January 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v1] sound: pci: ymfpci: ymfpci_main:- Handle return NULL error from ioremap_nocache
by Arvind Yadav 23 Jan '17

23 Jan '17

Here, If ioremap_nocache will fail. It will return NULL. Kernel can run into a NULL-pointer dereference. This error check will avoid NULL pointer dereference. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- sound/pci/ymfpci/ymfpci_main.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/ymfpci/ymfpci_main.c b/sound/pci/ymfpci/ymfpci_main.c index ffee284..8590c3f 100644 --- a/sound/pci/ymfpci/ymfpci_main.c +++ b/sound/pci/ymfpci/ymfpci_main.c @@ -2392,6 +2392,10 @@ int snd_ymfpci_create(struct snd_card *card, chip->rev = pci->revision; chip->reg_area_phys = pci_resource_start(pci, 0); chip->reg_area_virt = ioremap_nocache(chip->reg_area_phys, 0x8000); + if (!chip->reg_area_virt) { + pci_disable_device(pci); + return -ENOMEM; + } pci_set_master(pci); chip->src441_used = -1; -- 1.9.1

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[alsa-devel] [PATCH][resend] ASoC: simple-card: use devm_get_clk_from_child()
by Kuninori Morimoto 23 Jan '17

23 Jan '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Current simple-card-utils is getting clk by of_clk_get(), but didn't call clk_free(). Now we can use devm_get_clk_from_child() for this purpose. Let's use it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- > Mark Latest linus tree has below which is required by this patch. 71a2f11511b4d1dc8b8e326e10ec6533b534ddf1 ("clk: add devm_get_clk_from_child() API") include/sound/simple_card_utils.h | 11 ++++++----- sound/soc/generic/simple-card-utils.c | 8 ++++---- sound/soc/generic/simple-card.c | 4 ++-- sound/soc/generic/simple-scu-card.c | 4 ++-- 4 files changed, 14 insertions(+), 13 deletions(-) diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 64e90ca..af58d23 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -34,11 +34,12 @@ int asoc_simple_card_set_dailink_name(struct device *dev, int asoc_simple_card_parse_card_name(struct snd_soc_card *card, char *prefix); -#define asoc_simple_card_parse_clk_cpu(node, dai_link, simple_dai) \ - asoc_simple_card_parse_clk(node, dai_link->cpu_of_node, simple_dai) -#define asoc_simple_card_parse_clk_codec(node, dai_link, simple_dai) \ - asoc_simple_card_parse_clk(node, dai_link->codec_of_node, simple_dai) -int asoc_simple_card_parse_clk(struct device_node *node, +#define asoc_simple_card_parse_clk_cpu(dev, node, dai_link, simple_dai) \ + asoc_simple_card_parse_clk(dev, node, dai_link->cpu_of_node, simple_dai) +#define asoc_simple_card_parse_clk_codec(dev, node, dai_link, simple_dai) \ + asoc_simple_card_parse_clk(dev, node, dai_link->codec_of_node, simple_dai) +int asoc_simple_card_parse_clk(struct device *dev, + struct device_node *node, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai); diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index cf02625..4924575 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -98,7 +98,8 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); -int asoc_simple_card_parse_clk(struct device_node *node, +int asoc_simple_card_parse_clk(struct device *dev, + struct device_node *node, struct device_node *dai_of_node, struct asoc_simple_dai *simple_dai) { @@ -111,14 +112,13 @@ int asoc_simple_card_parse_clk(struct device_node *node, * or "system-clock-frequency = <xxx>" * or device's module clock. */ - clk = of_clk_get(node, 0); + clk = devm_get_clk_from_child(dev, node, NULL); if (!IS_ERR(clk)) { simple_dai->sysclk = clk_get_rate(clk); - simple_dai->clk = clk; } else if (!of_property_read_u32(node, "system-clock-frequency", &val)) { simple_dai->sysclk = val; } else { - clk = of_clk_get(dai_of_node, 0); + clk = devm_get_clk_from_child(dev, dai_of_node, NULL); if (!IS_ERR(clk)) simple_dai->sysclk = clk_get_rate(clk); } diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index a385ff6..85b4f18 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -278,11 +278,11 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_clk_cpu(cpu, dai_link, cpu_dai); + ret = asoc_simple_card_parse_clk_cpu(dev, cpu, dai_link, cpu_dai); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_parse_clk_codec(codec, dai_link, codec_dai); + ret = asoc_simple_card_parse_clk_codec(dev, codec, dai_link, codec_dai); if (ret < 0) goto dai_link_of_err; diff --git a/sound/soc/generic/simple-scu-card.c b/sound/soc/generic/simple-scu-card.c index bb86ee0..308ff4c 100644 --- a/sound/soc/generic/simple-scu-card.c +++ b/sound/soc/generic/simple-scu-card.c @@ -128,7 +128,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret) return ret; - ret = asoc_simple_card_parse_clk_cpu(np, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_cpu(dev, np, dai_link, dai_props); if (ret < 0) return ret; @@ -153,7 +153,7 @@ static int asoc_simple_card_dai_link_of(struct device_node *np, if (ret < 0) return ret; - ret = asoc_simple_card_parse_clk_codec(np, dai_link, dai_props); + ret = asoc_simple_card_parse_clk_codec(dev, np, dai_link, dai_props); if (ret < 0) return ret; -- 1.9.1

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[alsa-devel] Nokia N900: mixers changed between 4.6 and 4.10, no longer can use in-call speaker
by Pavel Machek 20 Jan '17

20 Jan '17

Hi! In v4.10 (and probably v4.9, too) I can no longer use the in-call speaker. I can no longer use the wired headset, either. v4.4 (and probably v4.6) works ok. Any ideas? Does wired headset / in-call speaker work for you? "Mono" and "Mono DAC" options are still there.. but something else changed, as alsamixer now shows way many more options (meaning they are shorter?) and I get complains from alsactl: alsactl: set_control:1328: failed to obtain info for control #49 (No such file or directory) ... alsactl: set_control:1328: failed to obtain info for control #229 (No such file or directory) Pavel -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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[alsa-devel] [PATCH v5 0/2] ALSA controls management using index/device/sub-devices fields
by Arnaud Pouliquen 20 Jan '17

20 Jan '17

Previous discussions summary: - Topic of the patchset: Patch concerns ASoc DAI drivers. Aim is to be able to instantiate PCM control using device field according to the associated PCM char device. For this, a relationship between DAIs PCM control and PCM char device needs to be establish. => Proposal is to add field in DAI driver struct to declare PCM controls that need to be linked to PCM character device on DAI link probing. - Limitation of the patchset: This patchset seems only a first step as 2 concerns are been highlighted: - Patch is limited to "static" DAI-link. This not responds to backend (no_pcm dai links) and some none-DAI codecs, that could register PCm controls. In theses cases it is not possible to establish relationship during probe. => No solution proposed for time beeing. - This patchset does not fix conflict of 2 "identical" PCM controls declared by 2 drivers. For instance, a conflict can exist between a PCM control created in DAI driver and the same control declared in a codec driver. As index field is forced to 0, prefix should be used in codec. => To discuss solution in a separate thread as issue already present without this patchset several approaches seem possible like using index field ( control auto indexation in ASoC) or prefix in Naming. V5: - [PATCH v5 1/2] ASoC: core: allow DAI PCM controls bound to PCM device Minor fixes in comment and error message. - [PATCH v5 2/2] ASoC: sti: bind pcm controls to pcm device. No change since V2. V4: - [PATCH v4 1/2] ASoC: core: allow DAI PCM controls bound to PCM device Fix and rework error management in soc_link_dai_pcm_controls function. - [PATCH v4 2/2] ASoC: sti: bind pcm controls to pcm device. No change since V2. V3: - [PATCH v3 1/2] ASoC: core: allow DAI PCM controls bound to PCM device Code optimization based on Takashi Sakamoto suggestion - [RFC V2 3/3] ASoC: hdmi-codec: Example of PCM control bound to PCM device for multi patch suppressed as it was send as example in previous version. Patch is valid but can have impact on existing drivers as control device field value is updated - [PATCH v3 2/2] ASoC: sti: bind pcm controls to pcm device. No change vs V2 V2: http://www.spinics.net/lists/alsa-devel/msg57045.html Aim of this version is to continue discussion on DAI PCM control focused on ASoC drivers. In this V2 implementation in Soc-core is simplified to limit impact on existing code. Update of the RFC V1 based on discussions: - [RFC 4/4] iecset: allow to select control with device and sub-device numbers no more part of the RFC V2, will be discussed in a separate thread - [RFC 2/4] ALSA: control: increment index field for duplicated control. no more part of the RFC V2, no more need as RFC subject is PCM controls - [RFC V2 1/3] ASoC: core: allow DAI PCM controls bound to PCM device Patch reworked from V1 to simplify implementation - Binding is not done for Dai links tagged with no_pcm (DPCM). - no more possibility to add the controls after the DAI link probing. - [RFC V2 2/3] ASoC: sti: bind PCM controls to PCM device. - [RFC V2 3/3] ASoC: hdmi-codec: Example of PCM control bound to PCM device for multi Example of implementation in STI DAI driver and HDMI-codec drivers V1: http://www.spinics.net/lists/alsa-devel/msg56479.html 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. 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. Arnaud Pouliquen (2): ASoC: core: allow DAI PCM controls bound to PCM device ASoC: sti: bind pcm controls to pcm device. include/sound/soc-dai.h | 8 ++++++++ sound/soc/soc-core.c | 35 +++++++++++++++++++++++++++++++++++ sound/soc/sti/sti_uniperif.c | 33 ++++----------------------------- 3 files changed, 47 insertions(+), 29 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/2] Generic HDMI codec: Add channel mapping control
by Arnaud Pouliquen 20 Jan '17

20 Jan '17

Aim of this patch is to add 'Playback Channel Map' control to export audio capabilities in term of HDMI sink speakers allocation. This patch follow discussion initiates here: [RFC] ASOC: HDMI audio info frame speaker allocation http://www.spinics.net/lists/alsa-devel/msg57363.html The code is fully inspired from HDA driver. On hw_param, HDMI sink speaker capabilities are exported via TLV ops and a CEA allocation is choson, based on ELD information and the number of channels requested by user. Mains differences with HDA implementation are: - Control is read only - Channel swap is not supported. Consequence is that unused channel in the mid of CEA audio infoframe channel mapping are considered as active. example for channel allocation 0x02: FL, FR, 0, FC) This configuration is only available for a 4 channels stream. - Channel allocation table has been reordered and HDMI 2.0 is not supported. Arnaud Pouliquen (2): DRM: add help to get ELD speaker allocation ASoC: hdmi-codec: add channel mapping control include/drm/drm_edid.h | 13 ++ sound/soc/codecs/hdmi-codec.c | 346 +++++++++++++++++++++++++++++++++++++++++- 2 files changed, 358 insertions(+), 1 deletion(-) -- 1.9.1

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[alsa-devel] [PATCH v2 0/4] Generic HDMI codec: Add channel mapping control
by Arnaud Pouliquen 20 Jan '17

20 Jan '17

Aim of this patch is to add 'Playback Channel Map' control to export audio capabilities in term of HDMI sink speakers allocation. V2: In this version I use chmap helper functions from pcm_lib.c. It is quite tricky to use it from ASoC due to the relation chip of the controls with the pcm runtime. After several tries, my conclusion is that it must be handled in ASoC DAI driver. Main reason is that the DAI driver can not provide snd_pcm_chmap struct through kcontrol structure. But this induces that soc-core provides pcm runtime structure to DAI driver during probe. Base on this conclusion. I reworked my patches by adding 2 new patches in patch-set 1) ALSA pcm: allow non constant snd_pcm_chmap_elem This patch allows to handle non constant channel mapping. As ELD information can change during runtime it is mandatory to properly handle the feature. In term of compatibility with legacy it should be straightforward, as update just consists in suppressing the 'const' constraint. 2) ASoC: core: add optional pcm_new callback for DAI driver This is the only way i found to be able to use snd_pcm_add_chmap_ctls and associated controls helper functions. From my point of view, this is the more simple way to add relationship between DAI and associated pcm devices. Notice that this patch, if accepted, makes the following one obsolete, as it also answer to the associated topic: [PATCH v5 0/2] ALSA controls management using index/device/sub-devices fields (http://www.spinics.net/lists/alsa-devel/msg57639.html). If you estimate that this it not reasonable i will come back to my first version. V1: This patch follow discussion initiates here: [RFC] ASOC: HDMI audio info frame speaker allocation http://www.spinics.net/lists/alsa-devel/msg57363.html The code is fully inspired from HDA driver. On hw_param, HDMI sink speaker capabilities are exported via TLV ops and a CEA allocation is choson, based on ELD information and the number of channels requested by user. Mains differences with HDA implementation are: - Control is read only - Channel swap is not supported. Consequence is that unused channel in the mid of CEA audio infoframe channel mapping are considered as active. example for channel allocation 0x02: FL, FR, 0, FC) This configuration is only available for a 4 channels stream. - Channel allocation table has been reordered and HDMI 2.0 is not supported. Arnaud Pouliquen (4): DRM: add help to get ELD speaker allocation ALSA pcm: allow non constant snd_pcm_chmap_elem ASoC: core: add optional pcm_new callback for DAI driver ASoC: hdmi-codec: add channel mapping control include/drm/drm_edid.h | 13 ++ include/sound/pcm.h | 4 +- include/sound/soc-dai.h | 3 + sound/core/pcm_lib.c | 6 +- sound/soc/codecs/hdmi-codec.c | 310 +++++++++++++++++++++++++++++++++++++++++- sound/soc/soc-core.c | 28 ++++ 6 files changed, 358 insertions(+), 6 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH v3 0/4] Add support for es8388 on the rock2
by Romain Perier 20 Jan '17

20 Jan '17

This set of patches adds a machine driver for rockchip boards that use ES8388 codecs. It also adds slave mode to the es8328 driver that currently only supported the master mode. Then, it adds the required DT definitions to link rockchip-i2s to the es8388 analog output. This work is based on the initial work that was done by Sjoerd Simons <sjoerd.simons(a)collabora.co.uk> with some improvements, changes and more commits. Changes in v3: - Cosmetic changes in rockchip_es8388.c - Added missing email to MODULE_AUTHOR in rockchip_es8388.c Changes in v2: - Fixed wrong use of the data structure i2c_device_id - Fixed wrong dependencies for SND_SOC_ROCKCHIP_ES8388 Romain Perier (4): ASoC: es8328-i2c: Add compatible for ES8388 ASoC: es8328: Add support for slave mode ASoC: rockchip: Add machine driver for ES8388 codecs arm: dts: Add support for ES8388 to the Radxa Rock 2 Documentation/devicetree/bindings/sound/es8328.txt | 2 +- .../devicetree/bindings/sound/rockchip-es8388.txt | 28 +++ arch/arm/boot/dts/rk3288-rock2-square.dts | 37 +++ sound/soc/codecs/es8328-i2c.c | 2 + sound/soc/codecs/es8328.c | 28 ++- sound/soc/rockchip/Kconfig | 9 + sound/soc/rockchip/Makefile | 2 + sound/soc/rockchip/rockchip_es8388.c | 269 +++++++++++++++++++++ 8 files changed, 370 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/rockchip-es8388.txt create mode 100644 sound/soc/rockchip/rockchip_es8388.c -- 2.9.3

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[alsa-devel] [PATCH] ASoC: sunxi: Add bindings for sun8i to SPDIF
by codekipper＠gmail.com 20 Jan '17

20 Jan '17

From: Marcus Cooper <codekipper(a)gmail.com> The H3 SoC uses the same SPDIF block as found in earlier SoCs, but the transmit fifo is at a different address. Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> --- Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt b/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt index 0230c4d20506..fe0a65e6d629 100644 --- a/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt +++ b/Documentation/devicetree/bindings/sound/sunxi,sun4i-spdif.txt @@ -10,6 +10,7 @@ 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. -- 2.11.0

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[alsa-devel] [PATCH] ASoC: sun4i-spdif: Add support for the H3 SoC
by codekipper＠gmail.com 20 Jan '17

20 Jan '17

From: Marcus Cooper <codekipper(a)gmail.com> The H3 SoC uses the same SPDIF block as found in earlier SoCs, but its TXFIFO is mapped to another address. Signed-off-by: Marcus Cooper <codekipper(a)gmail.com> --- sound/soc/sunxi/sun4i-spdif.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sunxi/sun4i-spdif.c b/sound/soc/sunxi/sun4i-spdif.c index fec62ee1fc72..c03cd07a9b19 100644 --- a/sound/soc/sunxi/sun4i-spdif.c +++ b/sound/soc/sunxi/sun4i-spdif.c @@ -103,6 +103,8 @@ #define SUN4I_SPDIF_ISTA_RXOSTA BIT(1) #define SUN4I_SPDIF_ISTA_RXASTA BIT(0) +#define SUN8I_SPDIF_TXFIFO (0x20) + #define SUN4I_SPDIF_TXCNT (0x24) #define SUN4I_SPDIF_RXCNT (0x28) @@ -417,6 +419,11 @@ static const struct sun4i_spdif_quirks sun6i_a31_spdif_quirks = { .has_reset = true, }; +static const struct sun4i_spdif_quirks sun8i_h3_spdif_quirks = { + .reg_dac_txdata = SUN8I_SPDIF_TXFIFO, + .has_reset = true, +}; + static const struct of_device_id sun4i_spdif_of_match[] = { { .compatible = "allwinner,sun4i-a10-spdif", @@ -426,6 +433,10 @@ static const struct of_device_id sun4i_spdif_of_match[] = { .compatible = "allwinner,sun6i-a31-spdif", .data = &sun6i_a31_spdif_quirks, }, + { + .compatible = "allwinner,sun8i-h3-spdif", + .data = &sun8i_h3_spdif_quirks, + }, { /* sentinel */ } }; MODULE_DEVICE_TABLE(of, sun4i_spdif_of_match); -- 2.11.0

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[alsa-devel] [PATCH] ASoC: dapm: fix some pointer error handling
by Linus Walleij 20 Jan '17

20 Jan '17

commit 66feeec9322132689d42723df2537d60f96f8e44 "RFC: ASoC: dapm: handle probe deferrals" forgot a to update some two sites where the call was used. The static codechecks quickly found them. Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Dan Carpenter <dan.carpenter(a)oracle.com> Reported-by: Dan Carpenter <dan.carpenter(a)oracle.com> Fixes: 66feeec93221 ("RFC: ASoC: dapm: handle probe deferrals") Signed-off-by: Linus Walleij <linus.walleij(a)linaro.org> --- sound/soc/soc-dapm.c | 20 ++++++++++++++++++++ 1 file changed, 20 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b218cc7bd994..dcef67a9bd48 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3935,6 +3935,16 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); + if (IS_ERR(w)) { + int ret = PTR_ERR(w); + + /* Do not nag about probe deferrals */ + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, + "ASoC: Failed to create %s widget (%d)\n", + dai->driver->playback.stream_name, ret); + return ret; + } if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->playback.stream_name); @@ -3954,6 +3964,16 @@ int snd_soc_dapm_new_dai_widgets(struct snd_soc_dapm_context *dapm, template.name); w = snd_soc_dapm_new_control_unlocked(dapm, &template); + if (IS_ERR(w)) { + int ret = PTR_ERR(w); + + /* Do not nag about probe deferrals */ + if (ret != -EPROBE_DEFER) + dev_err(dapm->dev, + "ASoC: Failed to create %s widget (%d)\n", + dai->driver->playback.stream_name, ret); + return ret; + } if (!w) { dev_err(dapm->dev, "ASoC: Failed to create %s widget\n", dai->driver->capture.stream_name); -- 2.9.3

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