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

[alsa-devel] Applied "ASoC: rockchip: cdn-dp sound output use spdif" to the asoc tree
by Mark Brown 23 May '18

23 May '18

The patch ASoC: rockchip: cdn-dp sound output use spdif 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 b18c6c3c7768b7b81c2250bbccd0c3c0aed6b71f Mon Sep 17 00:00:00 2001 From: Lin Huang <hl(a)rock-chips.com> Date: Tue, 22 May 2018 16:53:42 +0800 Subject: [PATCH] ASoC: rockchip: cdn-dp sound output use spdif some monitors care about the parity bit in the sub-frame of I2S, but the cdn-dp always set this bit to "1", so these monitors do not have sound output if use i2s, use spdif can fix this issue. Signed-off-by: Chris Zhong <zyw(a)rock-chips.com> Signed-off-by: Lin Huang <hl(a)rock-chips.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/rockchip/rk3399_gru_sound.c | 46 +-------------------------- 1 file changed, 1 insertion(+), 45 deletions(-) diff --git a/sound/soc/rockchip/rk3399_gru_sound.c b/sound/soc/rockchip/rk3399_gru_sound.c index 9a10181a0811..f184168f9a41 100644 --- a/sound/soc/rockchip/rk3399_gru_sound.c +++ b/sound/soc/rockchip/rk3399_gru_sound.c @@ -220,45 +220,6 @@ static int rockchip_sound_da7219_init(struct snd_soc_pcm_runtime *rtd) return 0; } -static int rockchip_sound_cdndp_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int mclk, ret; - - /* in bypass mode, the mclk has to be one of the frequencies below */ - switch (params_rate(params)) { - case 8000: - case 16000: - case 24000: - case 32000: - case 48000: - case 64000: - case 96000: - mclk = 12288000; - break; - case 11025: - case 22050: - case 44100: - case 88200: - mclk = 11289600; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_sysclk(cpu_dai, 0, mclk, - SND_SOC_CLOCK_OUT); - if (ret < 0) { - dev_err(codec_dai->dev, "Can't set cpu clock out %d\n", ret); - return ret; - } - - return 0; -} - static int rockchip_sound_dmic_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -293,10 +254,6 @@ static const struct snd_soc_ops rockchip_sound_da7219_ops = { .hw_params = rockchip_sound_da7219_hw_params, }; -static const struct snd_soc_ops rockchip_sound_cdndp_ops = { - .hw_params = rockchip_sound_cdndp_hw_params, -}; - static const struct snd_soc_ops rockchip_sound_dmic_ops = { .hw_params = rockchip_sound_dmic_hw_params, }; @@ -323,8 +280,7 @@ static const struct snd_soc_dai_link rockchip_dais[] = { [DAILINK_CDNDP] = { .name = "DP", .stream_name = "DP PCM", - .codec_dai_name = "i2s-hifi", - .ops = &rockchip_sound_cdndp_ops, + .codec_dai_name = "spdif-hifi", .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, }, -- 2.17.0

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[alsa-devel] [PATCH] ASoC: Intel: cht_bsw_nau8824: Fix jack_type to include SND_JACK_MICROPHONE
by Hans de Goede 23 May '18

23 May '18

The nau8824 codec can detect whether a headset or plain headphones is inserted (as well as button presses on the headset) as such the jack_type passed to snd_soc_card_jack_new() should include SND_JACK_MICROPHONE. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- sound/soc/intel/boards/cht_bsw_nau8824.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/cht_bsw_nau8824.c b/sound/soc/intel/boards/cht_bsw_nau8824.c index 072e94b69e57..30c46977d53c 100644 --- a/sound/soc/intel/boards/cht_bsw_nau8824.c +++ b/sound/soc/intel/boards/cht_bsw_nau8824.c @@ -120,7 +120,7 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) * KEY_VOLUMEUP * KEY_VOLUMEDOWN */ - jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | + jack_type = SND_JACK_HEADSET | SND_JACK_BTN_0 | SND_JACK_BTN_1 | SND_JACK_BTN_2 | SND_JACK_BTN_3; ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); -- 2.17.0

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[alsa-devel] [PATCH v3 0/6] ALSA: xen-front: Add Xen para-virtualized frontend driver
by Oleksandr Andrushchenko 23 May '18

23 May '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Please note: this patch series depends on [3]. This patch series adds support for Xen [1] para-virtualized sound frontend driver. It implements the protocol from include/xen/interface/io/sndif.h with the following limitations: - mute/unmute is not supported - get/set volume is not supported Volume control is not supported for the reason that most of the use-cases (at the moment) are based on scenarious where unprivileged OS (e.g. Android, AGL etc) use software mixers. Both capture and playback are supported. Corresponding backend, implemented as a user-space application, can be found at [2]. Thank you, Oleksandr Changes since v2: ***************** 1. Kconfig: select SND_PCM instead of depends 2. Changed spinlock to mutex for protecting the rings since all the interrupts are threaded. Make those per event channel, not a single global one. This also addresses Juergen's comment on calling xen_snd_front_alsa_handle_cur_pos. 3. Fixed page leak on error while allocating an event channel 4. Fixed comments to start from a capital letter 5. Performed XEN_PAGE_SIZE != PAGE_SIZE check after xen_domain and xen_has_pv_devices checks 6. Changed driver specific structures names not to mislead with ALSA structs 7. Added comments on ring counters and overflows 8. Added MAINTAINERS entry Changes since v1: ***************** 1. Moved driver from sound/drivers to sound/xen 2. Coding style changes to better meet Linux Kernel 3. Added explicit back and front synchronization In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events 4. Added explicit back and front parameter negotiation In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameters given: request passes desired parameter's intervals/masks and the response to this request returns allowed min/max intervals/masks to be used. [1] https://xenproject.org/ [2] https://github.com/xen-troops/snd_be [3] https://lkml.org/lkml/2018/4/12/522 Oleksandr Andrushchenko (6): ALSA: xen-front: Introduce Xen para-virtualized sound frontend driver ALSA: xen-front: Read sound driver configuration from Xen store ALSA: xen-front: Implement Xen event channel handling ALSA: xen-front: Implement handling of shared buffers ALSA: xen-front: Implement ALSA virtual sound driver MAINTAINERS: Add ALSA: xen-front: maintainer entry MAINTAINERS | 7 + sound/Kconfig | 2 + sound/Makefile | 2 +- sound/xen/Kconfig | 10 + sound/xen/Makefile | 9 + sound/xen/xen_snd_front.c | 397 +++++++++++++++ sound/xen/xen_snd_front.h | 54 ++ sound/xen/xen_snd_front_alsa.c | 821 ++++++++++++++++++++++++++++++ sound/xen/xen_snd_front_alsa.h | 23 + sound/xen/xen_snd_front_cfg.c | 517 +++++++++++++++++++ sound/xen/xen_snd_front_cfg.h | 46 ++ sound/xen/xen_snd_front_evtchnl.c | 496 ++++++++++++++++++ sound/xen/xen_snd_front_evtchnl.h | 95 ++++ sound/xen/xen_snd_front_shbuf.c | 193 +++++++ sound/xen/xen_snd_front_shbuf.h | 36 ++ 15 files changed, 2707 insertions(+), 1 deletion(-) create mode 100644 sound/xen/Kconfig create mode 100644 sound/xen/Makefile create mode 100644 sound/xen/xen_snd_front.c create mode 100644 sound/xen/xen_snd_front.h create mode 100644 sound/xen/xen_snd_front_alsa.c create mode 100644 sound/xen/xen_snd_front_alsa.h create mode 100644 sound/xen/xen_snd_front_cfg.c create mode 100644 sound/xen/xen_snd_front_cfg.h create mode 100644 sound/xen/xen_snd_front_evtchnl.c create mode 100644 sound/xen/xen_snd_front_evtchnl.h create mode 100644 sound/xen/xen_snd_front_shbuf.c create mode 100644 sound/xen/xen_snd_front_shbuf.h -- 2.17.0

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[alsa-devel] [BUG] sound: soc: sunxi: sun4i-codec.c left control in right mixer
by Markus Mitsch 22 May '18

22 May '18

Hi, in linux/sound/sunxi/sun4i-codec.c there is a control for the left channel in the section of routes for the right channel at line number 770. I dont know if this is wanted. And, i dont know if it is related, but to get sound out of my left speaker i have to MUTE the "Power Amplifier Mixer" channel with alsamixer. Regards

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[alsa-devel] [PATCH 31/33] ASoC: max98088: use match_string() helper
by Yisheng Xie 22 May '18

22 May '18

match_string() returns the index of an array for a matching string, which can be used intead of open coded variant. 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 Signed-off-by: Yisheng Xie <xieyisheng1(a)huawei.com> --- sound/soc/codecs/max98088.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index 865f64c..7383f90 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1382,14 +1382,11 @@ static int max98088_set_bias_level(struct snd_soc_component *component, static int max98088_get_channel(struct snd_soc_component *component, const char *name) { - int i; - - for (i = 0; i < ARRAY_SIZE(eq_mode_name); i++) - if (strcmp(name, eq_mode_name[i]) == 0) - return i; + int ret = match_string(eq_mode_name, ARRAY_SIZE(eq_mode_name), name); /* Shouldn't happen */ - dev_err(component->dev, "Bad EQ channel name '%s'\n", name); + if (ret < 0) + dev_err(component->dev, "Bad EQ channel name '%s'\n", name); return -EINVAL; } -- 1.7.12.4

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[alsa-devel] alsa plugins + slibtool
by orbea＠fredslev.dk 22 May '18

22 May '18

Hi, I am experiencing build issues with alsa-plugins when using slibtool instead of libtool. This build issue is because of a bug in alsa-plugins which libtool hides so I have attached a log showing the build error and a small patch to solve it. This patch will require regenerating the autotools files with a command like 'autoreconf -fi' to take effect. If this could be applied in the next alsa-plugins release I would be very grateful. Also note that I have tried to send this patch to the mailing list before and am unsure if it ever made it past moderation? For reference here is a link to the slibtool source. https://git.midipix.org/cgit.cgi/slibtool/ And to their README that explains the project. https://git.midipix.org/cgit.cgi/slibtool/plain/README Thanks, Hunter

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[alsa-devel] Xonar U7
by Steve Banks 22 May '18

22 May '18

Hi, In the USB-Audio.conf file, the Xonar U7 requires the following definition inthe following section USB-Audio.pcm.iec958_device { "Xonar U7" 1 for the spdif to function correctly Thanks Steve Banks

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[alsa-devel] [PATCH 1/2] ALSA: hda/conexant - Add fixup for HP Z2 G4 workstation
by Takashi Iwai 22 May '18

22 May '18

HP Z2 G4 requires the same workaround as other HP machines that have no mic-pin detection. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 093d2a9ece85..dad7e9d5e5a6 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -964,6 +964,7 @@ static const struct snd_pci_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x103c, 0x8455, "HP Z2 G4", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x138d, "Asus", CXT_FIXUP_HEADPHONE_MIC_PIN), SND_PCI_QUIRK(0x152d, 0x0833, "OLPC XO-1.5", CXT_FIXUP_OLPC_XO), SND_PCI_QUIRK(0x17aa, 0x20f2, "Lenovo T400", CXT_PINCFG_LENOVO_TP410), -- 2.16.3

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[alsa-devel] [PATCH 1/2] ASoC: pxa-ssp: allow more flexible setup order
by Daniel Mack 22 May '18

22 May '18

The pxa-ssp driver currently assumes that .set_fmt() is called before .set_clkdiv(), .set_pll() etc. Commit a8bd0ee558714 ("ASoC: raumfeld: Use static DAI format setup") broke support for Raumfeld hardware (and possible other PXA based ones) because it effectively changed the order of these calls. Also, as the call to .set_fmt() is now done at probe time, the port clock is not yet enabled. To fix this, strip all hardware register access code from the .set_fmt() callback and memorize the desired value, so we can use it from the .hw_params() callback. Also make the .set_fmt() callback less destructive by reading all registers that it writes to in the beginning and only masking out the bits that it possibly fiddles with. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/pxa/pxa-ssp.c | 82 +++++++++++++++++++++++++++++++++++++------------ 1 file changed, 62 insertions(+), 20 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index a0189b88f1d2..fbed87d824b8 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -43,7 +43,8 @@ struct ssp_priv { struct ssp_device *ssp; unsigned int sysclk; - int dai_fmt; + unsigned int dai_fmt; + unsigned int configured_dai_fmt; #ifdef CONFIG_PM uint32_t cr0; uint32_t cr1; @@ -432,36 +433,72 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, return 0; } +static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, + unsigned int fmt) +{ + struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + case SND_SOC_DAIFMT_CBM_CFS: + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + case SND_SOC_DAIFMT_NB_IF: + case SND_SOC_DAIFMT_IB_IF: + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_DSP_A: + case SND_SOC_DAIFMT_DSP_B: + break; + + default: + return -EINVAL; + } + + /* Settings will be applied in hw_params() */ + priv->dai_fmt = fmt; + + return 0; +} + /* * Set up the SSP DAI format. * The SSP Port must be inactive before calling this function as the * physical interface format is changed. */ -static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) +static int pxa_ssp_configure_dai_fmt(struct ssp_priv *priv) { - struct ssp_priv *priv = snd_soc_dai_get_drvdata(cpu_dai); struct ssp_device *ssp = priv->ssp; u32 sscr0, sscr1, sspsp, scfr; /* check if we need to change anything at all */ - if (priv->dai_fmt == fmt) + if (priv->configured_dai_fmt == priv->dai_fmt) return 0; - /* we can only change the settings if the port is not in use */ - if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) { - dev_err(&ssp->pdev->dev, - "can't change hardware dai format: stream is in use"); - return -EINVAL; - } - /* reset port settings */ sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & - ~(SSCR0_ECS | SSCR0_NCS | SSCR0_MOD | SSCR0_ACS); - sscr1 = SSCR1_RxTresh(8) | SSCR1_TxTresh(7); - sspsp = 0; + ~(SSCR0_PSP | SSCR0_MOD); + sscr1 = pxa_ssp_read_reg(ssp, SSCR1) & + ~(SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR | + SSCR1_RWOT | SSCR1_TRAIL | SSCR1_TFT | SSCR1_RFT); + sspsp = pxa_ssp_read_reg(ssp, SSPSP) & + ~(SSPSP_SFRMP | SSPSP_SCMODE(3)); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + sscr1 |= SSCR1_RxTresh(8) | SSCR1_TxTresh(7); + + switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: sscr1 |= SSCR1_SCLKDIR | SSCR1_SFRMDIR | SSCR1_SCFR; break; @@ -474,7 +511,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (priv->dai_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: sspsp |= SSPSP_SFRMP; break; @@ -490,7 +527,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; } - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; @@ -512,7 +549,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, pxa_ssp_write_reg(ssp, SSCR1, sscr1); pxa_ssp_write_reg(ssp, SSPSP, sspsp); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + switch (priv->dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: case SND_SOC_DAIFMT_CBM_CFS: scfr = pxa_ssp_read_reg(ssp, SSCR1) | SSCR1_SCFR; @@ -529,7 +566,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, * we have to defer some things until hw_params() where we * know parameters like the sample size. */ - priv->dai_fmt = fmt; + priv->configured_dai_fmt = priv->dai_fmt; return 0; } @@ -550,6 +587,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = pxa_ssp_read_reg(ssp, SSTSA) & 0xf; struct snd_dmaengine_dai_dma_data *dma_data; + int ret; dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream); @@ -565,6 +603,10 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, if (pxa_ssp_read_reg(ssp, SSCR0) & SSCR0_SSE) return 0; + ret = pxa_ssp_configure_dai_fmt(priv); + if (ret < 0) + return ret; + /* clear selected SSP bits */ sscr0 = pxa_ssp_read_reg(ssp, SSCR0) & ~(SSCR0_DSS | SSCR0_EDSS); -- 2.14.3

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[alsa-devel] [PATCH 1/2] ASoC: wm8782: add device-tree matching table
by Daniel Mack 22 May '18

22 May '18

This is needed when the codec is instanciated from from a device tree. Signed-off-by: Daniel Mack <daniel(a)zonque.org> --- sound/soc/codecs/wm8782.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/codecs/wm8782.c b/sound/soc/codecs/wm8782.c index 7949703f3d04..317db9a149a7 100644 --- a/sound/soc/codecs/wm8782.c +++ b/sound/soc/codecs/wm8782.c @@ -67,9 +67,18 @@ static int wm8782_probe(struct platform_device *pdev) &soc_component_dev_wm8782, &wm8782_dai, 1); } +#ifdef CONFIG_OF +static const struct of_device_id wm8782_of_match[] = { + { .compatible = "wlf,wm8782", }, + { } +}; +MODULE_DEVICE_TABLE(of, wm8782_of_match); +#endif + static struct platform_driver wm8782_codec_driver = { .driver = { .name = "wm8782", + .of_match_table = of_match_ptr(wm8782_of_match), }, .probe = wm8782_probe, }; -- 2.14.3

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