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

[alsa-devel] Applied "ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer 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 2a2aefa41ce48ace8e1e963cb10c3f5ff43aa994 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.cirrus.com> Date: Fri, 19 Oct 2018 13:25:15 +0100 Subject: [PATCH] ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer The wm_adsp_buffer struct is the control header of a circular buffer used to transfer data from the firmware over the control interface to an ALSA compressed stream. The original names of the fields pointing to the data buffer were based on ADSP2V2 memory layout where they correspond to {XM, XM, YM}. But this circular buffer could be used on other types of DSP core that have different memory region types. Also the names and description of the size fields were not very clear. The field names and descriptions have been changed to be generic and not imply any particular memory types. This patch updates the wm_adsp driver to the new field names. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f61656070225..7ae10c632614 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct { }; struct wm_adsp_buffer { - __be32 X_buf_base; /* XM base addr of first X area */ - __be32 X_buf_size; /* Size of 1st X area in words */ - __be32 X_buf_base2; /* XM base addr of 2nd X area */ - __be32 X_buf_brk; /* Total X size in words */ - __be32 Y_buf_base; /* YM base addr of Y area */ - __be32 wrap; /* Total size X and Y in words */ + __be32 buf1_base; /* Base addr of first buffer area */ + __be32 buf1_size; /* Size of buf1 area in DSP words */ + __be32 buf2_base; /* Base addr of 2nd buffer area */ + __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */ + __be32 buf3_base; /* Base addr of buf3 area */ + __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */ __be32 high_water_mark; /* Point at which IRQ is asserted */ __be32 irq_count; /* bits 1-31 count IRQ assertions */ __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ @@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def { static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base), - .size_offset = HOST_BUFFER_FIELD(X_buf_size), + .base_offset = HOST_BUFFER_FIELD(buf1_base), + .size_offset = HOST_BUFFER_FIELD(buf1_size), }, { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base2), - .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + .base_offset = HOST_BUFFER_FIELD(buf2_base), + .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size), }, { .mem_type = WMFW_ADSP2_YM, - .base_offset = HOST_BUFFER_FIELD(Y_buf_base), - .size_offset = HOST_BUFFER_FIELD(wrap), + .base_offset = HOST_BUFFER_FIELD(buf3_base), + .size_offset = HOST_BUFFER_FIELD(buf_total_size), }, }; -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate 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 e3a360b8cdede74d25807fc405e5d8bfb025692f Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.cirrus.com> Date: Fri, 19 Oct 2018 13:25:16 +0100 Subject: [PATCH] ASoC: wm_adsp: Log addresses as 8 digits in wm_adsp_buffer_populate Increase the address value width in the debug log from 4 digits to 8 digits to allow for DSP cores with larger memory address ranges. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 7ae10c632614..a53dc174bbf0 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3345,7 +3345,7 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) region->cumulative_size = offset; adsp_dbg(buf->dsp, - "region=%d type=%d base=%04x off=%04x size=%04x\n", + "region=%d type=%d base=%08x off=%08x size=%08x\n", i, region->mem_type, region->base_addr, region->offset, region->cumulative_size); } -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: tegra_sgtl5000: fix device_node refcounting" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: tegra_sgtl5000: fix device_node refcounting 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 a85227da2dcc291b762c8482a505bc7d0d2d4b07 Mon Sep 17 00:00:00 2001 From: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Date: Tue, 16 Oct 2018 12:47:29 +0200 Subject: [PATCH] ASoC: tegra_sgtl5000: fix device_node refcounting Similar to the following: commit 4321723648b0 ("ASoC: tegra_alc5632: fix device_node refcounting") commit 7c5dfd549617 ("ASoC: tegra: fix device_node refcounting") Signed-off-by: Marcel Ziswiler <marcel.ziswiler(a)toradex.com> Acked-by: Jon Hunter <jonathanh(a)nvidia.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/tegra/tegra_sgtl5000.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/tegra/tegra_sgtl5000.c b/sound/soc/tegra/tegra_sgtl5000.c index 45a4aa9d2a47..901457da25ec 100644 --- a/sound/soc/tegra/tegra_sgtl5000.c +++ b/sound/soc/tegra/tegra_sgtl5000.c @@ -149,14 +149,14 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) dev_err(&pdev->dev, "Property 'nvidia,i2s-controller' missing/invalid\n"); ret = -EINVAL; - goto err; + goto err_put_codec_of_node; } tegra_sgtl5000_dai.platform_of_node = tegra_sgtl5000_dai.cpu_of_node; ret = tegra_asoc_utils_init(&machine->util_data, &pdev->dev); if (ret) - goto err; + goto err_put_cpu_of_node; ret = snd_soc_register_card(card); if (ret) { @@ -169,6 +169,13 @@ static int tegra_sgtl5000_driver_probe(struct platform_device *pdev) err_fini_utils: tegra_asoc_utils_fini(&machine->util_data); +err_put_cpu_of_node: + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; +err_put_codec_of_node: + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; err: return ret; } @@ -183,6 +190,12 @@ static int tegra_sgtl5000_driver_remove(struct platform_device *pdev) tegra_asoc_utils_fini(&machine->util_data); + of_node_put(tegra_sgtl5000_dai.cpu_of_node); + tegra_sgtl5000_dai.cpu_of_node = NULL; + tegra_sgtl5000_dai.platform_of_node = NULL; + of_node_put(tegra_sgtl5000_dai.codec_of_node); + tegra_sgtl5000_dai.codec_of_node = NULL; + return ret; } -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: sta32x: Add support for XTI clock" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: sta32x: Add support for XTI clock 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 fce9ec954a8af7e04cbf5b9daa8bec9c1df5cfe6 Mon Sep 17 00:00:00 2001 From: Daniel Mack <daniel(a)zonque.org> Date: Wed, 17 Oct 2018 13:37:03 +0200 Subject: [PATCH] ASoC: sta32x: Add support for XTI clock The STA32x chips feature an XTI clock input that needs to be stable before the reset signal is released. Therefore, the chip driver needs to get a handle to the clock. Instead of relying on other parts of the system to enable the clock, let the codec driver grab a handle itself. In order to keep existing boards working, clock support is made optional. Signed-off-by: Daniel Mack <daniel(a)zonque.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/st,sta32x.txt | 6 ++++ sound/soc/codecs/sta32x.c | 28 +++++++++++++++++++ 2 files changed, 34 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/st,sta32x.txt b/Documentation/devicetree/bindings/sound/st,sta32x.txt index ff4a685a4303..52265fb757c5 100644 --- a/Documentation/devicetree/bindings/sound/st,sta32x.txt +++ b/Documentation/devicetree/bindings/sound/st,sta32x.txt @@ -19,6 +19,10 @@ Required properties: Optional properties: + - clocks, clock-names: Clock specifier for XTI input clock. + If specified, the clock will be enabled when the codec is probed, + and disabled when it is removed. The 'clock-names' must be set to 'xti'. + - st,output-conf: number, Selects the output configuration: 0: 2-channel (full-bridge) power, 2-channel data-out 1: 2 (half-bridge). 1 (full-bridge) on-board power @@ -79,6 +83,8 @@ Example: codec: sta32x@38 { compatible = "st,sta32x"; reg = <0x1c>; + clocks = <&clock>; + clock-names = "xti"; reset-gpios = <&gpio1 19 0>; power-down-gpios = <&gpio1 16 0>; st,output-conf = /bits/ 8 <0x3>; // set output to 2-channel diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 22de1593443c..f753d2db0a5a 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -21,6 +21,7 @@ #include <linux/module.h> #include <linux/moduleparam.h> #include <linux/init.h> +#include <linux/clk.h> #include <linux/delay.h> #include <linux/pm.h> #include <linux/i2c.h> @@ -142,6 +143,7 @@ static const char *sta32x_supply_names[] = { /* codec private data */ struct sta32x_priv { struct regmap *regmap; + struct clk *xti_clk; struct regulator_bulk_data supplies[ARRAY_SIZE(sta32x_supply_names)]; struct snd_soc_component *component; struct sta32x_platform_data *pdata; @@ -879,6 +881,18 @@ static int sta32x_probe(struct snd_soc_component *component) struct sta32x_priv *sta32x = snd_soc_component_get_drvdata(component); struct sta32x_platform_data *pdata = sta32x->pdata; int i, ret = 0, thermal = 0; + + sta32x->component = component; + + if (sta32x->xti_clk) { + ret = clk_prepare_enable(sta32x->xti_clk); + if (ret != 0) { + dev_err(component->dev, + "Failed to enable clock: %d\n", ret); + return ret; + } + } + ret = regulator_bulk_enable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); if (ret != 0) { @@ -981,6 +995,9 @@ static void sta32x_remove(struct snd_soc_component *component) sta32x_watchdog_stop(sta32x); regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); + + if (sta32x->xti_clk) + clk_disable_unprepare(sta32x->xti_clk); } static const struct snd_soc_component_driver sta32x_component = { @@ -1097,6 +1114,17 @@ static int sta32x_i2c_probe(struct i2c_client *i2c, } #endif + /* Clock */ + sta32x->xti_clk = devm_clk_get(dev, "xti"); + if (IS_ERR(sta32x->xti_clk)) { + ret = PTR_ERR(sta32x->xti_clk); + + if (ret == -EPROBE_DEFER) + return ret; + + sta32x->xti_clk = NULL; + } + /* GPIOs */ sta32x->gpiod_nreset = devm_gpiod_get_optional(dev, "reset", GPIOD_OUT_LOW); -- 2.19.0.rc2

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[alsa-devel] Applied "ASoC: rsnd: enable TDM settings for SSI parent" to the asoc tree
by Mark Brown 19 Oct '18

19 Oct '18

The patch ASoC: rsnd: enable TDM settings for SSI parent 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 6817d7593f3e3c8a9c11e7a07cb5646c70371f0a Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Date: Wed, 17 Oct 2018 01:55:37 +0000 Subject: [PATCH] ASoC: rsnd: enable TDM settings for SSI parent Some SSIs are sharing each pins (= WS/CLK pin for playback/capture). Then, SSI parent needs control WS/CLK setting for SSI slave. In such case, SSI parent needs TDM settings if SSI slave is working as TDM mode. But it is not cared in current driver. It can't capture TDM sound without this patch if SSIs were pin sharing. This patch is tested on R-Car H3 ulcb-kf board, SSI3/4 with TDM sound. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sh/rcar/ssi.c | 21 +++++++++++---------- 1 file changed, 11 insertions(+), 10 deletions(-) diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index 3adcc4f778f7..b42a0e0feab7 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -399,6 +399,17 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, if (rdai->sys_delay) cr_own |= DEL; + /* + * TDM Mode + * see + * rsnd_ssiu_init_gen2() + */ + wsr = ssi->wsr; + if (is_tdm) { + wsr |= WS_MODE; + cr_own |= CHNL_8; + } + /* * We shouldn't exchange SWSP after running. * This means, parent needs to care it. @@ -429,16 +440,6 @@ static void rsnd_ssi_config_init(struct rsnd_mod *mod, cr_mode = DIEN; /* PIO : enable Data interrupt */ } - /* - * TDM Extend Mode - * see - * rsnd_ssiu_init_gen2() - */ - wsr = ssi->wsr; - if (is_tdm) { - wsr |= WS_MODE; - cr_own |= CHNL_8; - } init_end: ssi->cr_own = cr_own; ssi->cr_mode = cr_mode; -- 2.19.0.rc2

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[alsa-devel] [PATCH 1/2] ASoC: wm_adsp: Rename memory fields in wm_adsp_buffer
by Richard Fitzgerald 19 Oct '18

19 Oct '18

The wm_adsp_buffer struct is the control header of a circular buffer used to transfer data from the firmware over the control interface to an ALSA compressed stream. The original names of the fields pointing to the data buffer were based on ADSP2V2 memory layout where they correspond to {XM, XM, YM}. But this circular buffer could be used on other types of DSP core that have different memory region types. Also the names and description of the size fields were not very clear. The field names and descriptions have been changed to be generic and not imply any particular memory types. This patch updates the wm_adsp driver to the new field names. Signed-off-by: Richard Fitzgerald <rf(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f61656070225..7ae10c632614 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -311,12 +311,12 @@ struct wm_adsp_alg_xm_struct { }; struct wm_adsp_buffer { - __be32 X_buf_base; /* XM base addr of first X area */ - __be32 X_buf_size; /* Size of 1st X area in words */ - __be32 X_buf_base2; /* XM base addr of 2nd X area */ - __be32 X_buf_brk; /* Total X size in words */ - __be32 Y_buf_base; /* YM base addr of Y area */ - __be32 wrap; /* Total size X and Y in words */ + __be32 buf1_base; /* Base addr of first buffer area */ + __be32 buf1_size; /* Size of buf1 area in DSP words */ + __be32 buf2_base; /* Base addr of 2nd buffer area */ + __be32 buf1_buf2_size; /* Size of buf1+buf2 in DSP words */ + __be32 buf3_base; /* Base addr of buf3 area */ + __be32 buf_total_size; /* Size of buf1+buf2+buf3 in DSP words */ __be32 high_water_mark; /* Point at which IRQ is asserted */ __be32 irq_count; /* bits 1-31 count IRQ assertions */ __be32 irq_ack; /* acked IRQ count, bit 0 enables IRQ */ @@ -393,18 +393,18 @@ struct wm_adsp_buffer_region_def { static const struct wm_adsp_buffer_region_def default_regions[] = { { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base), - .size_offset = HOST_BUFFER_FIELD(X_buf_size), + .base_offset = HOST_BUFFER_FIELD(buf1_base), + .size_offset = HOST_BUFFER_FIELD(buf1_size), }, { .mem_type = WMFW_ADSP2_XM, - .base_offset = HOST_BUFFER_FIELD(X_buf_base2), - .size_offset = HOST_BUFFER_FIELD(X_buf_brk), + .base_offset = HOST_BUFFER_FIELD(buf2_base), + .size_offset = HOST_BUFFER_FIELD(buf1_buf2_size), }, { .mem_type = WMFW_ADSP2_YM, - .base_offset = HOST_BUFFER_FIELD(Y_buf_base), - .size_offset = HOST_BUFFER_FIELD(wrap), + .base_offset = HOST_BUFFER_FIELD(buf3_base), + .size_offset = HOST_BUFFER_FIELD(buf_total_size), }, }; -- 2.11.0

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[alsa-devel] AGL / Questions about the upcoming features of Alsa
by Thierry Bultel 19 Oct '18

19 Oct '18

Hi, My name is Thierry Bultel from Iot.bzh, and I am part of the responsible team for the audio framework (aka, 4A) in Automotive Grade Linux. For the ones interested in the audio architecture in AGL (and the ones that will read my list of questions below), these are useful links to follow: New AVIRT driver in replacement of SND_ALOOP: https://iot.bzh/download/public/2018/amm-dresden/ALSA_Virtual_Dynamic_Sound… 4A framework kickstart: https://iot.bzh/download/public/2018/amm-dresden/4a-kickstart-Loic-IoTbzh-D… In AGL, we have used the Alsa framework intensively. I wish I could meet the Alsa team next week in Edimburg but unfortunately I will not be available at that date. Other colleagues of mine will be there, however. Here are some points that I would like to be addressed at this meeting: - ioplug PCMs: There are known issues about potential (effective !) deadlocks when using ioplug PCMs. We have worked around this by setting LIBASOUND_THREAD_SAFE=0 Is there any pending work to avoid doing this and fix the bug ? - capture dispatching According to the experts, what would be the most effective way to implement a capture dispatcher ? The need is to have several applications listening to the same microphone. Is JACK the most suitable solution (at this time) ? (more precisely, this could behave as a 1to1 switch or 1toN, by duplicating the audio stream). A kernel solution based on AVIRT with routing could easily be done, too. - Documentation: Is there any pending rework of the documentation for having it a little bit more detailed ? For instance, having a page explaining the roles of period_size and buffer_size, would be great, since it impossible to perform sound playback correctly in an application, without understanding those notions. - Could something like a dynamic setting of the parameters of the rate converter plugin, become a reality one day ? - What is you overall mind about the AVIRT driver that we (with Fiberdyne) are developing ? - Is there a release features roadmap for Alsa (kernel & library) ? - What is your feeling about PipeWire, which could eventually replace PulseAudio (and seems to also invade the JACK land) ? Thanks ! Thierry Bultel

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[alsa-devel] Audio miniconference schedule
by Mark Brown 18 Oct '18

18 Oct '18

As previously announced there's an audio miniconference happening on the 21st in Edinburgh, meeting around 9am and getting started once enough people have arrived (Cirrus people, please let me know if we need to start later, some people have said they'll be arriving late so we won't wait for everyone): https://blog.sirena.org.uk/2018/09/19/2018-linux-audio-miniconfernce (yes there is a typo in the URL). A number of people have proposed topics so I just went through them and pulled them into a proposed agenda, revised again after further feedback: - Introductions - Community governence issues (Takashi) - Virtualization (Liam) - AVB (Liam) - Testing (Takashi) - PCM improvements (Takashi) - Granularity - Timestamping - SoundWire (Liam) - Integrating the ALSA control core (Sakamoto-san) - Obsolete 'dimen' member in container for information to an element - Remove limitation on a container for value array to an element - User-defined element set left by finished process - Expand restriction to the number of user-defined control element set per sound card - Another approach to produce language bindings by GObject Introspection (Sakamoto-san) - ASoC rate domains/core refactoring (Charles) - ASoC/ALSA core merging (Takashi) - DSPs (Liam) - Sound Open Firmware (Liam) - Byte controls (Vinod/Patrick) - Calibration loading (Patrick) - Exposing graph at runtime (Patrick) - Batching commands (Patrick) - AGL (Stephane/Ronan) I hope I've not missed anything people suggested, the schedule is now looking pretty full but I'm hoping some of these topics can be relatively brief. I've dropped Y2038 from the expliit schedule, since Arnd Bergmann (who's been driving a lot of the general Y2038 work) will be in Edinburgh and may be free to join I'm suggesting that if he's free we cover that either first thing or just after lunch and otherwise Takashi and I will try to coordinate with him during the Maintainer's Summit on Monday (unless others have a great enthusiasm for the subject). What do people think? Any other topics? There's still time to sign up if you're interested in attending, you can sign up right up to the day of the event though registering early would help a lot with logistics.

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[alsa-devel] [PATCH] conf/ucm/Dell-WD15-Dock: Fix incorrect device names
by Takashi Iwai 18 Oct '18

18 Oct '18

The device name string for Dell WD15 (and its variants) dock is set as "WD15Dock", while the actual device name to be used is "Dock". Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=1112292 Fixes: 8ebb40c96970 ("conf/ucm: Add a UCM profile for Dell WD15 Dock USB-audio") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/conf/ucm/Dell-WD15-Dock/HiFi.conf | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf index b07f34400086..1b6bd21237f1 100644 --- a/src/conf/ucm/Dell-WD15-Dock/HiFi.conf +++ b/src/conf/ucm/Dell-WD15-Dock/HiFi.conf @@ -3,7 +3,7 @@ SectionDevice."Headphone" { Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,0" + PlaybackPCM "hw:Dock,0" } } @@ -12,7 +12,7 @@ SectionDevice."LineOut" { Value { PlaybackChannels "2" - PlaybackPCM "hw:WD15Dock,1" + PlaybackPCM "hw:Dock,1" } } @@ -21,6 +21,6 @@ SectionDevice."Mic" { Value { CaptureChannels "2" - CapturePCM "hw:WD15Dock,0" + CapturePCM "hw:Dock,0" } } -- 2.19.0

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[alsa-devel] [PATCH - Intervals 1/1] interval: Interpret (x x+1] correctly and return x+1
by twischer＠de.adit-jv.com 18 Oct '18

18 Oct '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this change an interval of (x x+1] will be interpreted as an empty interval but the right value would be x+1. This leads to a failing snd_pcm_hw_params() call which returns -EINVAL. An example issue log is given in the following: snd_pcm_hw_params failed with err -22 (Invalid argument) ACCESS: MMAP_NONINTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 16 CHANNELS: 1 RATE: 16000 PERIOD_TIME: (15999 16000] PERIOD_SIZE: (255 256] PERIOD_BYTES: (510 512] PERIODS: [2 3) BUFFER_TIME: 32000 BUFFER_SIZE: 512 BUFFER_BYTES: 1024 In case of (x x+1) we have to interpret it anyway as a single value of x to compensate rounding issues. For example the period size will result in an interval of (352 353) when the period time is 16ms and the sample rate 22050 Hz (16ms * 22,05 kHz = 352,8 frames). But 352 has to be chosen to allow a buffer size of 705 (32ms * 22,05 kHz = 705,6 frames) which has to be >= 2x period size to avoid Xruns. The buffer size will not end up with an interval of (705 706) similar to the period size because snd_pcm_rate_hw_refine_cchange() calls snd_interval_floor() for the buffer size. Therefore this value will be interpreted as an integer interval instead of a real interval further on. This issue seems to exist since the change of 9bb985c38 ("pcm: snd_interval_refine_first/last: exclude value only if also excluded before") Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/interval_inline.h b/src/pcm/interval_inline.h index a68e292..5e59d72 100644 --- a/src/pcm/interval_inline.h +++ b/src/pcm/interval_inline.h @@ -51,12 +51,14 @@ INTERVAL_INLINE int snd_interval_single(const snd_interval_t *i) { assert(!snd_interval_empty(i)); return (i->min == i->max || - (i->min + 1 == i->max && i->openmax)); + (i->min + 1 == i->max && (i->openmin || i->openmax))); } INTERVAL_INLINE int snd_interval_value(const snd_interval_t *i) { assert(snd_interval_single(i)); + if (!i->openmax) + return i->max; return i->min; } -- 2.7.4

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