Alsa-devel May 2019

alsa-devel@alsa-project.org

136 participants
372 discussions

[alsa-devel] Focusrite Scarlett 6i6 gen1 - input handling fix
by Jens Verwiebe 19 Nov '19

19 Nov '19

The Scarlett 6i6 has no padding on rear inputs 3/4 but a gainstage. This patch introduces this functionality as to be seen in the mac or windows scarlett control. The correct address could already be found in the dump info, but was never used. Cheers ... Jens Signed-off-by: Jens Verwiebe <info(a)jensverwiebe.de> diff --git a/sound/usb/mixer_scarlett.c b/sound/usb/mixer_scarlett.c index 7438e7c..f3bdac3 100644 --- a/sound/usb/mixer_scarlett.c +++ b/sound/usb/mixer_scarlett.c @@ -152,6 +152,7 @@ enum { SCARLETT_OUTPUTS, SCARLETT_SWITCH_IMPEDANCE, SCARLETT_SWITCH_PAD, + SCARLETT_SWITCH_GAIN, }; enum { @@ -202,6 +203,15 @@ struct scarlett_device_info { } }; +static const struct scarlett_mixer_elem_enum_info opt_gain = { + .start = 0, + .len = 2, + .offsets = {}, + .names = (char const * const []){ + "Lo", "Hi" + } +}; + static const struct scarlett_mixer_elem_enum_info opt_impedance = { .start = 0, .len = 2, @@ -664,8 +674,8 @@ static int add_output_ctls(struct usb_mixer_interface *mixer, { .num = 1, .type = SCARLETT_SWITCH_PAD, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_IMPEDANCE, .name = NULL}, { .num = 2, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 3, .type = SCARLETT_SWITCH_PAD, .name = NULL}, - { .num = 4, .type = SCARLETT_SWITCH_PAD, .name = NULL}, + { .num = 3, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, + { .num = 4, .type = SCARLETT_SWITCH_GAIN, .name = NULL}, }, .matrix_mux_init = { @@ -895,6 +905,15 @@ static int scarlett_controls_create_generic(struct usb_mixer_interface *mixer, if (err < 0) return err; break; + case SCARLETT_SWITCH_GAIN: + sprintf(mx, "Input %d Gain Switch", ctl->num); + err = add_new_ctl(mixer, &usb_scarlett_ctl_enum, + scarlett_ctl_enum_resume, 0x01, + 0x08, ctl->num, USB_MIXER_S16, 1, mx, + &opt_gain, &elem); + if (err < 0) + return err; + break; } } -- Jens Verwiebe Allerskehre 44 - 22309 Hamburg Tel.: +49 40 68 78 50 mobile: +49 172 400 49 07 mailto: info(a)jensverwiebe.de web: http://www.jensverwiebe.de

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[alsa-devel] [PATCH] ASoC: rsnd: dma: fix SSI9 4/5/6/7 busif dma address
by twischer＠de.adit-jv.com 22 Oct '19

22 Oct '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif dma address is calculated by following calculation formulation: 0xec540000 + 0x1000 * id + busif / 4 * 0xA000 + busif % 4 * 0x400 But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif data register address (SSI9_4_BUSIF/SSI9_5_BUSIF/SSI9_6_BUSIF/SSI9_7_BUSIF) are out of this rule. This patch updates the calculation formulation to correct ssi9 4/5/6/7 busif data register address Fixes: commit 5e45a6fab3b9 ("ASoc: rsnd: dma: Calculate dma address with consider of BUSIF") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/soc/sh/rcar/dma.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/dma.c b/sound/soc/sh/rcar/dma.c index 0324a5c..28f65eb 100644 --- a/sound/soc/sh/rcar/dma.c +++ b/sound/soc/sh/rcar/dma.c @@ -508,10 +508,10 @@ static struct rsnd_mod_ops rsnd_dmapp_ops = { #define RDMA_SSI_I_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0x8) #define RDMA_SSI_O_N(addr, i) (addr ##_reg - 0x00300000 + (0x40 * i) + 0xc) -#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_N(addr, i, j) (addr ##_reg - 0x00441000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_N(addr, i, j) RDMA_SSIU_I_N(addr, i, j) -#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400)) +#define RDMA_SSIU_I_P(addr, i, j) (addr ##_reg - 0x00141000 + (0x1000 * (i)) + (((j) / 4) * 0xA000) + (((j) % 4) * 0x400) - (0x4000 * ((i) / 9) * ((j) / 4))) #define RDMA_SSIU_O_P(addr, i, j) RDMA_SSIU_I_P(addr, i, j) #define RDMA_SRC_I_N(addr, i) (addr ##_reg - 0x00500000 + (0x400 * i)) -- 2.7.4

4 7

[alsa-devel] [PATCH] ASoC: rsnd: fixup 6ch settings to 8ch
by Kuninori Morimoto 30 Aug '19

30 Aug '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd need to use 8ch clock settings for 6ch for TDM. Otherwise, it can't work correctly. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 18 +++++++++++++----- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 ++ 3 files changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4fe83e61..37cb615 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -300,6 +300,18 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, return chan; } +int rsnd_channel_normalization(int chan) +{ + if ((chan > 8) || (chan < 0)) + return 0; + + /* TDM Extend Mode needs 8ch */ + if (chan == 6) + chan = 8; + + return chan; +} + int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, struct snd_pcm_hw_params *params) { @@ -312,11 +324,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, if (rsnd_runtime_is_multi_ssi(io)) chan /= rsnd_rdai_ssi_lane_get(rdai); - /* TDM Extend Mode needs 8ch */ - if (chan == 6) - chan = 8; - - return chan; + return rsnd_channel_normalization(chan); } int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0e6ef4e1..7727add 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -446,6 +446,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); +int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ rsnd_runtime_channel_original_with_params(io, NULL) int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f5afab6..44bda21 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -303,6 +303,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_runtime_is_tdm_split(io)) chan = rsnd_io_converted_chan(io); + chan = rsnd_channel_normalization(chan); + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

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[alsa-devel] [PATCH 2/3] input: Add SW_UNSUPPORT_INSERT define
by bgoswami＠codeaurora.org 22 Aug '19

22 Aug '19

From: Banajit Goswami <bgoswami(a)codeaurora.org> Some devices may not support specific type of input devices. For example, when a headset or extension cable with GND/MIC swap is plugged into a headset jack that does not support the headset/cable, it needs to be reported with a corresponding input event. Also, increase the max values for INPUT_DEVICE_ID_SW_MAX and SW_MAX, to accommodate future extension of the number of event. Signed-off-by: Gopikrishnaiah Anandan <agopik(a)codeaurora.org> Signed-off-by: Banajit Goswami <bgoswami(a)codeaurora.org> --- include/linux/mod_devicetable.h | 2 +- include/uapi/linux/input-event-codes.h | 3 ++- 2 files changed, 3 insertions(+), 2 deletions(-) diff --git a/include/linux/mod_devicetable.h b/include/linux/mod_devicetable.h index 448621c..7586099 100644 --- a/include/linux/mod_devicetable.h +++ b/include/linux/mod_devicetable.h @@ -299,7 +299,7 @@ struct pcmcia_device_id { #define INPUT_DEVICE_ID_LED_MAX 0x0f #define INPUT_DEVICE_ID_SND_MAX 0x07 #define INPUT_DEVICE_ID_FF_MAX 0x7f -#define INPUT_DEVICE_ID_SW_MAX 0x0f +#define INPUT_DEVICE_ID_SW_MAX 0x1f #define INPUT_DEVICE_ID_PROP_MAX 0x1f #define INPUT_DEVICE_ID_MATCH_BUS 1 diff --git a/include/uapi/linux/input-event-codes.h b/include/uapi/linux/input-event-codes.h index 85387c7..960fa86 100644 --- a/include/uapi/linux/input-event-codes.h +++ b/include/uapi/linux/input-event-codes.h @@ -808,7 +808,8 @@ #define SW_LINEIN_INSERT 0x0d /* set = inserted */ #define SW_MUTE_DEVICE 0x0e /* set = device disabled */ #define SW_PEN_INSERTED 0x0f /* set = pen inserted */ -#define SW_MAX 0x0f +#define SW_UNSUPPORT_INSERT 0x10 /* set = unsupported device inserted */ +#define SW_MAX 0x1f #define SW_CNT (SW_MAX+1) /* -- The Qualcomm Innovation Center, Inc. is a member of the Code Aurora Forum, a Linux Foundation Collaborative Project

2 1

[alsa-devel] [PATCH v4 0/7] Allwinner H6 SPDIF support
by Clément Péron 12 Aug '19

12 Aug '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 v3: - rename reg_fctl_ftx to val_fctl_ftx - rebase this series on sound-next - fix dt-bindings due to change in sound-next - change node name sound_spdif to sound-spdif 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 .../sound/allwinner,sun4i-a10-spdif.yaml | 1 + .../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, 87 insertions(+), 6 deletions(-) -- 2.20.1

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[alsa-devel] [PATCH] ALSA: usb-audio: fix PCM device order
by Alexander Tsoy 27 Jul '19

27 Jul '19

Some cards have alternate setting with non-PCM format as the first altsetting in the interface descriptors. This confuses userspace, since alsa-lib uses device 0 by default. So lets parse interfaces in two steps: 1. Parse altsettings with PCM formats. 2. Parse altsettings with non-PCM formats. This fixes at least following cards: - Audinst HUD-mx2 - Audinst HUD-mini Signed-off-by: Alexander Tsoy <alexander(a)tsoy.me> --- sound/usb/stream.c | 27 ++++++++++++++++++++++++++- 1 file changed, 26 insertions(+), 1 deletion(-) diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 8e9548bc1f1a..b2ebeae91396 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -477,7 +477,8 @@ static struct uac2_output_terminal_descriptor * return NULL; } -int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +static int __snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no, + bool *has_non_pcm, bool non_pcm) { struct usb_device *dev; struct usb_interface *iface; @@ -634,6 +635,10 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) iface_no, altno); continue; } + if (fmt->bFormatType != UAC_FORMAT_TYPE_I) + *has_non_pcm = true; + if ((fmt->bFormatType == UAC_FORMAT_TYPE_I) == non_pcm) + continue; if (((protocol == UAC_VERSION_1) && (fmt->bLength < 8)) || ((protocol == UAC_VERSION_2) && (fmt->bLength < 6))) { dev_err(&dev->dev, @@ -740,3 +745,23 @@ int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) return 0; } +int snd_usb_parse_audio_interface(struct snd_usb_audio *chip, int iface_no) +{ + int err; + bool has_non_pcm = false; + + /* parse PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, false); + if (err < 0) + return err; + + if (has_non_pcm) { + /* parse non-PCM formats */ + err = __snd_usb_parse_audio_interface(chip, iface_no, &has_non_pcm, true); + if (err < 0) + return err; + } + + return 0; +} + -- 2.13.0

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[alsa-devel] [PATCH v2 0/6]
by vanitha.channaiah＠in.bosch.com 16 Jul '19

16 Jul '19

- Updated the patches by incorporating review comments from Takashi Iwai-san [PATCH 1/5] pcm: direct: Add generic hw_ptr_alignment function for dmix, dshare and dsnoop [Takashi Iwai:] > The patch description needs rephrasing. What actually this does is to move the code from pcm_dmix.c to pcm_direct.c > and its header so that the helper code can be re-used by other direct-PCM plugins. - Commit message is rephrased as suggested. - New commit: [PATCH v2 1/6] pcm: direct: Add generic hw_ptr_alignment function for [PATCH 2/5] pcm: dshare: Added "hw_ptr_alignment" option in configuration for alignment of slave pointers [Takashi Iwai:] > Again, this patch description is too ambiguous. > And, if it enables the hw_ptr_alignment option, update the documentation as well. - Commit message is explained in detail for the changes done. - Documentation updated. - New commit: [PATCH v2 2/6] pcm: dshare: Added "hw_ptr_alignment" option in [PATCH 3/5] pcm: dsnoop: Added "hw_ptr_alignment" option in configuration for slave pointer alignment [Takashi Iwai:] > Ditto as patch 2, the description is too ambiguous, and the update of documentation is missing. > It's not good to change the helper function semantics out of sudden, even without any description. - Commit message is explained in detail for the changes done. - Documentation updated. - Divided the patch with commit ("pcm: dsnoop: Add hw_ptr_alignment option in configuration") into additional patch commit ("pcm: direct: Round up of slave_app_ptr pointer if buffer") - Usecase scenario is described for the changes done in helper function. - New commit: [PATCH v2 3/6] pcm: dsnoop: Added "hw_ptr_alignment" option in [PATCH v2 4/6] pcm: direct: Round up of slave_app_ptr pointer if buffer [PATCH 4/5] pcm: restructuring sw params function [Takashi Iwai:] > I see no reason to do that. Please describe. - Commit message is explained in detail why reformating was done. - New commit: [PATCH v2 5/6] pcm: restructuring sw params function [PATCH 5/5] pcm: Update pcm->avail_min with needed_slave_avail_min, after reading unaligned frames [Takashi Iwai:] > This kind of changes in the core code should be avoided as much as possible, especially if it's only relevant with the specific plugins. > Sorry, this isn't convincing enough. If this is a MUST, please clarify better. - Commit message is explained in detail with the generic usecase and specific use case we came across. - New commit: [PATCH v2 6/6] pcm: Update pcm->avail_min with needed_slave_avail_min,

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[alsa-devel] [PATCH] ALSA: pcm: Enable MMAP status and control for ARMv7 and ARMv8
by twischer＠de.adit-jv.com 10 Jul '19

10 Jul '19

From: Timo Wischer <twischer(a)de.adit-jv.com> Since ARMv7 hardware cache coherence is supported. "The SCU maintains coherency between the individual data caches in the Cortex-A5 MPCore processor using a variation of the MOESI protocol" [1]. Therefore this patch enables the MMAP access to the status and control structures. This avoids HWSYYNC ioctl calls and therefore lowers the CPU usage. [1] http://infocenter.arm.com/help/index.jsp?topic=/com.arm.doc.ddi0434c/ BABJECBF.html Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- sound/core/pcm_native.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 1d84529..b8019ef 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3225,7 +3225,8 @@ static __poll_t snd_pcm_poll(struct file *file, poll_table *wait) * Only on coherent architectures, we can mmap the status and the control records * for effcient data transfer. On others, we have to use HWSYNC ioctl... */ -#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA) +#if defined(CONFIG_X86) || defined(CONFIG_PPC) || defined(CONFIG_ALPHA) || \ + (defined(CONFIG_ARM) && defined(CONFIG_CPU_V7)) || defined(CONFIG_ARM64) /* * mmap status record */ -- 2.7.4

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[alsa-devel] [RFC PATCH 0/4] ASoC: hdac_hdmi: refine the hdac_hdmi jacks
by libin.yang＠intel.com 27 Jun '19

27 Jun '19

From: Libin Yang <libin.yang(a)intel.com> This patchset refines the hdac_hdmi jacks: 1. kctl setting should not impact on hdmi jack status 2. bind jack to pin-port instead of pcm 3. move creating jack into hdac_hdmi from machine drivers Libin Yang (4): ASoC: hdac_hdmi: amixer kctl setting should not impact on jack status ASoC: hdac_hdmi: jack_event represents the status ASoC: hdac_hdmi: refine hdmi jacks ASoC: hdac_hdmi: fix some coding style issue sound/soc/codecs/hdac_hdmi.c | 176 +++++++++++---------- 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, 92 insertions(+), 215 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH 0/9] ASoC: cleanup for ASoC
by Kuninori Morimoto 25 Jun '19

25 Jun '19

Hi Mark I'm tring to "no categorize DAI" now, but it will takes more time. Then, I noticed some current code is not good balance for it. It is small issue. But it is very helpful if it had been already solved before posting such patch-set. Kuninori Morimoto (9): ASoC: soc-core: use i on snd_soc_resume() ASoC: soc-core: don't use codec_dais on soc_bind_dai_link() ASoC: soc-core: tidyup soc_bind_dai_link() comment balance ASoC: soc-core: add soc_component_to_node() ASoC: soc-core: use soc_find_component() at snd_soc_get_dai_id() ASoC: soc.h: fe_compr can be bit field ASoC: soc-pcm: do cpu_dai related operation at same place ASoC: soc-pcm: add missing cpu_dai->rate = 0 ASoC: soc-pcm: add soc_pcm_components_open() include/sound/soc.h | 2 +- sound/soc/soc-core.c | 75 ++++++++++++++++++++---------------------------- sound/soc/soc-pcm.c | 80 ++++++++++++++++++++++++++++++++-------------------- 3 files changed, 81 insertions(+), 76 deletions(-) -- 2.7.4 Thank you for your help !! Best regards --- Kuninori Morimoto

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Alsa-devel May 2019