December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/4] alsa-utils: axfer: add help texts
by GitHub pull_request - opened 07 Dec '18

07 Dec '18

alsa-project/alsa-utils pull request #7 was opened from takaswie: This patchset adds help texts for command entry, list and transfer subcommands. In this time, contents for each backend of transfer subcommand is not added yet. Unfortunately, execution of aliases (arecord/aplay) can print help text for command text only. This bug will be fixed in future commits. Additionally, due to program design, execution of the aliases cannot print help text of list subcommand. This is unavoidable loss of backward compatibility. ``` Takashi Sakamoto (4): axfer: print help text of command entry axfer: print help text of list subcommand axfer: print help text of transfer subcommand just for common options axfer: enable each backend to print own help axfer/main.c | 12 +++++++++++- axfer/subcmd-list.c | 9 ++++++++- axfer/xfer-libasound.c | 6 ++++++ axfer/xfer-libffado.c | 6 ++++++ axfer/xfer-options.c | 35 +++++++++++++++++++++++++++++++++++ axfer/xfer.c | 5 +++++ axfer/xfer.h | 2 ++ 7 files changed, 73 insertions(+), 2 deletions(-) ``` Request URL : https://github.com/alsa-project/alsa-utils/pull/7 Patch URL : https://github.com/alsa-project/alsa-utils/pull/7.patch Repository URL: https://github.com/alsa-project/alsa-utils

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[alsa-devel] ALC700 headphone no sound issue
by Kailang 07 Dec '18

07 Dec '18

Hi Takashi, In Intel ENG board with ALC700. It had headphone no sound issue. If it plugged headphone or headset into the jack, then do the reboot, it will have a chance to cause headphone no sound. It just need to run the headphone mode procedure after boot time. The issue will be fixed. It also suitable for ALC234 ALC274 and ALC294. BR, Kailang

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[alsa-devel] [GIT PULL] soundwire updates
by Vinod Koul 07 Dec '18

07 Dec '18

Hey Greg, Please pull to receive a single commit for soundwire tree for this cycle. Am off next week so sending this bit early The following changes since commit 651022382c7f8da46cb4872a545ee1da6d097d2a: Linux 4.20-rc1 (2018-11-04 15:37:52 -0800) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-4.21-rc1 for you to fetch changes up to b1635596860dd3229645e41276f6c83ca788e4c1: soundwire: intel: constify snd_soc_dai_ops structures (2018-11-12 15:36:06 +0530) ---------------------------------------------------------------- soundwire updates for 4.21-rc1 A single change: - constify snd_soc_dai_ops struct by Julia ---------------------------------------------------------------- Julia Lawall (1): soundwire: intel: constify snd_soc_dai_ops structures drivers/soundwire/intel.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) -- ~Vinod

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[alsa-devel] [PATCH] tinycompress: Fix missing '}' around extern C
by Charles Keepax 07 Dec '18

07 Dec '18

From: Piotr Stankiewicz <piotrs(a)opensource.cirrus.com> Signed-off-by: Piotr Stankiewicz <piotrs(a)opensource.cirrus.com> Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- include/tinycompress/tinycompress.h | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/include/tinycompress/tinycompress.h b/include/tinycompress/tinycompress.h index 780dce6..f0fae30 100644 --- a/include/tinycompress/tinycompress.h +++ b/include/tinycompress/tinycompress.h @@ -276,4 +276,8 @@ int is_compress_ready(struct compress *compress); /* Returns a human readable reason for the last error */ const char *compress_get_error(struct compress *compress); +#if defined(__cplusplus) +} // extern "C" +#endif + #endif -- 2.11.0

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[alsa-devel] [PATCH 1/3] ASoC: rt5660: Add a new ACPI match ID
by Hui Wang 07 Dec '18

07 Dec '18

The Realtek codec ALC3277 is 100% compatible with the codec RT5660 in I2S mode. And on the Dell IoT platform, the codec is ALC3277, and the HID of the codec in the BIOS is 10EC3277, so adding this ID to the ACPI match table. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/soc/codecs/rt5660.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/rt5660.c b/sound/soc/codecs/rt5660.c index 27f7445b2432..e74b2e8cd423 100644 --- a/sound/soc/codecs/rt5660.c +++ b/sound/soc/codecs/rt5660.c @@ -1246,6 +1246,7 @@ MODULE_DEVICE_TABLE(of, rt5660_of_match); static const struct acpi_device_id rt5660_acpi_match[] = { { "10EC5660", 0 }, + { "10EC3277", 0 }, { }, }; MODULE_DEVICE_TABLE(acpi, rt5660_acpi_match); -- 2.17.1

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[alsa-devel] [PATCH v8 0/7] Add support for audiocodec in Allwinner A64
by Vasily Khoruzhick 07 Dec '18

07 Dec '18

v2: - Use simple-amplifier for speaker amp on Pinebook - Rename sun50i-a64-i2s to sun50i-a64-codec-i2s to preserve compatible string for other 3 I2S modules in A64 in case if there's any incompatibility with H3 v3: - renamed sunxi-adda-pr-regmap to sun8i-adda-pr-regmap - use ilog2() to calculate reg value for LRCK div instead of using a table v4: - dts: don't use 'Mic' and 'Headset Mic' widgets from sun8i-codec, define our board-level widgets instead. v5: - collect all the tags v6: - driver patches has been merged through ASoC tree - rebase onto 4.20-rc1 - Drop 'Speaker' from routes on sopine and pine64, they don't have speaker. v7: - drop 'Mic' and 'Headset Mic' from dtsi - move AIF1 Slot 0 ADC to ADC route to sun8i-codec - drop BIAS routes for Pine64 and SoPine v8 - drop merged patches - add regulator property for simple amplifier, make enable-gpio optional - add regulator to speaker amp, introduce headphone amp Vasily Khoruzhick (7): ASoC: dt-bindings: add regulator property to simple amplifier ASoC: simple-amplifier: add VCC regulator widget ASoC: dt-bindings: make enable-gpio optional for simple amplifier ASoC: simple-amplifier: make enable-gpio optional arm64: dts: allwinner: a64: add nodes necessary for analog sound support arm64: dts: allwinner: a64: enable sound on Pine64 and SoPine arm64: dts: allwinner: a64: enable sound on Pinebook .../bindings/sound/simple-amplifier.txt | 5 ++ .../boot/dts/allwinner/sun50i-a64-pine64.dts | 35 ++++++++++++ .../dts/allwinner/sun50i-a64-pinebook.dts | 53 ++++++++++++++++++ .../allwinner/sun50i-a64-sopine-baseboard.dts | 35 ++++++++++++ arch/arm64/boot/dts/allwinner/sun50i-a64.dtsi | 54 +++++++++++++++++++ sound/soc/codecs/simple-amplifier.c | 7 ++- 6 files changed, 188 insertions(+), 1 deletion(-) -- 2.19.1

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[alsa-devel] [PATCH v2] ASoC: max9867: Remove useless assignment
by Ladislav Michl 06 Dec '18

06 Dec '18

ret is assigned later, no need to initialize it. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- CHANGES: -v2: add missing Signed-off-by (I'm sorry for that...) sound/soc/codecs/max9867.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/max9867.c b/sound/soc/codecs/max9867.c index 1cda54b59854..1b155f40b106 100644 --- a/sound/soc/codecs/max9867.c +++ b/sound/soc/codecs/max9867.c @@ -476,7 +476,7 @@ static int max9867_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct max9867_priv *max9867; - int ret = 0, reg; + int ret, reg; max9867 = devm_kzalloc(&i2c->dev, sizeof(*max9867), GFP_KERNEL); if (!max9867) @@ -497,10 +497,8 @@ static int max9867_i2c_probe(struct i2c_client *i2c, dev_info(&i2c->dev, "device revision: %x\n", reg); ret = devm_snd_soc_register_component(&i2c->dev, &max9867_component, max9867_dai, ARRAY_SIZE(max9867_dai)); - if (ret < 0) { + if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret); - return ret; - } return ret; } -- 2.20.0.rc2

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[alsa-devel] [PATCH] ASoC: intel: cht_bsw_max98090_ti: Add pmc_plt_clk_0 quirk for Chromebook Gnawty
by Hans de Goede 06 Dec '18

06 Dec '18

The Gnawty model Chromebook uses pmc_plt_clk_0 instead of pmc_plt_clk_3 for the mclk, just like the Clapper and Swanky models. This commit adds a DMI based quirk for this. This fixing audio no longer working on these devices after commit 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") that commit fixes us unnecessary keeping unused clocks on, but in case of the Gnawty that was breaking audio support since we were not using the right clock in the cht_bsw_max98090_ti machine driver. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=201787 Cc: stable(a)vger.kernel.org Fixes: 648e921888ad ("clk: x86: Stop marking clocks as CLK_IS_CRITICAL") Reported-and-tested-by: Jaime Pérez <19.jaime.91(a)gmail.com> Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/intel/boards/cht_bsw_max98090_ti.c b/sound/soc/intel/boards/cht_bsw_max98090_ti.c index ad0c98383853..08a5152e635a 100644 --- a/sound/soc/intel/boards/cht_bsw_max98090_ti.c +++ b/sound/soc/intel/boards/cht_bsw_max98090_ti.c @@ -396,6 +396,13 @@ static const struct dmi_system_id cht_max98090_quirk_table[] = { }, .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, }, + { + /* Gnawty model Chromebook (Acer Chromebook CB3-111) */ + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Gnawty"), + }, + .driver_data = (void *)QUIRK_PMC_PLT_CLK_0, + }, { /* Swanky model Chromebook (Toshiba Chromebook 2) */ .matches = { -- 2.19.2

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[alsa-devel] [PATCH 1/2] ASoC: use DMA addr rather than CPU pa for acp_audio_dma
by Yu Zhao 06 Dec '18

06 Dec '18

We shouldn't assume CPU physical address we get from page_to_phys() is same as DMA address we get from dma_alloc_coherent(). On x86_64, we won't run into any problem with the assumption when dma_ops is nommu_dma_ops. However, DMA address is IOVA when IOMMU is enabled. And it's most likely different from CPU physical address when AMD IOMMU is not in passthrough mode. Signed-off-by: Yu Zhao <yuzhao(a)google.com> --- sound/soc/amd/acp-pcm-dma.c | 15 +++++---------- sound/soc/amd/acp.h | 2 +- 2 files changed, 6 insertions(+), 11 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index cdebab2f8ce5..fd3db4c37882 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -303,11 +303,10 @@ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, u32 size, } /* Create page table entries in ACP SRAM for the allocated memory */ -static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, +static void acp_pte_config(void __iomem *acp_mmio, dma_addr_t addr, u16 num_of_pages, u32 pte_offset) { u16 page_idx; - u64 addr; u32 low; u32 high; u32 offset; @@ -317,7 +316,6 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, /* Load the low address of page int ACP SRAM through SRBM */ acp_reg_write((offset + (page_idx * 8)), acp_mmio, mmACP_SRBM_Targ_Idx_Addr); - addr = page_to_phys(pg); low = lower_32_bits(addr); high = upper_32_bits(addr); @@ -333,7 +331,7 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, acp_reg_write(high, acp_mmio, mmACP_SRBM_Targ_Idx_Data); /* Move to next physically contiguos page */ - pg++; + addr += PAGE_SIZE; } } @@ -343,7 +341,7 @@ static void config_acp_dma(void __iomem *acp_mmio, { u16 ch_acp_sysmem, ch_acp_i2s; - acp_pte_config(acp_mmio, rtd->pg, rtd->num_of_pages, + acp_pte_config(acp_mmio, rtd->dma_addr, rtd->num_of_pages, rtd->pte_offset); if (rtd->direction == SNDRV_PCM_STREAM_PLAYBACK) { @@ -850,7 +848,6 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, int status; uint64_t size; u32 val = 0; - struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -986,16 +983,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, return status; memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); - pg = virt_to_page(substream->dma_buffer.area); - if (pg) { + if (substream->dma_buffer.area) { acp_set_sram_bank_state(rtd->acp_mmio, 0, true); /* Save for runtime private data */ - rtd->pg = pg; + rtd->dma_addr = substream->dma_buffer.addr; rtd->order = get_order(size); /* Fill the page table entries in ACP SRAM */ - rtd->pg = pg; rtd->size = size; rtd->num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; rtd->direction = substream->stream; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index dbbb1a85638d..e5ab6c6040a6 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -123,7 +123,7 @@ enum acp_dma_priority_level { }; struct audio_substream_data { - struct page *pg; + dma_addr_t dma_addr; unsigned int order; u16 num_of_pages; u16 i2s_instance; -- 2.20.0.rc1.387.gf8505762e3-goog

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[alsa-devel] [PATCH] ASoC: Use of_node_name_eq for node name comparisons
by Rob Herring 06 Dec '18

06 Dec '18

Convert string compares of DT node names to use of_node_name_eq helper instead. This removes direct access to the node name pointer. For the FSL ASoC card, the full node names appear to be "ssi", "esai", and "sai", so there's not any reason to use strstr and of_node_name_eq can be used instead. 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 <fabio.estevam(a)nxp.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 Signed-off-by: Rob Herring <robh(a)kernel.org> --- sound/soc/fsl/fsl-asoc-card.c | 6 +++--- sound/soc/generic/simple-scu-card.c | 2 +- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 44433b20435c..81f2fe2c6d23 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -571,17 +571,17 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) } /* Common settings for corresponding Freescale CPU DAI driver */ - if (strstr(cpu_np->name, "ssi")) { + if (of_node_name_eq(cpu_np, "ssi")) { /* Only SSI needs to configure AUDMUX */ ret = fsl_asoc_card_audmux_init(np, priv); if (ret) { dev_err(&pdev->dev, "failed to init audmux\n"); goto asrc_fail; } - } else if (strstr(cpu_np->name, "esai")) { + } else if (of_node_name_eq(cpu_np, "esai")) { priv->cpu_priv.sysclk_id[1] = ESAI_HCKT_EXTAL; priv->cpu_priv.sysclk_id[0] = ESAI_HCKR_EXTAL; - } else if (strstr(cpu_np->name, "sai")) { + } else if (of_node_name_eq(cpu_np, "sai")) { priv->cpu_priv.sysclk_id[1] = FSL_SAI_CLK_MAST1; priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index 85b46f0eae0f..e9a1acffcf5b 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -216,7 +216,7 @@ static int asoc_simple_card_parse_of(struct simple_card_data *priv) i = 0; for_each_child_of_node(node, np) { is_fe = false; - if (strcmp(np->name, PREFIX "cpu") == 0) + if (of_node_name_eq(np, PREFIX "cpu")) is_fe = true; ret = asoc_simple_card_dai_link_of(np, priv, daifmt, i, is_fe); -- 2.19.1

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