November 2016 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/8] ASoC: rsnd: DMA power fix and IOMMU support
by Kuninori Morimoto 22 Nov '16

22 Nov '16

Hi Mark These patches are for Renesas sound DMA. 1st feature is that it can power save for Audio DMAC. 2nd feature is that it can support IOMMU 1) - 5) : cleanup for 6) - 8) 6) - 7) : Audio DMAC power save 8) : IOMMU support Kuninori Morimoto (8): 1) ASoC: rsnd: depends on OF 2) ASoC: rsnd: remove non DT support for DMA 3) ASoC: rsnd: don't use devm_request_irq() for SSI 4) ASoC: rsnd: remove rsnd_dma_detach() 5) ASoC: rsnd: don't call unneeded of_node_put() on dma.c 6) ASoC: rsnd: add nolock_start/stop callback 7) ASoC: rsnd: Request/Release DMA channel each time 8) ASoC: rsnd: use dma_sync_single_for_xxx() for IOMMU sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/core.c | 30 +++++++++++++- sound/soc/sh/rcar/dma.c | 304 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++------------------------------------------------- sound/soc/sh/rcar/rsnd.h | 18 ++++++-- sound/soc/sh/rcar/src.c | 2 +- sound/soc/sh/rcar/ssi.c | 22 +++++----- 6 files changed, 248 insertions(+), 129 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: wm_adsp: Check return value from wm_adsp_buffer_init
by Charles Keepax 22 Nov '16

22 Nov '16

We are not checking the return value from a call to wm_adsp_buffer_init it looks like this used to be returned at the bottom of the function but probably got missed when more error paths were added. This patch adds the appropriate error check. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 6ccb731..7e4f8dd 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -2580,8 +2580,11 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, mutex_lock(&dsp->pwr_lock); - if (wm_adsp_fw[dsp->fw].num_caps != 0) + if (wm_adsp_fw[dsp->fw].num_caps != 0) { ret = wm_adsp_buffer_init(dsp); + if (ret < 0) + goto err; + } mutex_unlock(&dsp->pwr_lock); -- 2.1.4

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[alsa-devel] [PATCH 0/7] topology: Add support for physical DAIs
by mengdong.lin＠linux.intel.com 22 Nov '16

22 Nov '16

From: Mengdong Lin <mengdong.lin(a)linux.intel.com> This is the last series for topology user space atm. This series adds support for physical DAIs and includes a few minor fixes. All ABI updates here can be supported by kernel in a backward compatible way. Guneshwor Singh (3): topology: ABI - Define new types for physical DAI topology: Support configuring physical DAIs by C API topology: Export physical DAIs to the binary for kernel Mengdong Lin (4): topology: ABI - Update manifest and pump ABI version to 5 topology: Parse physical DAIs in text conf file topology: ABI - Add voice wake up flag for DAI links topology: Revise document and comments for ABI v5 include/sound/asoc.h | 37 +++++++- include/topology.h | 94 ++++++++++++------- src/topology/builder.c | 11 +++ src/topology/data.c | 4 + src/topology/elem.c | 4 + src/topology/parser.c | 16 ++++ src/topology/pcm.c | 223 +++++++++++++++++++++++++++++++++++++++++++++- src/topology/tplg_local.h | 7 ++ 8 files changed, 362 insertions(+), 34 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v4 00/12] ASoC: add OF graph base simple-card
by Kuninori Morimoto 22 Nov '16

22 Nov '16

Hi Rob, Mark These are v4 of OF graph base simple-card patch-set. v3 patch had ALSA SoC side prepare patches as [1/xx] - [5/xx], but these are already accepted by Mark, thus, this v4 doesn't include these. Mainly, v4 solved non-OF case compile error which was reported by kbuild. 1) - 5) : OF graph new feature 6) - 12) : OF graph base simple-card (depends on above 2 patch-set) Kuninori Morimoto (12): 1) of_graph: add of_graph_get_remote_endpoint() 2) of_graph: add of_graph_get_port_parent() 3) of_graph: add of_graph_get_top_port() 4) of_graph: add for_each_of_port() / for_each_of_endpoint_in_port() 5) of_graph: add of_graph_get_port/endpoint_count() 6) ASoC: simple-card-utils: add asoc_simple_card_parse_graph_dai() 7) ASoC: simple-card-utils: add asoc_simple_card_try_to_probe_graph_card() 8) ASoC: simple-card-utils: adjust for graph on asoc_simple_card_parse_card_name 9) ASoC: add simple-graph-card document 10) ASoC: add simple-graph-card support 11) ASoC: add simple-graph-scu-card document 12) ASoC: add simple-graph-scu-card support .../bindings/sound/simple-graph-card.txt | 65 +++ .../bindings/sound/simple-graph-scu-card.txt | 65 +++ drivers/of/base.c | 160 ++++++- include/linux/of_graph.h | 59 +++ include/sound/simple_card_utils.h | 19 + sound/soc/generic/Kconfig | 15 + sound/soc/generic/Makefile | 4 + sound/soc/generic/simple-card-utils.c | 98 ++++- sound/soc/generic/simple-card.c | 2 +- sound/soc/generic/simple-graph-card.c | 462 +++++++++++++++++++++ sound/soc/generic/simple-graph-scu-card.c | 417 +++++++++++++++++++ sound/soc/generic/simple-scu-card.c | 2 +- 12 files changed, 1355 insertions(+), 13 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-card.txt create mode 100644 Documentation/devicetree/bindings/sound/simple-graph-scu-card.txt create mode 100644 sound/soc/generic/simple-graph-card.c create mode 100644 sound/soc/generic/simple-graph-scu-card.c -- 1.9.1

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[alsa-devel] [PATCH 00/19] ASoC: Convert to module_i2c/spi_driver()
by Kuninori Morimoto 21 Nov '16

21 Nov '16

Hi Mark These patches uses module_i2c/spi_driver() on each driver to cleanup driver entries. I confirmed compile test only. 1) - 4): remove #ifdef CONFIG_I2C, add depends on CONFIG_I2C, and uses module_i2c_driver 5) - 19): uses module_i2c/spi_driver() and removed module_init/exit() Kuninori Morimoto (19): 1) ASoC: uda1380: Convert to module_i2c_driver() 2) ASoC: wm9081: Convert to module_i2c_driver() 3) ASoC: wm8580: Convert to module_i2c_driver() 4) ASoC: wm8523: Convert to module_i2c_driver() 5) ASoC: wm8510: Convert to module_i2c/spi_driver() 6) ASoC: wm8995: Convert to module_i2c/spi_driver() 7) ASoC: wm8988: Convert to module_i2c/spi_driver() 8) ASoC: wm8985: Convert to module_i2c/spi_driver() 9) ASoC: wm8983: Convert to module_i2c/spi_driver() 10) ASoC: wm8900: Convert to module_i2c/spi_driver() 11) ASoC: wm8776: Convert to module_i2c/spi_driver() 12) ASoC: wm8753: Convert to module_i2c/spi_driver() 13) ASoC: wm8750: Convert to module_i2c/spi_driver() 14) ASoC: wm8741: Convert to module_i2c/spi_driver() 15) ASoC: wm8731: Convert to module_i2c/spi_driver() 16) ASoC: wm8737: Convert to module_i2c/spi_driver() 17) ASoC: wm8728: Convert to module_i2c/spi_driver() 18) ASoC: wm8711: Convert to module_i2c/spi_driver() 19) ASoC: da7210: Convert to module_i2c/spi_driver() sound/soc/codecs/Kconfig | 2 ++ sound/soc/codecs/da7210.c | 30 ++---------------------------- sound/soc/codecs/uda1380.c | 22 +--------------------- sound/soc/codecs/wm8510.c | 34 ++-------------------------------- sound/soc/codecs/wm8523.c | 24 +----------------------- sound/soc/codecs/wm8580.c | 25 +------------------------ sound/soc/codecs/wm8711.c | 34 ++-------------------------------- sound/soc/codecs/wm8728.c | 34 ++-------------------------------- sound/soc/codecs/wm8731.c | 34 ++-------------------------------- sound/soc/codecs/wm8737.c | 34 ++-------------------------------- sound/soc/codecs/wm8741.c | 34 ++-------------------------------- sound/soc/codecs/wm8750.c | 34 ++-------------------------------- sound/soc/codecs/wm8753.c | 34 ++-------------------------------- sound/soc/codecs/wm8776.c | 34 ++-------------------------------- sound/soc/codecs/wm8900.c | 34 ++-------------------------------- sound/soc/codecs/wm8983.c | 35 ++--------------------------------- sound/soc/codecs/wm8985.c | 35 ++--------------------------------- sound/soc/codecs/wm8988.c | 35 ++--------------------------------- sound/soc/codecs/wm8995.c | 37 ++----------------------------------- sound/soc/codecs/wm9081.c | 2 -- 20 files changed, 35 insertions(+), 552 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH] ASoC: wm_adsp: Only write shutdown controls for active firmwares
by Charles Keepax 21 Nov '16

21 Nov '16

The control list may contain shutdown controls for firmwares that are not currently active, attempting to write this will at best fail. To avoid this issue we skip any control that is not active. Fixes: commit f4f0c4c60c39 ("ASoC: wm_adsp: Signal firmware shutdown through event control") Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm_adsp.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 4fb6e2f..6ccb731 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1194,6 +1194,9 @@ static void wm_adsp_signal_event_controls(struct wm_adsp *dsp, if (ctl->type != WMFW_CTL_TYPE_HOSTEVENT) continue; + if (!ctl->enabled) + continue; + ret = wm_coeff_write_acked_control(ctl, event); if (ret) adsp_warn(dsp, -- 2.1.4

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[alsa-devel] [RFC 0/4] ALSA controls management using index/device/sub-devices fields
by Arnaud Pouliquen 21 Nov '16

21 Nov '16

I tried to summarize my observation based on discussions concerning the instantiation of the ALSA controls, initiate in following thread: [PATCH] ASoC: core: allow control index different from 0 https://www.spinics.net/lists/ALSA-devel/msg55415.html My test environment: sti platform, 3 playback devices with associated instances of controls card 0: STIB2120 [STI-B2120], device 0: Uni Player #0 (HDMI)-i2s-hifi numid=1,iface=PCM,name='ELD' numid=3,iface=PCM,name='IEC958 Playback Default' numid=2,iface=PCM,name='PCM Playback Oversampling Freq. Adjustment' card 0: STIB2120 [STI-B2120], device 1: Uni Player #2 (DAC)-sas-dai-dac numid=4,iface=PCM,name='PCM Playback Oversampling Freq. Adjustment',device=2 card 0: STIB2120 [STI-B2120], device 2: Uni Player #3 (SPDIF)-sas-dai- spdif-out numid=6,iface=PCM,name='IEC958 Playback Default',device=2 numid=5,iface=PCM,name='PCM Playback Oversampling Freq. Adjustment',device=3 1) From user land point of view 1-1) How to Instantiate generic ALSA control The point is the handling of multi instance of generic ALSA controls. In this case "prefix" can not be used. Controls have to be identified by a combination of device/sub-device/index Examples of controls that seem generics: iface=PCM,name='Capture Channel Map' iface=PCM,name='Playback Channel Map' iface=PCM,name='ELD' iface=MIXER,name='Master Playback Switch' iface=MIXER,name='Master Playback Volume' iface=MIXER,name='Capture Switch' iface=MIXER,name='Capture Volume' iface=MIXER/PCM,name='IEC958 Playback Con Mask' iface=MIXER/PCM,name='IEC958 Playback Pro Mask' iface=MIXER/PCM,name='IEC958 Playback Default' iface=MIXER/PCM,name='IEC958 Playback Switch' 1-2) Different ways of using control instantiation Here is a short benchmark on applications using ALSA controls: - iecset and "amixer scontrols" base control instantiation on "index". - GStreamer and pulseaudio seems based on "device" (through the ALSA card conf files). - "amixer controls" support both. I supposse that this is the reason why "device" and "index fields are both fixed in HDA-intel.cong aligned with driver. 1-3) Questions: - If i consider description provided by Takashi Sakamoto in thread: https://www.spinics.net/lists/ALSA-devel/msg55451.html Should we always apply this rules? - MIXER control type: as it is not linked to PCM device but card, "index" is used to instantiate control. - PCM control type: as it is linked to PCM device, "device/subdevice" is used to instantiate control. Or could we consider that a control can be instantiated using device/subdevice fields and/or index fields? - Concerning IEC controls, should we consider them as PCM or MIXER type controls? In current implementations both are used... 2) From driver point of view 2-1) None SOC drivers: Today solution seems to base indexation on both "index" and "device" number. Device indexation seems not linked to PCM device number Example in ea9b43addc4d ("ALSA: hda - Fix broken workaround for HDMI/SPDIF conflicts"). http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ea9b… => The drawback is that each driver has to implement it. Could be nice to have in generic code.... 2-2) SOC drivers: - it is not possible to use "index" using helper functions as snd_soc_cnew forces index to 0. Solution could be to implement in soc_core as done in HDA driver. => duplication of the code in ASoC and None ASoC drivers. - Relation chip between control and PCM device. if rules mentioned in 1-3 should be respected, need to link control to PCM device. This point seems quite tricky... At least DAI controls can be simply linked to PCM device during DAI-link probing. 3) Patches proposed: Based on these observations, here are some patches that i tested in my environment. There are complementary, and could answer to some limitations mentioned above. 3-1) Alsa-utils patch - iecset: allow to select control with device and sub-device numbers This patch allows to access to 2 iec controls differentiated by device/sub-devices numbers => For me, this patch is mandatory to be able to address the ASoC IEC controls, in case of no fix is implemented to allows index field update in ASoC. 3-2) Alsa driver patches - ASoC: core: allow PCM control binding to PCM device Add relationship between DAIs PCM controls and PCM device. - ALSA: control: increment index field for duplicated control. Generic implementation of the patch proposed in HDA driver (http://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/?id=ea9b…) - ASoC: sti: use bind_pcm_ctl implementation of bind_pcm_ctl for sti driver. Best Regards, Arnaud -- 1.9.1

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[alsa-devel] [PATCH] ASoC: mioa701_wm9713: add missing white space in dev_err message
by Colin King 19 Nov '16

19 Nov '16

From: Colin Ian King <colin.king(a)canonical.com> There is a missing whitespace in the dev_err message between "will" and "lead". Add the whitespace. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/pxa/mioa701_wm9713.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index d1661fa..0fe0abe 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -187,7 +187,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev) mioa701.dev = &pdev->dev; rc = devm_snd_soc_register_card(&pdev->dev, &mioa701); if (!rc) - dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" + dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will " "lead to overheating and possible destruction of your device." " Do not use without a good knowledge of mio's board design!\n"); return rc; -- 2.10.2

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[alsa-devel] [PATCH] ASoC: intel: mfld: Make static string arrays 'const 'char * const []'
by Lars-Peter Clausen 18 Nov '16

18 Nov '16

const char * const [] is the preferred type for static string arrays since this states explicitly that the individual entries are not going to be changed. Due to limitations in the ASoC API it was not possible to use it for enum text arrays. Commit 87023ff74b23 ("ASoC: Declare const properly for enum texts") changed this, but most drivers still use 'const char * []' as the type for their enum text arrays. Change these occurrences of 'static * const char * []' to 'static const char * const []'. The conversion was done automatically using the following coccinelle semantic patch: // <smpl> @disable optional_qualifier@ identifier s; @@ static -const char * +const char * const s[] = ...; // </smpl> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/intel/boards/mfld_machine.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/mfld_machine.c b/sound/soc/intel/boards/mfld_machine.c index 34f46c7..4e08885 100644 --- a/sound/soc/intel/boards/mfld_machine.c +++ b/sound/soc/intel/boards/mfld_machine.c @@ -81,9 +81,9 @@ static struct snd_soc_jack_zone mfld_zones[] = { }; /* sound card controls */ -static const char *headset_switch_text[] = {"Earpiece", "Headset"}; +static const char * const headset_switch_text[] = {"Earpiece", "Headset"}; -static const char *lo_text[] = {"Vibra", "Headset", "IHF", "None"}; +static const char * const lo_text[] = {"Vibra", "Headset", "IHF", "None"}; static const struct soc_enum headset_enum = SOC_ENUM_SINGLE_EXT(2, headset_switch_text); -- 2.1.4

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[alsa-devel] [PATCH v3 0/3] ASoC driver for the TSE-850
by Peter Rosin 18 Nov '16

18 Nov '16

Hi! v2 -> v3 changes: - update the preparatory atmel_ssc_dai patch with error checking and fallback handling of the CMR_DIV divider in addition to the new fallback handling of TCMR_PERIOD/RCMR_PERIOD from v2. v1 -> v2 changes: - new subject for the bindings patch to make it easier to find - move the driver to the sound/soc/atmel directory - add cached values for add/loop1/loop2 to avoid gets from output gpios - use _cansleep when updating gpios - add comment on how the regulator voltage is mapped to the ana enum - drop the .hw_params hook that did set the cpu dai divider and... - ...add a preparatory patch that does this in the cpu dai driver instead - drop .init and set the dapm routes directly in the card struct instead The TSE-850 is an FM Transmitter Station Equipment, designed to generate baseband signals for FM, mainly the DARC subcarrier, but other signals are also possible. This adds a driver for the "sound" bits of the device (quoted since it is normally not used for normal sound output, but that works too of course). I have not provided a patch to add axentia as a devicetree vendor prefix, since such a patch is already pending in an IIO series [1] that seems close to being accepted. However, there are a couple of points that I'm not 100% satisfied with for this driver. First, I do not know how to describe the relays that control if the IN1/IN2 signals are directly routed towards OUT1/OUT2 or if they are routed to the "add" switch. The dapm routing treats this as if the IN1/IN2 signals are always routed to both the "add" switch and to the muxes feeding OUT1/OUT2. This is fine with me since nothing is powered in those sections anyway, so what dapm thinks does not really matter. But it is a wart all the same. Second, there's my comment in tse850_put_mix() when the "add" switch is updated. I believe this update should really happen as a side effect of the call to snd_soc_dapm_mixer_update_power(), so that it happens at the right point compared to other stuff that is powered. But I do not know how to hook that up and instead I flip the switch before the call since it doesn't really matter. I.e., any noise resulting from this badness is negligeble in practice. Cheers, Peter [1] http://lwn.net/Articles/705931 Peter Rosin (3): ASoC: atmel_ssc_dai: if not provided, default to sensible dividers ASoC: tse850: document axentia,tse850-pcm5142 bindings ASoC: atmel: tse850: add ASoC driver for the Axentia TSE-850 .../bindings/sound/axentia,tse850-pcm5142.txt | 88 ++++ MAINTAINERS | 7 + sound/soc/atmel/Kconfig | 10 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel_ssc_dai.c | 83 +++- sound/soc/atmel/atmel_ssc_dai.h | 1 + sound/soc/atmel/tse850-pcm5142.c | 472 +++++++++++++++++++++ 7 files changed, 656 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/axentia,tse850-pcm5142.txt create mode 100644 sound/soc/atmel/tse850-pcm5142.c -- 2.1.4

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