September 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 18 Sep '15

18 Sep '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..5c737f1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

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[alsa-devel] Intel SST on a Bay Trail tablet
by Antonio Ospite 18 Sep '15

18 Sep '15

Hi, I am trying to get the Intel SST driver working on a Teclast X98 Air 3G [1], it's a Bay Trail tablet. The tests below have been made with 4.0.0-rc1 and the firmware files from https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/tr… The DSDT on this device is quite broken, and I am patching it to get stuff working. The audio device works just fine under Android, I don't know if that's because the Android kernel is ignoring the DSDT and is hard-coding some platform data. Anyhow, with the untouched DSDT[2] I was getting this message in dmesg[3]: [ 11.344095] platform byt-rt5640: Driver byt-rt5640 requests probe deferral [ 11.404831] resource sanity check: requesting [mem 0x91000000-0x911fffff], which spans more than ADMA0F28:00 [mem 0x91000000-0x91000fff] [ 11.404833] ------------[ cut here ]------------ [ 11.404849] WARNING: CPU: 0 PID: 213 at arch/x86/mm/ioremap.c:197 __ioremap_caller+0x22c/0x350() ... I guessed this was because the DSDT defined an ADMA0F28 device with a memory resource overlapping with the LPMA device. So I removed the overlapping resources (ADMA0F28 and SSPX0000) in the DSDT and then the driver attempted to boot the audio device but failed with this message in dmesg[4]: [ 11.602291] baytrail-pcm-audio baytrail-pcm-audio: ipc: error DSP boot timeout ... Then I noticed that the Interrupt resources of the LPMA device were out of order in the DSDT, so I put the Interrupt 0x18 as the first one[5] and now the driver can boot the device and upload the firmware to it[6]: [ 11.433460] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 11.433468] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 11.433473] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 11.448257] byt-rt5640 byt-rt5640: rt5640-aif1 <-> baytrail-pcm-audio mapping ok ... But still no sound. I can see the mixer controls in alsamixer and I tried to initialize them copying from the alsactl output from the Android kernel[7], but no, still no sound. I do not know if that matters, but the driver still emits this message: [ 11.398304] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 11.398314] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 ... I noticed that on the Android kernel[8] the DMA channels for the ADMA0F28 device are actually configured, the intel_mid_dma driver sets up 4 channels for it along with 2 x 6 channels for the INTL9C60 device. On the mainline kernel only the INTL9C60 device gets configured, by the dw_dmac driver and with 2 x 8 channels. I do not know much about these details tho. Also, the codec driver on the Android kernel (byt_rt5642) seems to use one GPIO as an interrupt[9]: 390: 0 1 0 0 VLV-GPIO-gpio byt-codec-int Which is pin 4 on GPO2, but the codec driver in the mainline kernel does not seem to be using this; however AFAICS in sound/soc/codecs/rt5640.c a gpio pin in only used for suspend/resume, so it is not strictly needed if it's already in the correct state, right? Last question, what is the difference between having the device detected by sound/soc/intel/sst-acpi.c (like in my case) opposed to sound/soc/intel/sst/sst_acpi.c? I can see the 80860F28 id in both the files but they seem to load different firmwares. Any idea of what is missing? It's the DSDT which still needs adjusting, or are some changes needed (also) in the kernel driver? Thanks, Antonio [1] http://ao2.it/110 [2] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_original.dsl [3] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_ori… [4] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [5] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_without_ADMA0… [6] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [7] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_asound.state [8] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_Android_Teclast_… [9] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_proc_interrupt… -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

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[alsa-devel] Applied "ASoC: fsl-asoc-card: add AC'97 support" to the asoc tree
by Mark Brown 17 Sep '15

17 Sep '15

The patch ASoC: fsl-asoc-card: add AC'97 support has been applied to the asoc tree at git://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 e06b508481e1916b25038289b945104009d774c9 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" <mail(a)maciej.szmigiero.name> Date: Mon, 31 Aug 2015 17:11:35 +0200 Subject: [PATCH] ASoC: fsl-asoc-card: add AC'97 support Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 145 ++++++++++++++++----- 2 files changed, 120 insertions(+), 35 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed..86aa498 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -404,10 +451,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; - struct platform_device *cpu_pdev; + struct platform_device *cpu_pdev, *codec_pdev; struct fsl_asoc_card_priv *priv; - struct i2c_client *codec_dev; - struct clk *codec_clk; + struct i2c_client *codec_i2c_dev; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,11 +479,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) { + codec_pdev = of_find_device_by_node(codec_np); + codec_i2c_dev = of_find_i2c_device_by_node(codec_np); + } else { + codec_pdev = NULL; + codec_i2c_dev = NULL; } asrc_np = of_parse_phandle(np, "audio-asrc", 0); @@ -446,10 +493,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_pdev) { + struct clk *codec_clk = clk_get(&codec_pdev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,11 +536,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_pdev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -507,7 +567,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_i2c_dev ? codec_i2c_dev->name : codec_pdev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -531,8 +593,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -543,6 +623,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -578,14 +660,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, -- 2.5.0

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[alsa-devel] [PATCH v2] audio: sai: Add Power Management support
by Alison Wang 17 Sep '15

17 Sep '15

This patch adds Power Management support for SAI. Activate regmap cache with REGCACHE_FLAT, and use regmap cache code to save and restore registers in suspend and resume. The Transmit Control Register (TCSR) and Receive Control Register(RCSR) should be volatile registers. Signed-off-by: Alison Wang <alison.wang(a)freescale.com> --- Changes since v1: - Use REGCACHE_FLAT instead of REGCACHE_RBTREE. - Use SIMPLE_DEV_PM_OPS to simplify the code. sound/soc/fsl/fsl_sai.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..030f71e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,9 +637,11 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: case FSL_SAI_TFR: - case FSL_SAI_RFR: case FSL_SAI_TDR: + case FSL_SAI_RCSR: + case FSL_SAI_RFR: case FSL_SAI_RDR: return true; default: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -796,6 +799,31 @@ static int fsl_sai_probe(struct platform_device *pdev) return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } +#ifdef CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + /* Restore all registers */ + regcache_cache_only(sai->regmap, false); + regcache_sync(sai->regmap); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static SIMPLE_DEV_PM_OPS(fsl_sai_pm, fsl_sai_suspend, fsl_sai_resume); + static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, @@ -807,6 +835,7 @@ static struct platform_driver fsl_sai_driver = { .driver = { .name = "fsl-sai", .of_match_table = fsl_sai_ids, + .pm = &fsl_sai_pm, }, }; module_platform_driver(fsl_sai_driver); -- 2.1.0.27.g96db324

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[alsa-devel] Multiple codecs on one sound card for multi-channel sound card
by Caleb Crome 17 Sep '15

17 Sep '15

Hi all, What's the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus? I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved... I currently have something like this in my am335x-boneblack.dts file: (i.e. attempting to set tdm slot width and num in the i2c codec) but it doesn't seem to be pickin up the fact that I want all the codecs linked together as one. (FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves). Thanks for any help! BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort! Sincerely, -Caleb &i2c1 { clock-frequency = <100000>; status = "okay"; pinctrl-names = "default"; pinctrl-0 = <&i2c1_pins_default>; status="okay"; tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x18>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <0>; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x19>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <2>; }; tlv320aic3x_c: tlv320aic3x@1a { compatible = "ti,tlv320aic3x"; reg = <0x1a>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <4>; }; tlv320aic3x_d: tlv320aic3x@1b { compatible = "ti,tlv320aic3x"; reg = <0x1b>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <6>; }; }; sound { compatible = "simple-audio-card"; simple-audio-card,name = "puck audio"; // simple-audio-card,widgets= // simple-audio-card,routing= simple-audio-card,mclk-fs = <256>; status="okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0 0>; }; // *** Here's where I'm really confused -- this part doesn't seem to be supported to // have all the codecs on a single DAI. // The examples given seem to be for different codec DAIs // sharing a single CPU DAI, but not used at the same time. codec { #sound-dai-cells = <0>; sound-dai = < &tlv320aic3x_a &tlv320aic3x_b &tlv320aic3x_c &tlv320aic3x_d >; }; }; };

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Re: [alsa-devel] [linux-sunxi] Re: [PATCH 2/2] ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs
by Mark Brown 17 Sep '15

17 Sep '15

On Thu, Sep 17, 2015 at 11:10:41PM +0800, Chen-Yu Tsai wrote: > On Thu, Sep 17, 2015 at 9:31 PM, Maxime Ripard > >> > + .formats = SNDRV_PCM_FMTBIT_S16_LE | > >> > + SNDRV_PCM_FMTBIT_S32_LE, > >> > + .sig_bits = 24, > >> So presumably also S24_LE (ie, 24 bits packed into a 32 bit word)? > > Hmm, probably yes, I'll test that. > IIRC when Emilio first wrote the driver, we tried 24 bit and no sound > came out. Turns out it's an alignment issue. The codec's FIFO register > is 32 bits wide, and takes the higher 24 bits as input when set to 24 > bit mode. The internal FIFO is only 24 bits wide. A20 user manual P174 > describes how the bits are copied. > So for 24 bit audio, you would actually send it 32 bit audio samples, > and let it truncate or drop the least significant 8 bits. This is why > we have SNDRV_PCM_FMTBIT_S32_LE with .sig_bits = 24. > I don't know if this is just a workaround, but a few other drivers do > this as well, for example twl3040 and omap-mcpdm. That's very common - it's essentially what S24_LE mode is.

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[alsa-devel] [PATCH v2] ASoC: wm5110: Add additional analogue input enable for early revs
by Charles Keepax 17 Sep '15

17 Sep '15

Earlier revisions of the wm5110/8280 silicon require a slightly more complex procedure to enable analogue inputs. This patch adds this into the driver. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Added a comment to explain why we need to hold the DAPM lock when setting the PGA volume for the inputs Thanks, Charles sound/soc/codecs/wm5110.c | 187 +++++++++++++++++++++++++++++++++++++++------ 1 files changed, 163 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9756578..c04c0bc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -38,6 +38,12 @@ struct wm5110_priv { struct arizona_priv core; struct arizona_fll fll[2]; + + unsigned int in_value; + int in_pre_pending; + int in_post_pending; + + unsigned int in_pga_cache[6]; }; static const struct wm_adsp_region wm5110_dsp1_regions[] = { @@ -428,6 +434,127 @@ err: return ret; } +static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct snd_soc_card *card = dapm->card; + int ret; + + /* + * PGA Volume is also used as part of the enable sequence, so + * usage of it should be avoided whilst that is running. + */ + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_get_volsw_range(kcontrol, ucontrol); + + mutex_unlock(&card->dapm_mutex); + + return ret; +} + +static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct snd_soc_card *card = dapm->card; + int ret; + + /* + * PGA Volume is also used as part of the enable sequence, so + * usage of it should be avoided whilst that is running. + */ + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_put_volsw_range(kcontrol, ucontrol); + + mutex_unlock(&card->dapm_mutex); + + return ret; +} + +static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + unsigned int reg, mask; + struct reg_sequence analog_seq[] = { + { 0x80, 0x3 }, + { 0x35d, 0 }, + { 0x80, 0x0 }, + }; + + reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4); + mask = ARIZONA_IN1L_PGA_VOL_MASK; + + switch (event) { + case SND_SOC_DAPM_WILL_PMU: + wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2); + wm5110->in_pre_pending++; + wm5110->in_post_pending++; + return 0; + case SND_SOC_DAPM_PRE_PMU: + wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg); + + snd_soc_update_bits(codec, reg, mask, + 0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT); + + wm5110->in_pre_pending--; + if (wm5110->in_pre_pending == 0) { + analog_seq[1].def = wm5110->in_value; + regmap_multi_reg_write_bypassed(arizona->regmap, + analog_seq, + ARRAY_SIZE(analog_seq)); + + msleep(55); + + wm5110->in_value = 0; + } + + break; + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, reg, mask, + wm5110->in_pga_cache[w->shift]); + + wm5110->in_post_pending--; + if (wm5110->in_post_pending == 0) + regmap_multi_reg_write_bypassed(arizona->regmap, + analog_seq, + ARRAY_SIZE(analog_seq)); + break; + default: + break; + } + + return 0; +} + +static int wm5110_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + + switch (arizona->rev) { + case 0 ... 4: + if (arizona_input_analog(codec, w->shift)) + wm5110_in_analog_ev(w, kcontrol, event); + + break; + default: + break; + } + + return arizona_in_ev(w, kcontrol, event); +} + static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); @@ -454,18 +581,24 @@ SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), -SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, - ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, - ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, - ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum), @@ -896,29 +1029,35 @@ SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | -- 1.7.2.5

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[alsa-devel] [PATCH v2] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..af4999c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3294,7 +3294,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, unsigned int *mask) { u32 val; - const u32 *of_slot_mask = of_get_property(np, prop_name, &val); + const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); int i; if (!of_slot_mask) -- 1.9.1

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[alsa-devel] [PATCH 0/2] change PTR_ERR argument
by Julia Lawall 17 Sep '15

17 Sep '15

Apply PTR_ERR to the value that was recently assigned. --- var/julia/linuxcopy/drivers/spi/spi-ep93xx.c | 3 ++- var/julia/linuxcopy/sound/soc/qcom/lpass-cpu.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-)

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[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data
by Jyri Sarha 17 Sep '15

17 Sep '15

The slot_width is for essentially same thing. Instead of storing bclk_lrclk_ratio, just store the slot_width. Comments has been updated accordingly and some variable names changed to more descriptive. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Added the missing __func__ to dev_warn() warn call in __davinci_mcasp_set_clkdiv(). sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 25 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de1e3a8..9a9d633 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -87,7 +87,6 @@ struct davinci_mcasp { u8 *serial_dir; u8 version; u8 bclk_div; - u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; int dma_request[2]; @@ -558,8 +557,21 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, mcasp->bclk_div = div; break; - case 2: /* BCLK/LRCLK ratio */ - mcasp->bclk_lrclk_ratio = div; + case 2: /* + * BCLK/LRCLK ratio descries how many bit-clock cycles + * fit into one frame. The clock ratio is given for a + * full period of data (for I2S format both left and + * right channels), so it has to be divided by number + * of tdm-slots (for I2S - divided by 2). + * Instead of storing this ratio, we calculate a new + * tdm_slot width by dividing the the ratio by the + * number of configured tdm slots. + */ + mcasp->slot_width = div / mcasp->tdm_slots; + if (div % mcasp->tdm_slots) + dev_warn(mcasp->dev, + "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots", + __func__, div, mcasp->tdm_slots); break; default: @@ -682,11 +694,13 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, } static int davinci_config_channel_size(struct davinci_mcasp *mcasp, - int word_length) + int sample_width) { u32 fmt; - u32 tx_rotate = (word_length / 4) & 0x7; - u32 mask = (1ULL << word_length) - 1; + u32 tx_rotate = (sample_width / 4) & 0x7; + u32 mask = (1ULL << sample_width) - 1; + u32 slot_width = sample_width; + /* * For captured data we should not rotate, inversion and masking is * enoguh to get the data to the right position: @@ -699,31 +713,23 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, u32 rx_rotate = 0; /* - * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() - * callback, take it into account here. That allows us to for example - * send 32 bits per channel to the codec, while only 16 of them carry - * audio payload. - * The clock ratio is given for a full period of data (for I2S format - * both left and right channels), so it has to be divided by number of - * tdm-slots (for I2S - divided by 2). + * Setting the tdm slot width either with set_clkdiv() or + * set_tdm_slot() allows us to for example send 32 bits per + * channel to the codec, while only 16 of them carry audio + * payload. */ - if (mcasp->bclk_lrclk_ratio) { - u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots; - + if (mcasp->slot_width) { /* - * When we have more bclk then it is needed for the data, we - * need to use the rotation to move the received samples to have - * correct alignment. + * When we have more bclk then it is needed for the + * data, we need to use the rotation to move the + * received samples to have correct alignment. */ - rx_rotate = (slot_length - word_length) / 4; - word_length = slot_length; - } else if (mcasp->slot_width) { - rx_rotate = (mcasp->slot_width - word_length) / 4; - word_length = mcasp->slot_width; + slot_width = mcasp->slot_width; + rx_rotate = (slot_width - sample_width) / 4; } /* mapping of the XSSZ bit-field as described in the datasheet */ - fmt = (word_length >> 1) - 1; + fmt = (slot_width >> 1) - 1; if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), -- 1.9.1

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