc0 ffc1 00 018 08 240 42 1 10 {0x0,0x2,0x0,0xf0,0x90,0x1,0x4b,0x3d} Takashi Sakamoto (5): firewire-lib: use the same type of argument for CIP header for tracing event firewire-lib: add data_blocks/data_block_counter parameter to in_packet/out_packet tracing events firewire-lib: use the same unit for payload argument in tracing events firewire-lib: use dynamic array for CIP header of tracing events firewire-lib: unify tracing events to 'amdtp_packet' event sound/firewire/amdtp-stream-trace.h | 163 ++++------------------------ sound/firewire/amdtp-stream.c | 16 +-- 2 files changed, 28 insertions(+), 151 deletions(-) -- 2.20.1

2 6

[alsa-devel] [PATCH 07/10] mfd: madera: point to the right pinctrl binding file
by Mauro Carvalho Chehab 20 May '19

20 May '19

The reference to Documentation/pinctrl.txt doesn't exist, but there is an specific binding for the madera driver. So, point to it. Signed-off-by: Mauro Carvalho Chehab <mchehab+samsung(a)kernel.org> --- include/linux/mfd/madera/pdata.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/include/linux/mfd/madera/pdata.h b/include/linux/mfd/madera/pdata.h index 8dc852402dbb..c7e0658eb74b 100644 --- a/include/linux/mfd/madera/pdata.h +++ b/include/linux/mfd/madera/pdata.h @@ -34,7 +34,8 @@ struct madera_codec_pdata; * @micvdd: Substruct of pdata for the MICVDD regulator * @irq_flags: Mode for primary IRQ (defaults to active low) * @gpio_base: Base GPIO number - * @gpio_configs: Array of GPIO configurations (See Documentation/pinctrl.txt) + * @gpio_configs: Array of GPIO configurations + * (See Documentation/devicetree/bindings/pinctrl/cirrus,madera-pinctrl.txt) * @n_gpio_configs: Number of entries in gpio_configs * @gpsw: General purpose switch mode setting. Depends on the external * hardware connected to the switch. (See the SW1_MODE field -- 2.21.0

2 2

[alsa-devel] [PATCH] ASoC: Intel: bytcr_5640.c:Refactored if statement and removed buffer
by nariman 20 May '19

20 May '19

From: Nariman Etemadi <narimantos(a)gmail.com> in function snd_byt_rt5640_mc_probe and removed buffer yt_rt5640_codec_aif_name & byt_rt5640_cpu_dai_name Signed-off-by: Nariman Etemadi <narimantos(a)gmail.com> --- sound/soc/intel/boards/bytcr_rt5640.c | 26 ++++---------------------- 1 file changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5640.c b/sound/soc/intel/boards/bytcr_rt5640.c index 940eb27158da..0d91642ca287 100644 --- a/sound/soc/intel/boards/bytcr_rt5640.c +++ b/sound/soc/intel/boards/bytcr_rt5640.c @@ -1075,8 +1075,6 @@ static struct snd_soc_dai_link byt_rt5640_dais[] = { /* SoC card */ static char byt_rt5640_codec_name[SND_ACPI_I2C_ID_LEN]; -static char byt_rt5640_codec_aif_name[12]; /* = "rt5640-aif[1|2]" */ -static char byt_rt5640_cpu_dai_name[10]; /* = "ssp[0|2]-port" */ static char byt_rt5640_long_name[40]; /* = "bytcr-rt5640-*-spk-*-mic" */ static int byt_rt5640_suspend(struct snd_soc_card *card) @@ -1268,28 +1266,12 @@ static int snd_byt_rt5640_mc_probe(struct platform_device *pdev) log_quirks(&pdev->dev); if ((byt_rt5640_quirk & BYT_RT5640_SSP2_AIF2) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - - /* fixup codec aif name */ - snprintf(byt_rt5640_codec_aif_name, - sizeof(byt_rt5640_codec_aif_name), - "%s", "rt5640-aif2"); - - byt_rt5640_dais[dai_index].codec_dai_name = - byt_rt5640_codec_aif_name; - } + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + byt_rt5640_dais[dai_index].codec_dai_name = "rt5640-aif2"; if ((byt_rt5640_quirk & BYT_RT5640_SSP0_AIF1) || - (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) { - - /* fixup cpu dai name name */ - snprintf(byt_rt5640_cpu_dai_name, - sizeof(byt_rt5640_cpu_dai_name), - "%s", "ssp0-port"); - - byt_rt5640_dais[dai_index].cpu_dai_name = - byt_rt5640_cpu_dai_name; - } + (byt_rt5640_quirk & BYT_RT5640_SSP0_AIF2)) + byt_rt5640_dais[dai_index].cpu_dai_name = "ssp0-port"; if (byt_rt5640_quirk & BYT_RT5640_MCLK_EN) { priv->mclk = devm_clk_get(&pdev->dev, "pmc_plt_clk_3"); -- 2.20.1

3 5

[alsa-devel] [PATCH 00/12] ASoC: SOF: stability fixes
by Pierre-Louis Bossart 20 May '19

20 May '19

No new functionality here but fixes to the IPC, memory allocation, workqueues, control, runtime_pm and HDaudio support. These issues were identified during the integration/productization of new platforms. I added a 'Fixes' tag to make sure backports/stable can pick up these patches, in case it's too late for them to land in 5.2 The next batch for SOF will be a set of new capabilities for trace, IPC test, Hdaudio support, they will be submitted after additional tests and once these fixes are reviewed/merged. Thanks! Amadeusz Sławiński (1): ALSA: hdac: fix memory release for SST and SOF drivers Guennadi Liakhovetski (1): ASoC: SOF: ipc: fix a race, leading to IPC timeouts Keyon Jie (1): ASoC: SOF: control: correct the copy size for bytes kcontrol put Libin Yang (2): ASoC: SOF: pcm: clear hw_params_upon_resume flag correctly ASoC: SOF: Intel: hda-codec: fix memory allocation Pierre-Louis Bossart (2): ASoC: SOF: core: fix error handling with the probe workqueue ASoC: SOF: pcm: remove warning - initialize workqueue on open Ranjani Sridharan (3): ASoC: SOF: core: remove DSP after unregistering machine driver ASoC: SOF: core: remove snd_soc_unregister_component in case of error ASoC: SOF: pcm: remove runtime PM calls during pcm open/close Zhu Yingjiang (2): ASoC: SOF: Intel: hda: fix the hda init chip ASoC: SOF: Intel: hda: use the defined ppcap functions sound/hda/ext/hdac_ext_bus.c | 1 - sound/soc/sof/control.c | 9 +-- sound/soc/sof/core.c | 29 +++++++-- sound/soc/sof/intel/bdw.c | 11 ++-- sound/soc/sof/intel/byt.c | 12 ++-- sound/soc/sof/intel/cnl.c | 6 ++ sound/soc/sof/intel/hda-codec.c | 6 +- sound/soc/sof/intel/hda-ctrl.c | 102 +++++++++++++++++++++++++++--- sound/soc/sof/intel/hda-ipc.c | 19 +++++- sound/soc/sof/intel/hda.c | 109 +++++++------------------------- sound/soc/sof/ipc.c | 13 ---- sound/soc/sof/pcm.c | 37 ++--------- 12 files changed, 188 insertions(+), 166 deletions(-) -- 2.20.1

3 18

[alsa-devel] udev: Access permission of fw character device in which fw node includes unit handled by ALSA
by Takashi Sakamoto 20 May '19

20 May '19

Hi all, I'm an ALSA developer, mainly committing to drivers for audio and music units on IEEE 1394 bus (yes, it's legacy FireWire). I have a concern about access permission for fw character device. In the last few years 9 drivers have been added to support 120 models around[1]. The drivers allow ALSA applications to transfer audio data frames and MIDI messages, but they expect userspace application to use fw character device for operations of the other functionalities such as physical volume controls. At present, for some models, udevd configures fw character device for good access permission to userspace applications. On the other hand, for the other models, udevd doesn't. In my opinion, for one half of supported models, fw character device is configured with root:root/0500. I'd like to change this inconvenience, changing group ownership of the special file to 'audio', and add ACL +uaccess when logging in. For your information, I prepare a repository including image of configuration ROM from supported devices[2]. At present, 80 images are added. You can parse the image by crpp in linux-firewire-utils repository[3]. At present I assume three options: 1. Add entries into udev rules for each of device This is an enhancement of existent rules[4][5] and simple solution. But developers always take care of adding new entries when users suggests, perhaps. 2. Reconfigure fw character device when ALSA firewire driver is attached to unit IEEE 1394 specification refers to IEEE 1212 to represent device and its functionalities. Each device is represented as 'node' and its functionalities are represented as 'unit'. The information about 'node' and 'unit' is in configuration ROM. Linux firewire subsystem reads content of the ROM, parses it and adds fw character device to system for 'node', then binds in-kernel driver to 'unit'. Kevents for the node and unit are generated separately. If udevd handles unit kevent to seek binding driver then configures fw character device, developers don't need to maintain rule list. Although one node is allowed to have several units, nut the most of supported models have only one unit, except for Applie iSight[6]. 3. Fulfill hwdb to have supplemental information I'm not good at hwdb, but according to its name, hwdb is good to store device dependent information. If adding some hints to database and using the information, this issue might be solved. I'm happy to receive your comments to improve this situation. As a mockup of such control application, I wrote libhinawa[7] and hinawa-utils[8]. They're my (rough) private work and out of ALSA project, but for future I wish to propose better implementation to ALSA project as a control server program. [1] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/sou… [2] https://github.com/takaswie/am-config-roms/ [3] https://github.com/cladisch/linux-firewire-utils [4] https://github.com/systemd/systemd/blob/master/rules/50-udev-default.rules.… [5] https://github.com/systemd/systemd/blob/master/src/login/70-uaccess.rules.m4 This rule refers to a rule optionally added by FFADO project. [6] This device has four units; IIDC, Audio, vendor-dependent and IRIS. [7] https://github.com/takaswie/libhinawa/ [8] https://github.com/takaswie/hinawa-utils/ Regards Takashi Sakamoto

1 0

[alsa-devel] [PATCH] ASoC: Mediatek: MT8183: enable IIR filter
by Shunli Wang 20 May '19

20 May '19

This patch is to enable IIR filter when DMIC of PMIC is enabled. Signed-off-by: Shunli Wang <shunli.wang(a)mediatek.com> --- sound/soc/mediatek/mt8183/mt8183-dai-adda.c | 14 +++++++++++--- 1 file changed, 11 insertions(+), 3 deletions(-) diff --git a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c index 017d7d1d9148..2b758a18c2ea 100644 --- a/sound/soc/mediatek/mt8183/mt8183-dai-adda.c +++ b/sound/soc/mediatek/mt8183/mt8183-dai-adda.c @@ -176,9 +176,6 @@ static int mtk_adda_ul_event(struct snd_soc_dapm_widget *w, case SND_SOC_DAPM_POST_PMD: /* should delayed 1/fs(smallest is 8k) = 125us before afe off */ usleep_range(125, 135); - - /* reset dmic */ - afe_priv->mtkaif_dmic = 0; break; default: break; @@ -426,6 +423,17 @@ static int mtk_dai_adda_hw_params(struct snd_pcm_substream *substream, ul_src_con0 |= (voice_mode << 17) & (0x7 << 17); + /* enable iir */ + ul_src_con0 |= (1 << UL_IIR_ON_TMP_CTL_SFT) & + UL_IIR_ON_TMP_CTL_MASK_SFT; + + /* 35Hz @ 48k */ + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_02_01, 0x00000000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_04_03, 0x00003FB8); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_06_05, 0x3FB80000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_08_07, 0x3FB80000); + regmap_write(afe->regmap, AFE_ADDA_IIR_COEF_10_09, 0x0000C048); + regmap_write(afe->regmap, AFE_ADDA_UL_SRC_CON0, ul_src_con0); /* mtkaif_rxif_data_mode = 0, amic */ -- 2.18.0

3 2

[alsa-devel] [PATCH] ALSA: dice: add stream format parameters for PreSonus FireStudio
by Takashi Sakamoto 20 May '19

20 May '19

FireStudio was launched by PreSonus 2009. This model consists of three ICs for its packet processing on IEEE 1394 bus: - Texus Instruments TSB41AB2 for physical layer of IEEE 1394 bus - WaveFront semiconductor, Dice II STD ASIC for link layer of IEEE 1394 bus and protocol layer - Xilinx Spartan XG3S500E FPGA for signal processing This model don't support TCAT extended application protocol. For such devices, ALSA dice driver needs to have hard-coded parameters for stream formats. This commit adds hard-coded table for this model. As a result, sampling transfer frequencies of 88.2/96.0 kHz are supported. I note that this patch can be backported to Linux kernel v4.18 and later. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 04042eda bus_info_length 4, crc_length 4, crc 11994 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 000a9204 company_id 000a92 | 410 023a8b7f device_id 04023a8b7f | EUI-64 000a9204023a8b7f root directory ----------------------------------------------------------------- 414 000661b6 directory_length 6, crc 25014 418 03000a92 vendor 41c 8100000a --> descriptor leaf at 444 420 17000008 model 424 8100000d --> descriptor leaf at 458 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 00041c75 directory_length 4, crc 7285 434 12000a92 specifier id 438 13000001 version 43c 17000008 model 440 8100000c --> descriptor leaf at 470 descriptor leaf at 444 ----------------------------------------------------------------- 444 00047c11 leaf_length 4, crc 31761 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 50726553 "PreS" 454 6f6e7573 "onus" descriptor leaf at 458 ----------------------------------------------------------------- 458 0005d7b3 leaf_length 5, crc 55219 45c 00000000 textual descriptor 460 00000000 minimal ASCII 464 46495245 "FIRE" 468 53545544 "STUD" 46c 494f0000 "IO" descriptor leaf at 470 ----------------------------------------------------------------- 470 0005d7b3 leaf_length 5, crc 55219 474 00000000 textual descriptor 478 00000000 minimal ASCII 47c 46495245 "FIRE" 480 53545544 "STUD" 484 494f0000 "IO" Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/dice/Makefile | 2 +- sound/firewire/dice/dice-presonus.c | 62 +++++++++++++++++++++++++++++ sound/firewire/dice/dice.c | 9 +++++ sound/firewire/dice/dice.h | 1 + 4 files changed, 73 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/dice/dice-presonus.c diff --git a/sound/firewire/dice/Makefile b/sound/firewire/dice/Makefile index 37062a233f6a..7efca3648c6a 100644 --- a/sound/firewire/dice/Makefile +++ b/sound/firewire/dice/Makefile @@ -1,4 +1,4 @@ snd-dice-objs := dice-transaction.o dice-stream.o dice-proc.o dice-midi.o \ dice-pcm.o dice-hwdep.o dice.o dice-tcelectronic.o \ - dice-alesis.o dice-extension.o dice-mytek.o + dice-alesis.o dice-extension.o dice-mytek.o dice-presonus.o obj-$(CONFIG_SND_DICE) += snd-dice.o diff --git a/sound/firewire/dice/dice-presonus.c b/sound/firewire/dice/dice-presonus.c new file mode 100644 index 000000000000..503f462a83f4 --- /dev/null +++ b/sound/firewire/dice/dice-presonus.c @@ -0,0 +1,62 @@ +// SPDX-License-Identifier: GPL-2.0 +// dice-presonus.c - a part of driver for DICE based devices +// +// Copyright (c) 2019 Takashi Sakamoto +// +// Licensed under the terms of the GNU General Public License, version 2. + +#include "dice.h" + +struct dice_presonus_spec { + unsigned int tx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + unsigned int rx_pcm_chs[MAX_STREAMS][SND_DICE_RATE_MODE_COUNT]; + bool has_midi; +}; + +static const struct dice_presonus_spec dice_presonus_firesutio = { + .tx_pcm_chs = {{16, 16, 0}, {10, 2, 0} }, + .rx_pcm_chs = {{16, 16, 0}, {10, 2, 0} }, + .has_midi = true, +}; + +int snd_dice_detect_presonus_formats(struct snd_dice *dice) +{ + static const struct { + u32 model_id; + const struct dice_presonus_spec *spec; + } *entry, entries[] = { + {0x000008, &dice_presonus_firesutio}, + }; + struct fw_csr_iterator it; + int key, val, model_id; + int i; + + model_id = 0; + fw_csr_iterator_init(&it, dice->unit->directory); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_MODEL) { + model_id = val; + break; + } + } + + for (i = 0; i < ARRAY_SIZE(entries); ++i) { + entry = entries + i; + if (entry->model_id == model_id) + break; + } + if (i == ARRAY_SIZE(entries)) + return -ENODEV; + + memcpy(dice->tx_pcm_chs, entry->spec->tx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + memcpy(dice->rx_pcm_chs, entry->spec->rx_pcm_chs, + MAX_STREAMS * SND_DICE_RATE_MODE_COUNT * sizeof(unsigned int)); + + if (entry->spec->has_midi) { + dice->tx_midi_ports[0] = 1; + dice->rx_midi_ports[0] = 1; + } + + return 0; +} diff --git a/sound/firewire/dice/dice.c b/sound/firewire/dice/dice.c index eee184b05d93..ac600e061d7b 100644 --- a/sound/firewire/dice/dice.c +++ b/sound/firewire/dice/dice.c @@ -19,6 +19,7 @@ MODULE_LICENSE("GPL v2"); #define OUI_MAUDIO 0x000d6c #define OUI_MYTEK 0x001ee8 #define OUI_SSL 0x0050c2 // Actually ID reserved by IEEE. +#define OUI_PRESONUS 0x000a92 #define DICE_CATEGORY_ID 0x04 #define WEISS_CATEGORY_ID 0x00 @@ -371,6 +372,14 @@ static const struct ieee1394_device_id dice_id_table[] = { .vendor_id = OUI_SSL, .model_id = 0x000070, }, + // Presonus FireStudio. + { + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID, + .vendor_id = OUI_PRESONUS, + .model_id = 0x000008, + .driver_data = (kernel_ulong_t)snd_dice_detect_presonus_formats, + }, { .match_flags = IEEE1394_MATCH_VERSION, .version = DICE_INTERFACE, diff --git a/sound/firewire/dice/dice.h b/sound/firewire/dice/dice.h index 83353a3559e8..9699adc2a96d 100644 --- a/sound/firewire/dice/dice.h +++ b/sound/firewire/dice/dice.h @@ -227,5 +227,6 @@ int snd_dice_detect_tcelectronic_formats(struct snd_dice *dice); int snd_dice_detect_alesis_formats(struct snd_dice *dice); int snd_dice_detect_extension_formats(struct snd_dice *dice); int snd_dice_detect_mytek_formats(struct snd_dice *dice); +int snd_dice_detect_presonus_formats(struct snd_dice *dice); #endif -- 2.20.1

2 1

[alsa-devel] [asoc:for-5.3 34/45] sound/soc/mediatek/common/mtk-afe-fe-dai.c:268:2: error: too few arguments to function 'mtk_regmap_update_bits'
by kbuild test robot 20 May '19

20 May '19

Hi Shunli, FYI, the error/warning still remains. tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 head: bfa8130f50a63563eae10ef933fe01b50b3e87a0 commit: 1628fc3f47717fc3c8d9cd5fd6c7bc7d3796927d [34/45] ASoC: Mediatek: add memory interface data align config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 1628fc3f47717fc3c8d9cd5fd6c7bc7d3796927d # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): sound/soc/mediatek/common/mtk-afe-fe-dai.c: In function 'mtk_afe_fe_prepare': >> sound/soc/mediatek/common/mtk-afe-fe-dai.c:268:2: error: too few arguments to function 'mtk_regmap_update_bits' mtk_regmap_update_bits(afe->regmap, memif->data->hd_align_reg, ^~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/common/mtk-afe-fe-dai.c:19:12: note: declared here static int mtk_regmap_update_bits(struct regmap *map, int reg, ^~~~~~~~~~~~~~~~~~~~~~ vim +/mtk_regmap_update_bits +268 sound/soc/mediatek/common/mtk-afe-fe-dai.c 236 237 int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, 238 struct snd_soc_dai *dai) 239 { 240 struct snd_soc_pcm_runtime *rtd = substream->private_data; 241 struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); 242 struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; 243 int hd_audio = 0; 244 int hd_align = 1; 245 246 /* set hd mode */ 247 switch (substream->runtime->format) { 248 case SNDRV_PCM_FORMAT_S16_LE: 249 hd_audio = 0; 250 break; 251 case SNDRV_PCM_FORMAT_S32_LE: 252 hd_audio = 1; 253 hd_align = 1; 254 break; 255 case SNDRV_PCM_FORMAT_S24_LE: 256 hd_audio = 1; 257 hd_align = 0; 258 break; 259 default: 260 dev_err(afe->dev, "%s() error: unsupported format %d\n", 261 __func__, substream->runtime->format); 262 break; 263 } 264 265 mtk_regmap_update_bits(afe->regmap, memif->data->hd_reg, 266 1, hd_audio, memif->data->hd_shift); 267 > 268 mtk_regmap_update_bits(afe->regmap, memif->data->hd_align_reg, 269 memif->data->hd_align_mshift, 270 hd_align ? memif->data->hd_align_mshift : 0); 271 272 return 0; 273 } 274 EXPORT_SYMBOL_GPL(mtk_afe_fe_prepare); 275 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH][RFC] ASoC: soc-pcm: fixup try_module_get() calling timing
by Kuninori Morimoto 20 May '19

20 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_pcm_components_open() try to call try_module_get() based on component->driver->module_get_upon_open. But, it should be always called, not relatead to .open callback. It should be called at (A) istead of (B) => (A) if (!component->driver->ops || !component->driver->ops->open) continue; => (B) if (component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) { ... } ret = component->driver->ops->open(substream); Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Pierre-Louis, Vinod, Liam I think this patch is correct, but I'm not sure. I'm happy if someone can confirm it. sound/soc/soc-pcm.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7737e00..7b4cda6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -440,10 +440,6 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, component = rtdcom->component; *last = component; - if (!component->driver->ops || - !component->driver->ops->open) - continue; - if (component->driver->module_get_upon_open && !try_module_get(component->dev->driver->owner)) { dev_err(component->dev, @@ -452,6 +448,10 @@ static int soc_pcm_components_open(struct snd_pcm_substream *substream, return -ENODEV; } + if (!component->driver->ops || + !component->driver->ops->open) + continue; + ret = component->driver->ops->open(substream); if (ret < 0) { dev_err(component->dev, -- 2.7.4

3 4

[alsa-devel] [PATCH][RFC] ASoC: soc-pcm: fixup module_put() calling timing
by Kuninori Morimoto 18 May '19

18 May '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> soc_pcm_components_close() try to call module_put() based on component->driver->module_get_upon_open. But, it should be called even though it doesn't have .close callback. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Mark, Pierre-Louis, Vinod, Liam I think this patch is correct, but I'm not sure. I'm happy if someone can confirm it. sound/soc/soc-pcm.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 7b4cda6..e24eab3 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -477,11 +477,9 @@ static int soc_pcm_components_close(struct snd_pcm_substream *substream, if (component == last) break; - if (!component->driver->ops || - !component->driver->ops->close) - continue; - - component->driver->ops->close(substream); + if (component->driver->ops && + component->driver->ops->close) + component->driver->ops->close(substream); if (component->driver->module_get_upon_open) module_put(component->dev->driver->owner); -- 2.7.4

2 1

[alsa-devel] [PATCH] sound: fix typos in code comments
by Weitao Hou 18 May '19

18 May '19

fix lenght to length Signed-off-by: Weitao Hou <houweitaoo(a)gmail.com> --- sound/soc/qcom/qdsp6/q6asm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/qdsp6/q6asm.c b/sound/soc/qcom/qdsp6/q6asm.c index 4f85cb19a309..e8141a33a55e 100644 --- a/sound/soc/qcom/qdsp6/q6asm.c +++ b/sound/soc/qcom/qdsp6/q6asm.c @@ -1194,7 +1194,7 @@ EXPORT_SYMBOL_GPL(q6asm_open_read); * q6asm_write_async() - non blocking write * * @ac: audio client pointer - * @len: lenght in bytes + * @len: length in bytes * @msw_ts: timestamp msw * @lsw_ts: timestamp lsw * @wflags: flags associated with write -- 2.18.0

1 0

[alsa-devel] [PATCH] ASoC: fsl_esai: fix the channel swap issue after xrun
by S.j. Wang 17 May '19

17 May '19

There is chip errata ERR008000, the reference doc is (https://www.nxp.com/docs/en/errata/IMX6DQCE.pdf), The issue is "While using ESAI transmit or receive and an underrun/overrun happens, channel swap may occur. The only recovery mechanism is to reset the ESAI." In this commit add a tasklet to handle reset of ESAI after xrun happens Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_esai.c | 166 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 166 insertions(+) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index 10d2210c91ef..149972894c95 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -52,17 +52,20 @@ struct fsl_esai { struct clk *extalclk; struct clk *fsysclk; struct clk *spbaclk; + struct tasklet_struct task; u32 fifo_depth; u32 slot_width; u32 slots; u32 tx_mask; u32 rx_mask; + u32 tx_channels; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; bool sck_div[2]; bool slave_mode; bool synchronous; + bool reset_at_xrun; char name[32]; }; @@ -71,8 +74,14 @@ static irqreturn_t esai_isr(int irq, void *devid) struct fsl_esai *esai_priv = (struct fsl_esai *)devid; struct platform_device *pdev = esai_priv->pdev; u32 esr; + u32 saisr; regmap_read(esai_priv->regmap, REG_ESAI_ESR, &esr); + regmap_read(esai_priv->regmap, REG_ESAI_SAISR, &saisr); + + if ((saisr & (ESAI_SAISR_TUE | ESAI_SAISR_ROE)) + && esai_priv->reset_at_xrun) + tasklet_schedule(&esai_priv->task); if (esr & ESAI_ESR_TINIT_MASK) dev_dbg(&pdev->dev, "isr: Transmission Initialized\n"); @@ -552,6 +561,9 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); u32 mask; + if (tx) + esai_priv->tx_channels = channels; + switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: @@ -585,10 +597,16 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + ESAI_xCR_xEIE_MASK, ESAI_xCR_xEIE); + break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), + ESAI_xCR_xEIE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), @@ -618,6 +636,145 @@ static const struct snd_soc_dai_ops fsl_esai_dai_ops = { .set_tdm_slot = fsl_esai_set_dai_tdm_slot, }; +static void fsl_esai_reset(unsigned long arg) +{ + struct fsl_esai *esai_priv = (struct fsl_esai *)arg; + u32 saisr; + u32 tsma, tsmb, rsma, rsmb, tcr, rcr, tfcr, rfcr; + int i; + + /* + * stop the tx & rx + */ + regmap_read(esai_priv->regmap, REG_ESAI_TSMA, &tsma); + regmap_read(esai_priv->regmap, REG_ESAI_TSMB, &tsmb); + regmap_read(esai_priv->regmap, REG_ESAI_RSMA, &rsma); + regmap_read(esai_priv->regmap, REG_ESAI_RSMB, &rsmb); + + regmap_read(esai_priv->regmap, REG_ESAI_TCR, &tcr); + regmap_read(esai_priv->regmap, REG_ESAI_RCR, &rcr); + + regmap_read(esai_priv->regmap, REG_ESAI_TFCR, &tfcr); + regmap_read(esai_priv->regmap, REG_ESAI_RFCR, &rfcr); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xEIE_MASK | ESAI_xCR_TE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xEIE_MASK | ESAI_xCR_RE_MASK, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMB_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMB_xS_MASK, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR | ESAI_xFCR_xFEN, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR, 0); + + /* + * reset the esai, and restore the registers + */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + ESAI_ECR_ESAIEN_MASK | ESAI_ECR_ERST_MASK, + ESAI_ECR_ESAIEN | ESAI_ECR_ERST); + regmap_update_bits(esai_priv->regmap, REG_ESAI_ECR, + ESAI_ECR_ESAIEN_MASK | ESAI_ECR_ERST_MASK, + ESAI_ECR_ESAIEN); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xPR_MASK, + ESAI_xCR_xPR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xPR_MASK, + ESAI_xCR_xPR); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, 0); + + /* + * Add fifo reset here, because the regcache_sync will + * write one more data to ETDR. + * Which will cause channel shift. + */ + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR_MASK, ESAI_xFCR_xFR); + + regcache_mark_dirty(esai_priv->regmap); + regcache_sync(esai_priv->regmap); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFR_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFR_MASK, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xPR_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xPR_MASK, 0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_PRRC, + ESAI_PRRC_PDC_MASK, + ESAI_PRRC_PDC(ESAI_GPIO)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_PCRC, + ESAI_PCRC_PC_MASK, + ESAI_PCRC_PC(ESAI_GPIO)); + + regmap_read(esai_priv->regmap, REG_ESAI_SAISR, &saisr); + + /* + * restart tx / rx, if they already enabled + */ + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TFCR, + ESAI_xFCR_xFEN_MASK, tfcr & ESAI_xFCR_xFEN); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_RFCR, + ESAI_xFCR_xFEN_MASK, rfcr & ESAI_xFCR_xFEN); + + /* Write initial words reqiured by ESAI as normal procedure */ + for (i = 0; i < esai_priv->tx_channels; i++) + regmap_write(esai_priv->regmap, REG_ESAI_ETDR, 0x0); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_TE_MASK, + ESAI_xCR_TE_MASK & tcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_RE_MASK, + ESAI_xCR_RE_MASK & rcr); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, + ESAI_xSMB_xS_MASK, + ESAI_xSMB_xS_MASK & tsmb); + regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, + ESAI_xSMA_xS_MASK, + ESAI_xSMA_xS_MASK & tsma); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, + ESAI_xSMB_xS_MASK, + ESAI_xSMB_xS_MASK & rsmb); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, + ESAI_xSMA_xS_MASK, + ESAI_xSMA_xS_MASK & rsma); + + regmap_update_bits(esai_priv->regmap, REG_ESAI_TCR, + ESAI_xCR_xEIE_MASK, ESAI_xCR_xEIE & tcr); + regmap_update_bits(esai_priv->regmap, REG_ESAI_RCR, + ESAI_xCR_xEIE_MASK, ESAI_xCR_xEIE & rcr); +} + static int fsl_esai_dai_probe(struct snd_soc_dai *dai) { struct fsl_esai *esai_priv = snd_soc_dai_get_drvdata(dai); @@ -787,6 +944,10 @@ static int fsl_esai_probe(struct platform_device *pdev) esai_priv->pdev = pdev; snprintf(esai_priv->name, sizeof(esai_priv->name), "%pOFn", np); + if (of_device_is_compatible(np, "fsl,vf610-esai") || + of_device_is_compatible(np, "fsl,imx35-esai")) + esai_priv->reset_at_xrun = true; + /* Get the addresses and IRQ */ res = platform_get_resource(pdev, IORESOURCE_MEM, 0); regs = devm_ioremap_resource(&pdev->dev, res); @@ -899,6 +1060,8 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + tasklet_init(&esai_priv->task, fsl_esai_reset, (unsigned long)esai_priv); + pm_runtime_enable(&pdev->dev); regcache_cache_only(esai_priv->regmap, true); @@ -912,7 +1075,10 @@ static int fsl_esai_probe(struct platform_device *pdev) static int fsl_esai_remove(struct platform_device *pdev) { + struct fsl_esai *esai_priv = platform_get_drvdata(pdev); + pm_runtime_disable(&pdev->dev); + tasklet_kill(&esai_priv->task); return 0; } -- 2.21.0

2 1

[alsa-devel] [PATCH] sound: open brace should be on the previous line
by parna.naveenkumar＠gmail.com 17 May '19

17 May '19

From: Naveen Kumar Parna <parna.naveenkumar(a)gmail.com> Resolved open brace { should be on the previous line checkpatch.pl error. While addressing this error, also corrected the affected code for below mentioned checkpatch errors. ERROR: spaces required around that '<' (ctx:VxV) ERROR: spaces required around that '==' (ctx:VxV) ERROR: space required before the open parenthesis '(' Signed-off-by: Naveen Kumar Parna <parna.naveenkumar(a)gmail.com> --- sound/sound_core.c | 21 +++++++-------------- 1 file changed, 7 insertions(+), 14 deletions(-) diff --git a/sound/sound_core.c b/sound/sound_core.c index 40ad000c2e3c..ce794a2afc6b 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -111,8 +111,7 @@ module_exit(cleanup_soundcore); #define SOUND_STEP 16 -struct sound_unit -{ +struct sound_unit { int unit_minor; const struct file_operations *unit_fops; struct sound_unit *next; @@ -151,8 +150,7 @@ module_param(preclaim_oss, int, 0444); static int soundcore_open(struct inode *, struct file *); -static const struct file_operations soundcore_fops = -{ +static const struct file_operations soundcore_fops = { /* We must have an owner or the module locking fails */ .owner = THIS_MODULE, .open = soundcore_open, @@ -173,8 +171,7 @@ static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, while (*list && (*list)->unit_minor<n) list=&((*list)->next); - while(n<top) - { + while (n < top) { /* Found a hole ? */ if(*list==NULL || (*list)->unit_minor>n) break; @@ -219,11 +216,9 @@ static int __sound_insert_unit(struct sound_unit * s, struct sound_unit **list, static struct sound_unit *__sound_remove_unit(struct sound_unit **list, int unit) { - while(*list) - { + while (*list) { struct sound_unit *p=*list; - if(p->unit_minor==unit) - { + if (p->unit_minor == unit) { *list=p->next; return p; } @@ -528,8 +523,7 @@ static struct sound_unit *__look_for_unit(int chain, int unit) struct sound_unit *s; s=chains[chain]; - while(s && s->unit_minor <= unit) - { + while (s && s->unit_minor <= unit) { if(s->unit_minor==unit) return s; s=s->next; @@ -545,8 +539,7 @@ static int soundcore_open(struct inode *inode, struct file *file) const struct file_operations *new_fops = NULL; chain=unit&0x0F; - if(chain==4 || chain==5) /* dsp/audio/dsp16 */ - { + if (chain == 4 || chain == 5) { /* dsp/audio/dsp16 */ unit&=0xF0; unit|=3; chain=3; -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: cs42xx8: Add regcache mask dirty
by S.j. Wang 17 May '19

17 May '19

Add regcache_mark_dirty before regcache_sync for power of codec may be lost at suspend, then all the register need to be reconfigured. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/codecs/cs42xx8.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..28a4ac36c4f8 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -558,6 +558,7 @@ static int cs42xx8_runtime_resume(struct device *dev) msleep(5); regcache_cache_only(cs42xx8->regmap, false); + regcache_mark_dirty(cs42xx8->regmap); ret = regcache_sync(cs42xx8->regmap); if (ret) { -- 2.21.0

2 1

[alsa-devel] [PATCH V2] ASoC: cs42xx8: Add reset gpio handling
by S.j. Wang 17 May '19

17 May '19

Handle the reset GPIO and reset the device every time we start it. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio sound/soc/codecs/cs42xx8.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 28a4ac36c4f8..b377cddaf2e6 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -14,6 +14,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/of_device.h> +#include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ struct cs42xx8_priv { bool slave_mode; unsigned long sysclk; u32 tx_channels; + struct gpio_desc *gpiod_reset; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -467,6 +469,13 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) return -EINVAL; } + cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(cs42xx8->gpiod_reset)) + return PTR_ERR(cs42xx8->gpiod_reset); + + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); + cs42xx8->clk = devm_clk_get(dev, "mclk"); if (IS_ERR(cs42xx8->clk)) { dev_err(dev, "failed to get the clock: %ld\n", @@ -547,6 +556,8 @@ static int cs42xx8_runtime_resume(struct device *dev) return ret; } + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); if (ret) { @@ -586,6 +597,8 @@ static int cs42xx8_runtime_suspend(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 1); + clk_disable_unprepare(cs42xx8->clk); return 0; -- 2.21.0

2 2

[alsa-devel] [PATCH v4] ASoC: tlv320aic3x: Add support for high power analog output
by Saravanan Sekar 17 May '19

17 May '19

Add support to output level control for the analog high power output drivers HPOUT and HPCOM. Signed-off-by: Saravanan Sekar <sravanhome(a)gmail.com> --- Notes: Notes: Changes in V4: -Added separate mono playback volume control -grouped volume controls with corresponding switch Changes in V3: -Fixed compilation error Changes in V2: - Removed power control as it is handled by DAPM - Added level control for left channel sound/soc/codecs/tlv320aic3x.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 516d17cb2182..599e4ed3850b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -324,6 +324,9 @@ static DECLARE_TLV_DB_SCALE(adc_tlv, 0, 50, 0); */ static DECLARE_TLV_DB_SCALE(output_stage_tlv, -5900, 50, 1); +/* Output volumes. From 0 to 9 dB in 1 dB steps */ +static const DECLARE_TLV_DB_SCALE(out_tlv, 0, 100, 0); + static const struct snd_kcontrol_new aic3x_snd_controls[] = { /* Output */ SOC_DOUBLE_R_TLV("PCM Playback Volume", @@ -386,11 +389,17 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, 0, 118, 1, output_stage_tlv), - /* Output pin mute controls */ + /* Output pin controls */ + SOC_DOUBLE_R_TLV("Line Playback Volume", LLOPM_CTRL, RLOPM_CTRL, 4, + 9, 0, out_tlv), SOC_DOUBLE_R("Line Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R_TLV("HP Playback Volume", HPLOUT_CTRL, HPROUT_CTRL, 4, + 9, 0, out_tlv), SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R_TLV("HPCOM Playback Volume", HPLCOM_CTRL, HPRCOM_CTRL, + 4, 9, 0, out_tlv), SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), @@ -472,6 +481,9 @@ static const struct snd_kcontrol_new aic3x_mono_controls[] = { 0, 118, 1, output_stage_tlv), SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE_TLV("Mono Playback Volume", MONOLOPM_CTRL, 4, 9, 0, + out_tlv), + }; /* -- 2.17.1

3 2

[alsa-devel] [PATCH V2] ASoC: cs42xx8: Add regcache mask dirty
by S.j. Wang 17 May '19

17 May '19

Add regcache_mark_dirty before regcache_sync for power of codec may be lost at suspend, then all the register need to be reconfigured. Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for CS42448/CS42888") Cc: <stable(a)vger.kernel.org> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changs in V2 - Don't split Fixes tag. sound/soc/codecs/cs42xx8.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index ebb9e0cf8364..28a4ac36c4f8 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -558,6 +558,7 @@ static int cs42xx8_runtime_resume(struct device *dev) msleep(5); regcache_cache_only(cs42xx8->regmap, false); + regcache_mark_dirty(cs42xx8->regmap); ret = regcache_sync(cs42xx8->regmap); if (ret) { -- 2.21.0

1 0

Re: [alsa-devel] [PATCH] ASoC: cs42xx8: Add regcache mask dirty
by S.j. Wang 17 May '19

17 May '19

Hi > Add regcache_mark_dirty before regcache_sync for power of codec may be > lost at suspend, then all the register need to be reconfigured. > > Fixes: 0c516b4ff85c ("ASoC: cs42xx8: Add codec driver support for > CS42448/CS42888") The Fixes tag is split, will send v2. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH] ASoC: sun4i-codec: fix first delay on Speaker
by Georgii Staroselskii 17 May '19

17 May '19

This commit fixes the same issue as bf14da7 but for another codepath. The issue can be triggered by having Speaker in codec audio routing. &codec { allwinner,pa-gpios = <&r_pio 0 11 GPIO_ACTIVE_HIGH>; /* PL11 */ allwinner,audio-routing = "Speaker", "LINEOUT"; status = "okay"; } Signed-off-by: Georgii Staroselskii <georgii.staroselskii(a)emlid.com> --- sound/soc/sunxi/sun4i-codec.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c index 15d08e3..e0099519 100644 --- a/sound/soc/sunxi/sun4i-codec.c +++ b/sound/soc/sunxi/sun4i-codec.c @@ -1329,6 +1329,15 @@ static int sun4i_codec_spk_event(struct snd_soc_dapm_widget *w, gpiod_set_value_cansleep(scodec->gpio_pa, !!SND_SOC_DAPM_EVENT_ON(event)); + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* + * Need a delay to have the amplifier up. 700ms seems the best + * compromise between the time to let the amplifier up and the + * time not to feel this delay while playing a sound. + */ + msleep(700); + } + return 0; } -- 2.7.4

3 4

[alsa-devel] [PATCH v3 1/2] dt-bindings: sound: Convert Allwinner SPDIF binding to YAML
by Maxime Ripard 16 May '19

16 May '19

The Allwinner SoCs feature an SPDIF controller across multiple SoC generations. However, earlier generations were a bit simpler than the subsequent ones, and for example would always have RX and TX capabilities, and no reset lines. In order to express this, let's create two YAML schemas instead of the free form text we had before. Signed-off-by: Maxime Ripard <maxime.ripard(a)bootlin.com> --- Changes from v2: - Add comments Changes from v1: - Merged the two schemas together and used the draft-7 conditionals --- Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml | 101 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++- Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 42 +------------------------------ 2 files changed, 101 insertions(+), 42 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml delete mode 100644 Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt diff --git a/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml new file mode 100644 index 000000000000..5d72d48e923e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/allwinner,sun4i-a10-spdif.yaml @@ -0,0 +1,101 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/allwinner,sun4i-a10-spdif.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Allwinner A10 S/PDIF Controller Device Tree Bindings + +maintainers: + - Chen-Yu Tsai <wens(a)csie.org> + - Liam Girdwood <lgirdwood(a)gmail.com> + - Mark Brown <broonie(a)kernel.org> + - Maxime Ripard <maxime.ripard(a)bootlin.com> + +properties: + "#sound-dai-cells": + const: 0 + + compatible: + oneOf: + - const: allwinner,sun4i-a10-spdif + - const: allwinner,sun6i-a31-spdif + - const: allwinner,sun8i-h3-spdif + - items: + - const: allwinner,sun8i-a83t-spdif + - const: allwinner,sun8i-h3-spdif + - items: + - const: allwinner,sun50i-a64-spdif + - const: allwinner,sun8i-h3-spdif + + reg: + maxItems: 1 + + interrupts: + maxItems: 1 + + clocks: + items: + - description: Bus Clock + - description: Module Clock + + clock-names: + items: + - const: apb + - const: spdif + + dmas: + items: + - description: RX DMA Channel + - description: TX DMA Channel + + dma-names: + items: + - const: rx + - const: tx + + # Even though it only applies to subschemas under the conditionals, + # not listing them here will trigger a warning because of the + # additionalsProperties set to false. + resets: + maxItems: 1 + +allOf: + - if: + properties: + compatible: + contains: + enum: + - allwinner,sun6i-a31-spdif + - allwinner,sun8i-h3-spdif + + then: + required: + - resets + +required: + - "#sound-dai-cells" + - compatible + - reg + - interrupts + - clocks + - clock-names + - dmas + - dma-names + +additionalProperties: false + +examples: + - | + spdif: spdif@1c21000 { + #sound-dai-cells = <0>; + compatible = "allwinner,sun4i-a10-spdif"; + reg = <0x01c21000 0x40>; + interrupts = <13>; + clocks = <&apb0_gates 1>, <&spdif_clk>; + clock-names = "apb", "spdif"; + dmas = <&dma 0 2>, <&dma 0 2>; + dma-names = "rx", "tx"; + }; + +... diff --git a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt b/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt deleted file mode 100644 index 0c64a209c2e9..000000000000 --- a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt +++ /dev/null @@ -1,42 +0,0 @@ -Allwinner Sony/Philips Digital Interface Format (S/PDIF) Controller - -The Allwinner S/PDIF audio block is a transceiver that allows the -processor to receive and transmit digital audio via an coaxial cable or -a fibre cable. -For now only playback is supported. - -Required properties: - - - compatible : should be one of the following: - - "allwinner,sun4i-a10-spdif": for the Allwinner A10 SoC - - "allwinner,sun6i-a31-spdif": for the Allwinner A31 SoC - - "allwinner,sun8i-h3-spdif": for the Allwinner H3 SoC - - - reg : Offset and length of the register set for the device. - - - interrupts : Contains the spdif interrupt. - - - dmas : Generic dma devicetree binding as described in - Documentation/devicetree/bindings/dma/dma.txt. - - - dma-names : Two dmas have to be defined, "tx" and "rx". - - - clocks : Contains an entry for each entry in clock-names. - - - clock-names : Includes the following entries: - "apb" clock for the spdif bus. - "spdif" clock for spdif controller. - - - resets : reset specifier for the ahb reset (A31 and newer only) - -Example: - -spdif: spdif@1c21000 { - compatible = "allwinner,sun4i-a10-spdif"; - reg = <0x01c21000 0x40>; - interrupts = <13>; - clocks = <&apb0_gates 1>, <&spdif_clk>; - clock-names = "apb", "spdif"; - dmas = <&dma 0 2>, <&dma 0 2>; - dma-names = "rx", "tx"; -}; base-commit: 00c3bc00f0cadbb48f2414370733892676511e79 -- git-series 0.9.1

3 5

[alsa-devel] [PATCH] ASoC: AK4458: add regulator for ak4458
by Viorel Suman 16 May '19

16 May '19

From: Shengjiu Wang <shengjiu.wang(a)nxp.com> Add regulator for ak4458. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> Signed-off-by: Viorel Suman <viorel.suman(a)nxp.com> --- sound/soc/codecs/ak4458.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ak4458.c b/sound/soc/codecs/ak4458.c index 7156215..06dcf13 100644 --- a/sound/soc/codecs/ak4458.c +++ b/sound/soc/codecs/ak4458.c @@ -12,6 +12,7 @@ #include <linux/of_device.h> #include <linux/of_gpio.h> #include <linux/pm_runtime.h> +#include <linux/regulator/consumer.h> #include <linux/slab.h> #include <sound/initval.h> #include <sound/pcm_params.h> @@ -21,6 +22,12 @@ #include "ak4458.h" +#define AK4458_NUM_SUPPLIES 2 +static const char *ak4458_supply_names[AK4458_NUM_SUPPLIES] = { + "DVDD", + "AVDD", +}; + struct ak4458_drvdata { struct snd_soc_dai_driver *dai_drv; const struct snd_soc_component_driver *comp_drv; @@ -37,6 +44,7 @@ struct ak4458_priv { int fmt; int slots; int slot_width; + struct regulator_bulk_data supplies[AK4458_NUM_SUPPLIES]; }; static const struct reg_default ak4458_reg_defaults[] = { @@ -666,7 +674,7 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) { struct ak4458_priv *ak4458; const struct ak4458_drvdata *drvdata; - int ret; + int ret, i; ak4458 = devm_kzalloc(&i2c->dev, sizeof(*ak4458), GFP_KERNEL); if (!ak4458) @@ -691,6 +699,23 @@ static int ak4458_i2c_probe(struct i2c_client *i2c) if (IS_ERR(ak4458->mute_gpiod)) return PTR_ERR(ak4458->mute_gpiod); + for (i = 0; i < ARRAY_SIZE(ak4458->supplies); i++) + ak4458->supplies[i].supply = ak4458_supply_names[i]; + + ret = devm_regulator_bulk_get(ak4458->dev, ARRAY_SIZE(ak4458->supplies), + ak4458->supplies); + if (ret != 0) { + dev_err(ak4458->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(ak4458->supplies), + ak4458->supplies); + if (ret != 0) { + dev_err(ak4458->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + ret = devm_snd_soc_register_component(ak4458->dev, drvdata->comp_drv, drvdata->dai_drv, 1); if (ret < 0) { -- 2.7.4

4 4

Re: [alsa-devel] [PATCH] ASoC: rt1308: Add RT1308 amplifier driver
by Mark Brown 16 May '19

16 May '19

On Wed, Apr 17, 2019 at 11:40:28AM +0000, Derek [方德義] wrote: Please fix your mail client to word wrap within paragraphs at something substantially less than 80 columns. Doing this makes your messages much easier to read and reply to. > > The ones here with comments look an awful lot like they're system specific config > > which should be left up to either platform data (or DT/ACPI configuration) or > > done at runtime (like gains). Some of the others like the pads configuration > > might also fit into that. > We don't open the DA gain setting for user at runtime, so have to set > a default level in the init list. That's not the way Linux drivers generally work... is there some technical reason for that or is it just some kind of policy decision? > > > +static void rt1308_reset(struct regmap *regmap) { > > > + regmap_write(regmap, RT1308_RESET, 0); } > > > > If we do need the init list shouldn't we also do a reninit every time we do a reset? > The init list is only set at driver probe, not every time after resetting. This is what I'm pointing out; if you don't restore these values then why set them at all?

2 10

[alsa-devel] [asoc:for-5.3 34/40] sound/soc/mediatek/common/mtk-afe-fe-dai.c:268:2: error: too few arguments to function 'mtk_regmap_update_bits'
by kbuild test robot 16 May '19

16 May '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.3 head: c8609f3870f7078fc7922eb816ed4908a9bd44f3 commit: 1628fc3f47717fc3c8d9cd5fd6c7bc7d3796927d [34/40] ASoC: Mediatek: add memory interface data align config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout 1628fc3f47717fc3c8d9cd5fd6c7bc7d3796927d # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm If you fix the issue, kindly add following tag Reported-by: kbuild test robot <lkp(a)intel.com> All errors (new ones prefixed by >>): sound/soc/mediatek/common/mtk-afe-fe-dai.c: In function 'mtk_afe_fe_prepare': >> sound/soc/mediatek/common/mtk-afe-fe-dai.c:268:2: error: too few arguments to function 'mtk_regmap_update_bits' mtk_regmap_update_bits(afe->regmap, memif->data->hd_align_reg, ^~~~~~~~~~~~~~~~~~~~~~ sound/soc/mediatek/common/mtk-afe-fe-dai.c:19:12: note: declared here static int mtk_regmap_update_bits(struct regmap *map, int reg, ^~~~~~~~~~~~~~~~~~~~~~ vim +/mtk_regmap_update_bits +268 sound/soc/mediatek/common/mtk-afe-fe-dai.c 236 237 int mtk_afe_fe_prepare(struct snd_pcm_substream *substream, 238 struct snd_soc_dai *dai) 239 { 240 struct snd_soc_pcm_runtime *rtd = substream->private_data; 241 struct mtk_base_afe *afe = snd_soc_dai_get_drvdata(dai); 242 struct mtk_base_afe_memif *memif = &afe->memif[rtd->cpu_dai->id]; 243 int hd_audio = 0; 244 int hd_align = 1; 245 246 /* set hd mode */ 247 switch (substream->runtime->format) { 248 case SNDRV_PCM_FORMAT_S16_LE: 249 hd_audio = 0; 250 break; 251 case SNDRV_PCM_FORMAT_S32_LE: 252 hd_audio = 1; 253 hd_align = 1; 254 break; 255 case SNDRV_PCM_FORMAT_S24_LE: 256 hd_audio = 1; 257 hd_align = 0; 258 break; 259 default: 260 dev_err(afe->dev, "%s() error: unsupported format %d\n", 261 __func__, substream->runtime->format); 262 break; 263 } 264 265 mtk_regmap_update_bits(afe->regmap, memif->data->hd_reg, 266 1, hd_audio, memif->data->hd_shift); 267 > 268 mtk_regmap_update_bits(afe->regmap, memif->data->hd_align_reg, 269 memif->data->hd_align_mshift, 270 hd_align ? memif->data->hd_align_mshift : 0); 271 272 return 0; 273 } 274 EXPORT_SYMBOL_GPL(mtk_afe_fe_prepare); 275 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

Re: [alsa-devel] [PATCH V4] ASoC: cs42xx8: add reset-gpios in binding document
by S.j. Wang 16 May '19

16 May '19

Hi > On Thu, May 16, 2019 at 8:36 AM S.j. Wang <shengjiu.wang(a)nxp.com> > wrote: > > > cs42888: codec@48 { > > @@ -25,4 +30,5 @@ cs42888: codec@48 { > > VD-supply = <&reg_audio>; > > VLS-supply = <&reg_audio>; > > VLC-supply = <&reg_audio>; > > + reset-gpios = <&pca9557_b 1 1>; > > reset-gpios = <&pca9557_b 1 GPIO_ACTIVE_LOW>; please Ok, sent v5. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH V4] ASoC: cs42xx8: add reset-gpios in binding document
by S.j. Wang 16 May '19

16 May '19

Add reset-gpios property, which is optional. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in V4 - use gpios instead of gpio Changes in RESEND v3 - send updated binding document only Changes in v3 - update binding document. Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio Documentation/devicetree/bindings/sound/cs42xx8.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt index 8619a156d038..0138ef559cc4 100644 --- a/Documentation/devicetree/bindings/sound/cs42xx8.txt +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -14,6 +14,11 @@ Required properties: - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt +Optional properties: + + - reset-gpios : a GPIO spec to define which pin is connected to the chip's + !RESET pin + Example: cs42888: codec@48 { @@ -25,4 +30,5 @@ cs42888: codec@48 { VD-supply = <&reg_audio>; VLS-supply = <&reg_audio>; VLC-supply = <&reg_audio>; + reset-gpios = <&pca9557_b 1 1>; }; -- 2.21.0

2 1

[alsa-devel] [PATCH RESEND V3] ASoC: cs42xx8: add reset-gpio in binding document
by S.j. Wang 16 May '19

16 May '19

Add reset-gpio property, which is an optional option Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in RESEND v3 - send updated binding document only Changes in v3 - update binding document. Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio Documentation/devicetree/bindings/sound/cs42xx8.txt | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt index 8619a156d038..ab8f54095269 100644 --- a/Documentation/devicetree/bindings/sound/cs42xx8.txt +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -14,6 +14,11 @@ Required properties: - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt +Optional properties: + + - reset-gpio : a GPIO spec to define which pin is connected to the chip's + !RESET pin + Example: cs42888: codec@48 { @@ -25,4 +30,5 @@ cs42888: codec@48 { VD-supply = <&reg_audio>; VLS-supply = <&reg_audio>; VLC-supply = <&reg_audio>; + reset-gpio = <&pca9557_b 1 1>; }; -- 2.21.0

3 3

Re: [alsa-devel] [PATCH V2] ASoC: cs42xx8: Add reset gpio handling
by S.j. Wang 16 May '19

16 May '19

Hi > > On Thu, May 16, 2019 at 06:04:58AM +0000, S.j. Wang wrote: > > > + cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset", > > + GPIOD_OUT_HIGH); > > + if (IS_ERR(cs42xx8->gpiod_reset)) > > + return PTR_ERR(cs42xx8->gpiod_reset); > > You also need a binding document update for this. ok, will send v3 best regards wang shengjiu

2 2

[alsa-devel] [PATCH 0/5] ASoC: meson: add hdmitx glue support
by Jerome Brunet 16 May '19

16 May '19

On the Amlogic SoC, there is a glue between the SoC audio outputs and the input of the embedded Synopsys HDMI controller. On the g12a, this glue is mostly a couple of muxes to select the i2s and spdif inputs of the hdmi controller. Each of these inputs may have different hw_params and fmt which makes our life a little bit more interesting, especially when switching between to active inputs. This glue is modeled as codec driver and uses codec-to-codec links to connect to the Synopsys controller. This allows to use the regular hdmi-codec driver (used by dw-hdmi i2s). To avoid glitches while switching input, the trick is to temporarily force a disconnection of the mux output, which shutdowns the output dai link. This also ensure that the stream parameters and fmt are updated when the output is connected back. Jerome Brunet (5): ASoC: meson: axg-card: set link name based on link node name ASoC: dapm: allow muxes to force a disconnect ASoC: meson: add tohdmitx DT bindings ASoC: meson: axg-card: add basic codec-to-codec link support ASoC: meson: add g12a tohdmitx control .../bindings/sound/amlogic,g12a-tohdmitx.txt | 55 +++ .../dt-bindings/sound/meson-g12a-tohdmitx.h | 13 + sound/soc/meson/Kconfig | 8 + sound/soc/meson/Makefile | 2 + sound/soc/meson/axg-card.c | 31 +- sound/soc/meson/g12a-tohdmitx.c | 413 ++++++++++++++++++ sound/soc/soc-dapm.c | 2 +- 7 files changed, 518 insertions(+), 6 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/amlogic,g12a-tohdmitx.txt create mode 100644 include/dt-bindings/sound/meson-g12a-tohdmitx.h create mode 100644 sound/soc/meson/g12a-tohdmitx.c -- 2.20.1

4 12

[alsa-devel] [PATCH V3] ASoC: cs42xx8: Add reset gpio handling
by S.j. Wang 16 May '19

16 May '19

Handle the reset GPIO and reset the device every time we start it. Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v3 - update binding document. Changes in v2 - use devm_gpiod_get_optional instead of of_get_named_gpio Documentation/devicetree/bindings/sound/cs42xx8.txt | 6 ++++++ sound/soc/codecs/cs42xx8.c | 13 +++++++++++++ 2 files changed, 19 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt index 8619a156d038..ab8f54095269 100644 --- a/Documentation/devicetree/bindings/sound/cs42xx8.txt +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -14,6 +14,11 @@ Required properties: - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, as covered in Documentation/devicetree/bindings/regulator/regulator.txt +Optional properties: + + - reset-gpio : a GPIO spec to define which pin is connected to the chip's + !RESET pin + Example: cs42888: codec@48 { @@ -25,4 +30,5 @@ cs42888: codec@48 { VD-supply = <&reg_audio>; VLS-supply = <&reg_audio>; VLC-supply = <&reg_audio>; + reset-gpio = <&pca9557_b 1 1>; }; diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 28a4ac36c4f8..b377cddaf2e6 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -14,6 +14,7 @@ #include <linux/delay.h> #include <linux/module.h> #include <linux/of_device.h> +#include <linux/of_gpio.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> #include <sound/pcm_params.h> @@ -45,6 +46,7 @@ struct cs42xx8_priv { bool slave_mode; unsigned long sysclk; u32 tx_channels; + struct gpio_desc *gpiod_reset; }; /* -127.5dB to 0dB with step of 0.5dB */ @@ -467,6 +469,13 @@ int cs42xx8_probe(struct device *dev, struct regmap *regmap) return -EINVAL; } + cs42xx8->gpiod_reset = devm_gpiod_get_optional(dev, "reset", + GPIOD_OUT_HIGH); + if (IS_ERR(cs42xx8->gpiod_reset)) + return PTR_ERR(cs42xx8->gpiod_reset); + + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); + cs42xx8->clk = devm_clk_get(dev, "mclk"); if (IS_ERR(cs42xx8->clk)) { dev_err(dev, "failed to get the clock: %ld\n", @@ -547,6 +556,8 @@ static int cs42xx8_runtime_resume(struct device *dev) return ret; } + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 0); + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); if (ret) { @@ -586,6 +597,8 @@ static int cs42xx8_runtime_suspend(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + gpiod_set_value_cansleep(cs42xx8->gpiod_reset, 1); + clk_disable_unprepare(cs42xx8->clk); return 0; -- 2.21.0

1 0

[alsa-devel] ASoc : cs42xx8 : merge cs42xx8-i2c.c into cs42xx8.c
by Matt Flax 16 May '19

16 May '19

Do you think it would be a good idea to merge cs42xx8-i2c.c into cs42xx8.c ? Is there a reason to keep these two related driver functions separate ? Here is their source code : https://github.com/torvalds/linux/blob/master/sound/soc/codecs/cs42xx8-i2c.c https://github.com/torvalds/linux/blob/master/sound/soc/codecs/cs42xx8.c Matt

3 3

[alsa-devel] [PATCH 0/2] ASoC: Mediatek: MT8183: support both S32_LE and S24_LE
by Shunli Wang 16 May '19

16 May '19

This patch adds support both S32_LE and S24_LE. The patch is based on broonie tree "for-next" branch. Shunli Wang (2): ASoC: Mediatek: add memory interface data align ASoC: Mediatek: MT8183: set data align sound/soc/mediatek/common/mtk-afe-fe-dai.c | 7 +++++++ sound/soc/mediatek/common/mtk-base-afe.h | 2 ++ sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 18 ++++++++++++++++++ 3 files changed, 27 insertions(+) -- 2.18.0

2 4

Re: [alsa-devel] [PATCH] ASoC: cs42xx8: Add reset gpio handling
by S.j. Wang 16 May '19

16 May '19

Hi Mark > > On Mon, Apr 29, 2019 at 10:46:03AM +0000, S.j. Wang wrote: > > > + cs42xx8->gpio_reset = of_get_named_gpio(dev->of_node, "gpio- > reset", 0); > > + if (gpio_is_valid(cs42xx8->gpio_reset)) { > > + ret = devm_gpio_request_one(dev, cs42xx8->gpio_reset, > > + GPIOF_OUT_INIT_LOW, "cs42xx8 reset"); > > You should just be able to request the GPIO by name without going > through of_get_named_gpio() using devm_gpio_get(). > Will send v2. > > @@ -559,6 +577,7 @@ static int cs42xx8_runtime_resume(struct device > > *dev) > > > > regcache_cache_only(cs42xx8->regmap, false); > > > > + regcache_mark_dirty(cs42xx8->regmap); > > ret = regcache_sync(cs42xx8->regmap); > > if (ret) { > > dev_err(dev, "failed to sync regmap: %d\n", ret); > > This looks like an unrelated bugfix. Will separate it to another patch. Best regards Wang shengjiu

1 0

[alsa-devel] [PATCH v1 0/2] Huawei WMI laptop extras driver
by Ayman Bagabas 15 May '19

15 May '19

Using the WMI BIOS management interface found on Huawei laptops, extra features such as micmute LED, charging thresholds, and fn-lock can be controlled. A platform driver has been implemented along with the existing WMI driver. The platform driver controls the LED, battery charging thresholds, and fn-lock. We enable sound micmute hooks to platform driver on Huawei laptops to get the LED to work. Ayman Bagabas (2): platform/x86: Huawei WMI laptop extras driver update sound: Enable micmute led for Huawei laptops drivers/platform/x86/Kconfig | 8 +- drivers/platform/x86/huawei-wmi.c | 578 +++++++++++++++++++++++++----- sound/pci/hda/patch_realtek.c | 9 +- 3 files changed, 504 insertions(+), 91 deletions(-) -- 2.20.1

2 4

[alsa-devel] WARNING in snd_info_free_entry
by syzbot 15 May '19

15 May '19

Hello, syzbot found the following crash on: HEAD commit: 43151d6c usb-fuzzer: main usb gadget fuzzer driver git tree: https://github.com/google/kasan.git usb-fuzzer console output: https://syzkaller.appspot.com/x/log.txt?x=16ae9974a00000 kernel config: https://syzkaller.appspot.com/x/.config?x=95aff7278e7ff25e dashboard link: https://syzkaller.appspot.com/bug?extid=131dee8780c719bc6afb compiler: gcc (GCC) 9.0.0 20181231 (experimental) syz repro: https://syzkaller.appspot.com/x/repro.syz?x=173f3574a00000 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+131dee8780c719bc6afb(a)syzkaller.appspotmail.com snd_usb_toneport 6-1:0.0: set_interface failed snd_usb_toneport 6-1:0.0: Line 6 POD Studio UX2 now disconnected ------------[ cut here ]------------ list_del corruption. prev->next should be ffff8881d13fa728, but was ffff8881d82ebd68 WARNING: CPU: 1 PID: 1477 at lib/list_debug.c:51 __list_del_entry_valid+0x101/0x170 lib/list_debug.c:51 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 1477 Comm: kworker/1:2 Not tainted 5.1.0-rc3+ #8 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Workqueue: usb_hub_wq hub_event Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0xca/0x13e lib/dump_stack.c:113 panic+0x292/0x5e1 kernel/panic.c:214 __warn.cold+0x20/0x53 kernel/panic.c:571 report_bug+0x262/0x2a0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:179 [inline] fixup_bug arch/x86/kernel/traps.c:174 [inline] do_error_trap+0x12b/0x1e0 arch/x86/kernel/traps.c:272 do_invalid_op+0x32/0x40 arch/x86/kernel/traps.c:291 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:973 RIP: 0010:__list_del_entry_valid+0x101/0x170 lib/list_debug.c:51 Code: c4 08 31 c0 41 5c 41 5d c3 4c 89 e2 48 c7 c7 20 f7 d8 85 e8 1c 2a 35 ff 0f 0b 31 c0 eb c6 48 c7 c7 80 f7 d8 85 e8 0a 2a 35 ff <0f> 0b 31 c0 eb b4 48 c7 c7 e0 f7 d8 85 e8 f8 29 35 ff 0f 0b 31 c0 RSP: 0018:ffff8881d7a1f078 EFLAGS: 00010282 RAX: 0000000000000000 RBX: ffff8881d13fa630 RCX: 0000000000000000 RDX: 0000000000000000 RSI: ffffffff8127bbcd RDI: ffffed103af43e01 RBP: ffff8881d13fa640 R08: ffff8881d7a10000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: ffff8881d13fad68 R13: ffff8881d7f1d858 R14: ffff8881d13fa630 R15: ffff8881d13fa718 __list_del_entry include/linux/list.h:117 [inline] list_del include/linux/list.h:125 [inline] snd_info_free_entry+0x132/0x340 sound/core/info.c:795 snd_info_card_free+0x3e/0x80 sound/core/info.c:606 snd_card_do_free sound/core/init.c:469 [inline] release_card_device+0x63/0xd0 sound/core/init.c:156 device_release+0x78/0x210 drivers/base/core.c:1064 kobject_cleanup lib/kobject.c:662 [inline] kobject_release lib/kobject.c:691 [inline] kref_put include/linux/kref.h:67 [inline] kobject_put+0x17d/0x270 lib/kobject.c:708 put_device+0x1c/0x30 drivers/base/core.c:2205 snd_card_free_when_closed+0x35/0x50 sound/core/init.c:493 line6_disconnect.cold+0xa3/0xbc sound/usb/line6/driver.c:833 line6_probe+0xb1e/0x1140 sound/usb/line6/driver.c:799 usb_probe_interface+0x30d/0x7b0 drivers/usb/core/driver.c:361 really_probe+0x296/0x680 drivers/base/dd.c:509 driver_probe_device+0xf9/0x200 drivers/base/dd.c:671 __device_attach_driver+0x1c4/0x230 drivers/base/dd.c:778 bus_for_each_drv+0x15e/0x1e0 drivers/base/bus.c:454 __device_attach+0x21e/0x360 drivers/base/dd.c:844 bus_probe_device+0x1ec/0x2a0 drivers/base/bus.c:514 device_add+0xaf4/0x1700 drivers/base/core.c:2106 usb_set_configuration+0xdf2/0x1670 drivers/usb/core/message.c:2023 generic_probe+0x9d/0xd5 drivers/usb/core/generic.c:210 usb_probe_device+0xa8/0x110 drivers/usb/core/driver.c:266 really_probe+0x296/0x680 drivers/base/dd.c:509 driver_probe_device+0xf9/0x200 drivers/base/dd.c:671 __device_attach_driver+0x1c4/0x230 drivers/base/dd.c:778 bus_for_each_drv+0x15e/0x1e0 drivers/base/bus.c:454 __device_attach+0x21e/0x360 drivers/base/dd.c:844 bus_probe_device+0x1ec/0x2a0 drivers/base/bus.c:514 device_add+0xaf4/0x1700 drivers/base/core.c:2106 usb_new_device.cold+0x8b8/0x1030 drivers/usb/core/hub.c:2534 hub_port_connect drivers/usb/core/hub.c:5089 [inline] hub_port_connect_change drivers/usb/core/hub.c:5204 [inline] port_event drivers/usb/core/hub.c:5350 [inline] hub_event+0x1ac9/0x35a0 drivers/usb/core/hub.c:5432 process_one_work+0x90a/0x1580 kernel/workqueue.c:2269 process_scheduled_works kernel/workqueue.c:2331 [inline] worker_thread+0x7ab/0xe20 kernel/workqueue.c:2417 kthread+0x30e/0x420 kernel/kthread.c:253 ret_from_fork+0x3a/0x50 arch/x86/entry/entry_64.S:352 Kernel Offset: disabled Rebooting in 86400 seconds.. --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot. syzbot can test patches for this bug, for details see: https://goo.gl/tpsmEJ#testing-patches

2 1

[alsa-devel] [PATCH v2] ACPI / device_sysfs: change _ADR representation to 64 bits
by Pierre-Louis Bossart 15 May '19

15 May '19

Standards such as the MIPI DisCo for SoundWire 1.0 specification assume the _ADR field is 64 bits. _ADR is defined as an "Integer" represented as 64 bits since ACPI 2.0 released in 2002. The low levels already use _ADR as 64 bits, e.g. in struct acpi_device_info. This patch bumps the representation used for sysfs to 64 bits. To avoid any compatibility/ABI issues, the printf format is only extended to 16 characters when the actual _ADR value exceeds the 32 bit maximum. Example with a SoundWire device, the results show the complete vendorID and linkID which were omitted before: Before: $ more /sys/bus/acpi/devices/device\:38/adr 0x5d070000 After: $ more /sys/bus/acpi/devices/device\:38/adr 0x000010025d070000 Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- v2: only use 64 bits when required to avoid compatibility issues (feedback from Vinod and Rafael) drivers/acpi/device_sysfs.c | 6 ++++-- include/acpi/acpi_bus.h | 2 +- 2 files changed, 5 insertions(+), 3 deletions(-) diff --git a/drivers/acpi/device_sysfs.c b/drivers/acpi/device_sysfs.c index 8940054d6250..7dda0ee05cd1 100644 --- a/drivers/acpi/device_sysfs.c +++ b/drivers/acpi/device_sysfs.c @@ -428,8 +428,10 @@ static ssize_t acpi_device_adr_show(struct device *dev, { struct acpi_device *acpi_dev = to_acpi_device(dev); - return sprintf(buf, "0x%08x\n", - (unsigned int)(acpi_dev->pnp.bus_address)); + if (acpi_dev->pnp.bus_address > 0xFFFFFFFF) + return sprintf(buf, "0x%016llx\n", acpi_dev->pnp.bus_address); + else + return sprintf(buf, "0x%08llx\n", acpi_dev->pnp.bus_address); } static DEVICE_ATTR(adr, 0444, acpi_device_adr_show, NULL); diff --git a/include/acpi/acpi_bus.h b/include/acpi/acpi_bus.h index f7981751ac77..9075e28ea60a 100644 --- a/include/acpi/acpi_bus.h +++ b/include/acpi/acpi_bus.h @@ -230,7 +230,7 @@ struct acpi_device_dir { /* Plug and Play */ typedef char acpi_bus_id[8]; -typedef unsigned long acpi_bus_address; +typedef u64 acpi_bus_address; typedef char acpi_device_name[40]; typedef char acpi_device_class[20]; -- 2.17.1

5 5

[alsa-devel] [PATCH 0/5] ASoC: mediatek: Add basic PCM driver for MT8516
by Fabien Parent 14 May '19

14 May '19

This patch series add a basic PCM driver for MediaTek MT8516 with only support for ADDA Playback & Recording for now. Fabien Parent (5): ASoC: mediatek: make agent_disable, msb & hd fields optional dt-bindings: sound: Add MT8516 AFE PCM bindings ASoC: mediatek: Add MT8516 PCM driver ASoC: mediatek: mt8516: Add ADDA DAI driver ASoC: mediatek: mt8516: register ADDA DAI .../bindings/sound/mt8516-afe-pcm.txt | 28 + sound/soc/mediatek/Kconfig | 10 + sound/soc/mediatek/Makefile | 1 + sound/soc/mediatek/common/mtk-afe-fe-dai.c | 23 +- sound/soc/mediatek/mt8516/Makefile | 7 + sound/soc/mediatek/mt8516/mt8516-afe-common.h | 18 + sound/soc/mediatek/mt8516/mt8516-afe-pcm.c | 794 ++++++++++++++++++ sound/soc/mediatek/mt8516/mt8516-afe-regs.h | 218 +++++ sound/soc/mediatek/mt8516/mt8516-dai-adda.c | 316 +++++++ 9 files changed, 1406 insertions(+), 9 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/mt8516-afe-pcm.txt create mode 100644 sound/soc/mediatek/mt8516/Makefile create mode 100644 sound/soc/mediatek/mt8516/mt8516-afe-common.h create mode 100644 sound/soc/mediatek/mt8516/mt8516-afe-pcm.c create mode 100644 sound/soc/mediatek/mt8516/mt8516-afe-regs.h create mode 100644 sound/soc/mediatek/mt8516/mt8516-dai-adda.c -- 2.20.1

3 7

[alsa-devel] [PATCH] Fix for Lenovo B50-70 inverted internal microphone bug.
by Michał Wadowski 14 May '19

14 May '19

Add patch for realtek codec in Lenovo B50-70. Bugzilla: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1524215 Signed-off-by: Michał Wadowski <wadosm(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 42cd3945e0de..36c18a5ba288 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5676,6 +5676,7 @@ enum { ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE, ALC233_FIXUP_LENOVO_MULTI_CODECS, ALC233_FIXUP_ACER_HEADSET_MIC, + ALC233_FIXUP_INV_DMIC, ALC294_FIXUP_LENOVO_MIC_LOCATION, ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE, ALC700_FIXUP_INTEL_REFERENCE, @@ -6384,6 +6385,10 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc233_fixup_lenovo_line2_mic_hotkey, }, + [ALC233_FIXUP_INV_DMIC] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_inv_dmic, + }, [ALC255_FIXUP_DELL_SPK_NOISE] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_disable_aamix, @@ -6975,6 +6980,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x313c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI), SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC), + SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC233_FIXUP_INV_DMIC), SND_PCI_QUIRK(0x17aa, 0x3978, "IdeaPad Y410P", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x17aa, 0x5013, "Thinkpad", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x501a, "Thinkpad", ALC283_FIXUP_INT_MIC), @@ -7157,6 +7163,7 @@ static const struct hda_model_fixup alc269_fixup_models[] = { {.id = ALC256_FIXUP_ASUS_AIO_GPIO2, .name = "alc256-asus-aio"}, {.id = ALC233_FIXUP_ASUS_MIC_NO_PRESENCE, .name = "alc233-asus"}, {.id = ALC233_FIXUP_EAPD_COEF_AND_MIC_NO_PRESENCE, .name = "alc233-eapd"}, + {.id = ALC233_FIXUP_INV_DMIC, .name = "acl233-inv-dmic"}, {.id = ALC294_FIXUP_LENOVO_MIC_LOCATION, .name = "alc294-lenovo-mic"}, {.id = ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE, .name = "alc225-wyse"}, {.id = ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, .name = "alc274-dell-aio"}, -- 2.7.4

2 5

[alsa-devel] [PATCH v4][RFC] ASoC: modern style CPU
by Kuninori Morimoto 14 May '19

14 May '19

Hi Mark, Pierre-Louis I posted v3 refactoring modern style macro 2 weeks ago, and there is no response yet. I repost it, then, fixup a little bit. These are v4 of RFC of modern style CPU support. v3 -> v4 - rename xxx_CPU to xxx_FE rename xxx_Codec to xxx_BE rename xxx_PLATFORM to xxx_DMA v2 -> v3 - use __VA_ARGS__ We can use same macro for both With / Without Platform v1 -> v2 - Macro naming fixup - no more _CC, _CCP - has manual CPU/Codec/Platform setup

2 6

[alsa-devel] [PATCH V2 0/2] ASoC: ak4458: fail on probe if codec is not present
by Viorel Suman 13 May '19

13 May '19

AK4458 is probed successfully even if AK4458 is not present - this is caused by probe function returning no error on i2c access failure. The patchset fixes this. Changes since V1: Conditional statement rewritten as suggested by Mark. Viorel Suman (2): ASoC: ak4458: rstn_control - return a non-zero on error only ASoC: ak4458: add return value for ak4458_probe sound/soc/codecs/ak4458.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) -- 2.7.4

3 5

[alsa-devel] Fix for Lenovo B50-70 inverted internal microphone bug
by Michał Wadowski 13 May '19

13 May '19

Hi. I prepared patch for this bug: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/1524215 I tested this patch on Linux 5.1.0 and it fixes problem on my Lenovo B50-70 laptop.

2 1

Re: [alsa-devel] Boot procudure on HDA driver
by Takashi Iwai 13 May '19

13 May '19

On Mon, 13 May 2019 15:26:12 +0200, Kailang wrote: > > >That is, the step 2 and 3 are already out of the initialization phase, > >and they are normal runtime PM procedure. If the runtime depop code > >is superfluous for the runtime PM, you can skip it by checking the > >power_state, for example. But, again, it has nothing to do with the > >boot or not. It's just the normal runtime suspend and resume. > > Sorry! runtime suspend also need to run depop procedure. > It's not need to run depop procedure which was system power up during boot on alc225_shutup(). Hrm, it's still not clear to me. The alc255_shutup() is called during the boot up because the codec goes to the runtime suspend/resume. There should be no difference between this call during the boot time and the normal runtime PM behavior. Takashi > So, I think I can put checking runtime PM and Power_state and shutdown code in alc225_shutup(). > > for example: (put below code in alc225_shutup) > > If ( !pm_runtime_suspended(hda_codec_dev(codec)) || !codec->bus->shutdown || > codec->core.dev.power.power_state.event != PM_EVENT_SUSPEND || > codec->core.dev.power.power_state.event != PM_EVENT_HIBERNATE) > return; > But I don't know which runtime_suspend state was pm_runtime_suspended(hda_codec_dev(codec)). > ________________________________________ > 從: Takashi Iwai [tiwai(a)suse.de] > 寄件日期: 2019年5月13日下午 08:01 > 至: Kailang > 副本: (alsa-devel(a)alsa-project.org) > 主旨: Re: Boot procudure on HDA driver > > On Mon, 13 May 2019 13:52:32 +0200, > Kailang wrote: > > > > > > > > >Well, it's not clear to me whether you'd like to reduce the depop > > >procedure above or to add it somewhere else. Could you clarify? > > > > This depop procedure in alc225_shutup. it doesn't need to run in system boot up. > > > > If system boot up, it will run alc225_init() already. alc225_init() include power up depop procedure. > > So, if system boot up, it only need to run depop procedure in alc225_init(). > > This is the runtime suspend and resume. That is: > > > > The current boot up procedure was below. > > 1. alc225_init() > > ... which is called at the driver probe time, from > snd_hda_codec_build_controls(). > > Then the codec driver initialization finishes at this point, hence it > goes to runtime suspend. > > > 2. alc225_shutup() ==> this include power off and suspend depop procedure. some codec run this field will have pop noise at boot up. > > ... which is a part of the codec runtime suspend. > > Then, the codec gets woken up for some access, and it performs the > runtime resume. > > > 3. alc225_init() > > ... which is a part of the codec runtime resume. > > That is, the step 2 and 3 are already out of the initialization phase, > and they are normal runtime PM procedure. If the runtime depop code > is superfluous for the runtime PM, you can skip it by checking the > power_state, for example. But, again, it has nothing to do with the > boot or not. It's just the normal runtime suspend and resume. > > > thanks, > > Takashi > > > > > > > > ________________________________________ > > 從: Takashi Iwai [tiwai(a)suse.de] > > 寄件日期: 2019年5月13日下午 07:30 > > 至: Kailang > > 副本: (alsa-devel(a)alsa-project.org) > > 主旨: Re: Boot procudure on HDA driver > > > > On Mon, 13 May 2019 12:36:36 +0200, > > Kailang wrote: > > > > > > > > > Maybe I confuse and I also confuse you. Sorry!! > > > > > > When system suspend, it will run alc_suspend(). alc_suspend() was include alc_shutup(). > > > > Right. > > > > > When system shutdown(power off), will it run alc_shutup()? > > > > Yes, it's called via alc_reboot_notify(). > > If spec->reboot_notify is defined, this supersedes the call, so you > > may define the callback if you'd need to avoid alc_shutup() call from > > there. > > > > > > > in our patch_realtek.c. > > > spec->shutup = alc225_shutup; > > > Some codec didn't run the depop procudure during boot up. > > > > > > Maybe disable EAPD and shutup PIN will cause pop noise. > > > But this procudure need to run in SUSPEND and HIBERNATE and Power off. > > > It also need to run it at runtime suspend. > > > > > > static void alc225_shutup(struct hda_codec *codec) > > > { > > > struct alc_spec *spec = codec->spec; > > > hda_nid_t hp_pin = spec->gen.autocfg.hp_pins[0]; > > > bool hp1_pin_sense, hp2_pin_sense; > > > > > > *********** depop procudure ****************************************** > > > if (!hp_pin) > > > hp_pin = 0x21; > > > > > > /* 3k pull low control for Headset jack. */ > > > alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); > > > > > > hp1_pin_sense = snd_hda_jack_detect(codec, hp_pin); > > > hp2_pin_sense = snd_hda_jack_detect(codec, 0x16); > > > > > > if (hp1_pin_sense || hp2_pin_sense) > > > msleep(2); > > > > > > if (hp1_pin_sense) > > > snd_hda_codec_write(codec, hp_pin, 0, > > > AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); > > > if (hp2_pin_sense) > > > snd_hda_codec_write(codec, 0x16, 0, > > > AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE); > > > > > > if (hp1_pin_sense || hp2_pin_sense) > > > msleep(85); > > > > > > if (hp1_pin_sense) > > > snd_hda_codec_write(codec, hp_pin, 0, > > > AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); > > > if (hp2_pin_sense) > > > snd_hda_codec_write(codec, 0x16, 0, > > > AC_VERB_SET_PIN_WIDGET_CONTROL, 0x0); > > > > > > if (hp1_pin_sense || hp2_pin_sense) > > > msleep(100); > > > > > > alc_auto_setup_eapd(codec, false); > > > snd_hda_shutup_pins(codec); > > > ************** depop procudure ************************************* > > > } > > > > Well, it's not clear to me whether you'd like to reduce the depop > > procedure above or to add it somewhere else. Could you clarify? > > > > > > thanks, > > > > Takashi > > > > > > > > > > > > > > > > > > > > > > ________________________________________ > > > 從: Takashi Iwai [tiwai(a)suse.de] > > > 寄件日期: 2019年5月13日下午 05:42 > > > 至: Kailang > > > 副本: (alsa-devel(a)alsa-project.org) > > > 主旨: Re: Boot procudure on HDA driver > > > > > > On Mon, 13 May 2019 11:30:56 +0200, > > > Kailang wrote: > > > > > > > > > > > > > > > > > -----Original Message----- > > > > > From: Takashi Iwai <tiwai(a)suse.de> > > > > > Sent: Monday, May 13, 2019 5:11 PM > > > > > To: Kailang <kailang(a)realtek.com> > > > > > Cc: (alsa-devel(a)alsa-project.org) <alsa-devel(a)alsa-project.org> > > > > > Subject: Re: Boot procudure on HDA driver > > > > > > > > > > On Mon, 13 May 2019 11:00:18 +0200, > > > > > Kailang wrote: > > > > > > > > > > > > Hi Takashi, > > > > > > > > > > > > When System Boot up. > > > > > > The Hda Driver running step was as below. > > > > > > > > > > > > alc_init(); > > > > > > alc_shutup(); > > > > > > alc_init(); > > > > > > > > > > > > The depop procedure was put in spec->init_hook and spec->shutup. > > > > > > > > > > > > But I find more codec which run spec->shutup at boot up. It will occur pop > > > > > noise. > > > > > > If it doesn't run spec->shutup, it will not occur pop noise or reduce pop noise. > > > > > > > > > > > > How could the spec->shutup not run at boot up? > > > > > > > > > > > > I ever test to put the check in spec->shutup() for PM_EVENT_SUSPEND and > > > > > PM_EVENT_HIBERNATE and shutdown. > > > > > > But if power_save=1, it was have issue for this. > > > > > > Codec was idle already in power_save=1 state. If system go suspend, it will > > > > > not run spec->shutup() again. > > > > > > > > > > I guess it's a runtime PM, hence it's neither PM_EVENT_SUSPEND nor > > > > > PM_EVENT_HIBERNATE. > > > > > > > > > runtime PM and suspend and hibernate and shutdown need to run > > > > spec->shutup(). > > > > > > Is the call really mandatory? > > > > > > > It's no problem. > > > > But spec->shutup() doesn't need to run in boot up. > > > > > > The call of spec->shutup() at boot up *is* the runtime PM. > > > Or any other call path I overlooked? > > > > > > > And it will set power_save=1 on all dell machine. > > > > > > > > If (codec->auto_runtime_pm || codec->bus->shutdown || > > > > codec->core.dev.power.power_state.event == PM_EVENT_SUSPEND || > > > > codec->core.dev.power.power_state.event == PM_EVENT_HIBERNATE) > > > > > > > > So, I need to put upper check code in spec->shutup(). Right? > > > > Thanks. > > > > > > No, the auto_runtime_pm is for a completely different purpose. > > > > > > At the boot up, the runtime PM can be kicked in at any time. So if > > > you disable the shutup during the runtime PM, it means you'd need to > > > call the shutup at runtime PM completely. > > > > > > > > > Takashi > > > > > > > > > > > > Actually, if the shutup procedure makes the problem on a certain platform, just > > > > > skip it. It's an optional behavior and would be fine without it (of course only if > > > > > it's confirmed to work :) > > > > > > > > > > > > > > > Takashi > > > > > > > > > > ------Please consider the environment before printing this e-mail. > > > >

1 0

[alsa-devel] [PATCH] ASoC: max98090: remove 24-bit format support
by Yu-Hsuan Hsu 13 May '19

13 May '19

Remove 24-bit format support because it doesn't work now. We can revert this change after it really supports. (https://patchwork.kernel.org/patch/10783561/) Signed-off-by: Yu-Hsuan Hsu <yuhsuan(a)chromium.org> --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 7619ea31ab50..b25b7efa9118 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2313,7 +2313,7 @@ int max98090_mic_detect(struct snd_soc_component *component, EXPORT_SYMBOL_GPL(max98090_mic_detect); #define MAX98090_RATES SNDRV_PCM_RATE_8000_96000 -#define MAX98090_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) +#define MAX98090_FORMATS SNDRV_PCM_FMTBIT_S16_LE static const struct snd_soc_dai_ops max98090_dai_ops = { .set_sysclk = max98090_dai_set_sysclk, -- 2.21.0.1020.gf2820cf01a-goog

2 1

[alsa-devel] [PATCH v2] ALSA: hda: Fix race between creating and refreshing sysfs entries
by Amadeusz Sławiński 13 May '19

13 May '19

hda_widget_sysfs_reinit() can free underlying codec->widgets structure on which widget_tree_create() operates. Add locking to prevent such issues from happening. Bugzilla: https://bugs.freedesktop.org/show_bug.cgi?id=110382 Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- Changes since v1: - Moved mutexes around the callers - Added comments noting that functions should be called with mutex held --- include/sound/hdaudio.h | 1 + sound/hda/hdac_device.c | 7 +++++++ sound/hda/hdac_sysfs.c | 3 +++ 3 files changed, 11 insertions(+) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index 45f944d57982..85835d0c33cc 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -81,6 +81,7 @@ struct hdac_device { atomic_t in_pm; /* suspend/resume being performed */ /* sysfs */ + struct mutex widget_lock; struct hdac_widget_tree *widgets; /* regmap */ diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index 95b073ee4b32..4769f4c03e14 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -55,6 +55,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus, codec->bus = bus; codec->addr = addr; codec->type = HDA_DEV_CORE; + mutex_init(&codec->widget_lock); pm_runtime_set_active(&codec->dev); pm_runtime_get_noresume(&codec->dev); atomic_set(&codec->in_pm, 0); @@ -141,7 +142,9 @@ int snd_hdac_device_register(struct hdac_device *codec) err = device_add(&codec->dev); if (err < 0) return err; + mutex_lock(&codec->widget_lock); err = hda_widget_sysfs_init(codec); + mutex_unlock(&codec->widget_lock); if (err < 0) { device_del(&codec->dev); return err; @@ -158,7 +161,9 @@ EXPORT_SYMBOL_GPL(snd_hdac_device_register); void snd_hdac_device_unregister(struct hdac_device *codec) { if (device_is_registered(&codec->dev)) { + mutex_lock(&codec->widget_lock); hda_widget_sysfs_exit(codec); + mutex_unlock(&codec->widget_lock); device_del(&codec->dev); snd_hdac_bus_remove_device(codec->bus, codec); } @@ -404,7 +409,9 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs) } if (sysfs) { + mutex_lock(&codec->widget_lock); err = hda_widget_sysfs_reinit(codec, start_nid, nums); + mutex_unlock(&codec->widget_lock); if (err < 0) return err; } diff --git a/sound/hda/hdac_sysfs.c b/sound/hda/hdac_sysfs.c index fb2aa344981e..909d5ef1179c 100644 --- a/sound/hda/hdac_sysfs.c +++ b/sound/hda/hdac_sysfs.c @@ -395,6 +395,7 @@ static int widget_tree_create(struct hdac_device *codec) return 0; } +/* call with codec->widget_lock held */ int hda_widget_sysfs_init(struct hdac_device *codec) { int err; @@ -411,11 +412,13 @@ int hda_widget_sysfs_init(struct hdac_device *codec) return 0; } +/* call with codec->widget_lock held */ void hda_widget_sysfs_exit(struct hdac_device *codec) { widget_tree_free(codec); } +/* call with codec->widget_lock held */ int hda_widget_sysfs_reinit(struct hdac_device *codec, hda_nid_t start_nid, int num_nodes) { -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: kbl: fix wrong number of channels
by Tzung-Bi Shih 13 May '19

13 May '19

Fix wrong setting on number of channels. The context wants to set constraint to 2 channels instead of 4. Signed-off-by: Tzung-Bi Shih <tzungbi(a)google.com> --- sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c index 7044d8c2b187..879f14257a3e 100644 --- a/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c +++ b/sound/soc/intel/boards/kbl_rt5663_rt5514_max98927.c @@ -405,7 +405,7 @@ static const struct snd_pcm_hw_constraint_list constraints_dmic_channels = { }; static const unsigned int dmic_2ch[] = { - 4, + 2, }; static const struct snd_pcm_hw_constraint_list constraints_dmic_2ch = { -- 2.21.0.392.gf8f6787159e-goog

3 3

[alsa-devel] [PATCH] ASoC: SOF: nocodec: fix undefined reference
by Pierre-Louis Bossart 13 May '19

13 May '19

The nocodec option can be selected individually, leading to the following issue: sound/soc/sof/core.o: In function `snd_sof_device_probe': core.c:(.text+0x4af): undefined reference to `sof_nocodec_setup' Fix by selecting the SND_SOF_NOCODEC option as needed. Reported-by: Hulk Robot <hulkci(a)huawei.com> Reported-by: YueHaibing <yuehaibing(a)huawei.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/sof/Kconfig | 8 ++++++-- 1 file changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/Kconfig b/sound/soc/sof/Kconfig index b204c65698f9..a9a1d502daae 100644 --- a/sound/soc/sof/Kconfig +++ b/sound/soc/sof/Kconfig @@ -44,7 +44,10 @@ config SND_SOC_SOF_OPTIONS if SND_SOC_SOF_OPTIONS config SND_SOC_SOF_NOCODEC - tristate "SOF nocodec mode Support" + tristate + +config SND_SOC_SOF_NOCODEC_SUPPORT + bool "SOF nocodec mode support" help This adds support for a dummy/nocodec machine driver fallback option if no known codec is detected. This is typically only @@ -80,7 +83,7 @@ if SND_SOC_SOF_DEBUG config SND_SOC_SOF_FORCE_NOCODEC_MODE bool "SOF force nocodec Mode" - depends on SND_SOC_SOF_NOCODEC + depends on SND_SOC_SOF_NOCODEC_SUPPORT help This forces SOF to use dummy/nocodec as machine driver, even though there is a codec detected on the real platform. This is @@ -135,6 +138,7 @@ endif ## SND_SOC_SOF_OPTIONS config SND_SOC_SOF tristate select SND_SOC_TOPOLOGY + select SND_SOC_SOF_NOCODEC if SND_SOC_SOF_NOCODEC_SUPPORT help This option is not user-selectable but automagically handled by 'select' statements at a higher level -- 2.17.1

2 1

[alsa-devel] [PATCH] ASoC : cs4265 : readable register too low
by Matt Flax 13 May '19

13 May '19

The cs4265_readable_register function stopped short of the maximum register. An example bug is taken from : https://github.com/Audio-Injector/Ultra/issues/25 Where alsactl store fails with : Cannot read control '2,0,0,C Data Buffer,0': Input/output error This patch fixes the bug by setting the cs4265 to have readable registers up to the maximum hardware register CS4265_MAX_REGISTER. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/codecs/cs4265.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/cs4265.c b/sound/soc/codecs/cs4265.c index ab27d2b94d02..c0190ec59e74 100644 --- a/sound/soc/codecs/cs4265.c +++ b/sound/soc/codecs/cs4265.c @@ -60,7 +60,7 @@ static const struct reg_default cs4265_reg_defaults[] = { static bool cs4265_readable_register(struct device *dev, unsigned int reg) { switch (reg) { - case CS4265_CHIP_ID ... CS4265_SPDIF_CTL2: + case CS4265_CHIP_ID ... CS4265_MAX_REGISTER: return true; default: return false; -- 2.19.1

3 2

[alsa-devel] [PATCH 0/2] ASoC: ak4458: fail on probe if codec is not present
by Viorel Suman 13 May '19

13 May '19

AK4458 is probed successfully even if AK4458 is not present - this is caused by probe function returning no error on i2c access failure. The patchset fixes this. Viorel Suman (2): ASoC: ak4458: rstn_control - return a non-zero on error only ASoC: ak4458: add return value for ak4458_probe sound/soc/codecs/ak4458.c | 16 +++++++++------- 1 file changed, 9 insertions(+), 7 deletions(-) -- 2.7.4

2 4

[alsa-devel] [PATCH] ASoC: amd: Reporting accurate hw_ptr for acp3x dma
by Vijendar Mukunda 13 May '19

13 May '19

From: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> acp3x dma pointer callback has issues in reporting hw_ptr. Modified logic to use linear position registers to retrieve accurate hw_ptr. Signed-off-by: Ravulapati Vishnu vardhan rao <Vishnuvardhanrao.Ravulapati(a)amd.com> Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 43 ++++++++++++++++++++++++++----------- 1 file changed, 31 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index 9775bda..a4ade6b 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -32,6 +32,7 @@ struct i2s_stream_instance { u16 channels; u32 xfer_resolution; struct page *pg; + u64 bytescount; void __iomem *acp3x_base; }; @@ -317,6 +318,24 @@ static int acp3x_dma_open(struct snd_pcm_substream *substream) return 0; } +static u64 acp_get_byte_count(struct i2s_stream_instance *rtd, int direction) +{ + u64 byte_count; + + if (direction == SNDRV_PCM_STREAM_PLAYBACK) { + byte_count = rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_BT_TX_LINEARPOSITIONCNTR_LOW); + } else { + byte_count = rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINEARPOSITIONCNTR_HIGH); + byte_count |= rv_readl(rtd->acp3x_base + + mmACP_BT_RX_LINEARPOSITIONCNTR_LOW); + } + return byte_count; +} + static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -350,18 +369,17 @@ static int acp3x_dma_hw_params(struct snd_pcm_substream *substream, static snd_pcm_uframes_t acp3x_dma_pointer(struct snd_pcm_substream *substream) { u32 pos = 0; - struct i2s_stream_instance *rtd = substream->runtime->private_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - pos = rv_readl(rtd->acp3x_base + - mmACP_BT_TX_LINKPOSITIONCNTR); - else - pos = rv_readl(rtd->acp3x_base + - mmACP_BT_RX_LINKPOSITIONCNTR); - - if (pos >= MAX_BUFFER) - pos = 0; - + u32 buffersize = 0; + u64 bytescount = 0; + struct i2s_stream_instance *rtd = + substream->runtime->private_data; + + buffersize = frames_to_bytes(substream->runtime, + substream->runtime->buffer_size); + bytescount = acp_get_byte_count(rtd, substream->stream); + if (bytescount > rtd->bytescount) + bytescount -= rtd->bytescount; + pos = do_div(bytescount, buffersize); return bytes_to_frames(substream->runtime, pos); } @@ -521,6 +539,7 @@ static int acp3x_dai_i2s_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + rtd->bytescount = acp_get_byte_count(rtd, substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { rv_writel(period_bytes, rtd->acp3x_base + mmACP_BT_TX_INTR_WATERMARK_SIZE); -- 2.7.4

2 1

[alsa-devel] [PATCH v3 0/2] ASoC: Intel: Add Cometlake PCI IDs
by Evan Green 13 May '19

13 May '19

This small series adds PCI IDs for Cometlake platforms, for a dazzling audio experience. This is based on linux-next's next-20190510. Changes in v3: - Copy cnl_desc to new cml_desc, and avoid selecting cannonlake (Pierre-Louis) - Don't select CML_* in SND_SOC_INTEL_SKYLAKE (Pierre-Louis) Changes in v2: - Add CML-H ID 0x06c8 (Pierre-Louis) - Add 0x06c8 for CML-H (Pierre-Louis) Evan Green (2): ASoC: SOF: Add Comet Lake PCI IDs ASoC: Intel: Skylake: Add Cometlake PCI IDs sound/soc/intel/Kconfig | 16 +++++++++++++ sound/soc/intel/skylake/skl-messages.c | 16 +++++++++++++ sound/soc/intel/skylake/skl.c | 10 ++++++++ sound/soc/sof/intel/Kconfig | 32 ++++++++++++++++++++++++++ sound/soc/sof/sof-pci-dev.c | 28 ++++++++++++++++++++++ 5 files changed, 102 insertions(+) -- 2.20.1

3 5