June 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/2] snd-firewire-lib: add MIDI stream support
by o-takashi＠sakamocchi.jp 05 Jun '13

05 Jun '13

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> This series of patch add MIDI stream support to snd-firewire-lib according to IEC 61883-6 and MMA/AMEI RP-027. This series of patch is based on my previous patches for snd-firewire-lib. [PATCH v2 0/3] snd-firewire-lib: add handling CMP output connection http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061607.html [PATCH v2 0/4] snd-firewire-lib: add handling AMDTP receive stream http://mailman.alsa-project.org/pipermail/alsa-devel/2013-April/061611.html Takashi Sakamoto (2): Add amdtp_stream_running() and amdtp_stream_pcm_running() Add MIDI stream support with data rate restriction sound/firewire/amdtp.c | 171 +++++++++++++++++++++++++++++++++++++++++++----- sound/firewire/amdtp.h | 37 +++++++++++ 2 files changed, 193 insertions(+), 15 deletions(-) -- 1.7.10.4

3 5

Re: [alsa-devel] [PATCH] ASoC: add RT5640 CODEC driver
by Stephen Warren 05 Jun '13

05 Jun '13

On 05/31/2013 01:04 AM, bardliao(a)realtek.com wrote: > From: Bard Liao <bardliao(a)realtek.com> > > * Use regmap_range_cfg to replace index read/write function. > * Remove I2S3 related code since there is no I2S3 in ALC5640. > * Remove Voice DSP related code since there is no Voice DSP in ALC5640. > * Remove MICBIAS2 since there is no MICBIAS2 in ALC5640. > * Change DMIC1/2 CLK to DMIC1/2 Power since it is for enable/disable DMIC1/2 > * Modify some texts for consistent coding style. > * Merge sto adc l/r mux since it shares the same control bits. > * Other minor changes. > +static const struct snd_soc_dapm_widget rt5640_dapm_widgets[] = { > + /* Input Lines */ > + SND_SOC_DAPM_INPUT("DMIC1"), > + SND_SOC_DAPM_INPUT("DMIC2"), Can you explain that more? In the spec for this part, I don't see any DMIC1/2 input pins. Instead, I think they're alternative uses for the IN1P/N pins, right? > + SND_SOC_DAPM_MIXER("HPO MIX L", SND_SOC_NOPM, 0, 0, > + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), > + SND_SOC_DAPM_MIXER("HPO MIX R", SND_SOC_NOPM, 0, 0, > + rt5640_hpo_mix, ARRAY_SIZE(rt5640_hpo_mix)), In the version of the driver I posted, I had replaced this with a single "HPO MIX" DAPM_MIXER, because there is a single bit in HW that controls both the L and R channels. I think you should incorporate that same change. That relies on commit 85762e7 "ASoC: dapm: Implement mixer control sharing". In an earlier review, IIRC, Mark had asked you to implement platform data to configure whether the microphones were single-ended or not. I don't see that in this patch. Finally, I really would prefer if you could implement the support for the LDO1_EN GPIO via platform data. Getting that right requires a bit of knowledge of how set_bias_level() is supposed to work on this CODEC, and I think Realtek have the best information to get that right. I'll certainly send a patch to implement this all by device tree after that. Anyway, I'll go test this patch and see if it works for me on my HW.

2 2

[alsa-devel] [PATCH] ASoC: fsl: add imx-wm8962 machine driver
by Nicolin Chen 05 Jun '13

05 Jun '13

This is the initial imx-wm8962 device-tree-only machine driver working with fsl_ssi driver. More features can be added on top of it later. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- .../devicetree/bindings/sound/imx-audio-wm8962.txt | 61 +++ sound/soc/fsl/Kconfig | 12 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8962.c | 387 ++++++++++++++++++++ 4 files changed, 462 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt create mode 100644 sound/soc/fsl/imx-wm8962.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt new file mode 100644 index 0000000..cb188a6 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8962.txt @@ -0,0 +1,61 @@ +Freescale i.MX audio complex with WM8962 codec + +Required properties: +- compatible : "fsl,imx-audio-wm8962" +- model : The user-visible name of this sound complex +- ssi-controller : The phandle of the i.MX SSI controller +- audio-codec : The phandle of the WM8962 audio codec +- audio-routing : A list of the connections between audio components. + Each entry is a pair of strings, the first being the connection's sink, + the second being the connection's source. Valid names could be power + supplies, WM8962 pins, and the jacks on the board: + + Power supplies: + * Mic Bias + + WM8962 pins: + * HPOUTL + * HPOUTR + * SPKOUTL + * SPKOUTR + * MICBIAS + * IN3R + * DMIC + * DMICDAT + + Board connectors: + * Mic Jack + * Headphone Jack + * Ext Spk + +- mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) +- mux-ext-port : The external port of the i.MX audio muxer + +- hp-det-gpios : The gpio port of Headphone detection. +- mic-det-gpios: The gpio port of Micphone detection. + +Note: The AUDMUX port numbering should start at 1, which is consistent with +hardware manual. + +Example: + +sound { + compatible = "fsl,imx6q-sabresd-wm8962", + "fsl,imx-audio-wm8962"; + model = "wm8962-audio"; + ssi-controller = <&ssi2>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUTL", + "Headphone Jack", "HPOUTR", + "Ext Spk", "SPKOUTL", + "Ext Spk", "SPKOUTR", + "MICBIAS", "AMIC", + "IN3R", "MICBIAS", + "DMIC", "MICBIAS", + "DMICDAT", "DMIC"; + mux-int-port = <2>; + mux-ext-port = <3>; + hp-det-gpios = <&gpio7 8 1>; /* active low */ + mic-det-gpios = <&gpio1 9 1>; /* active low */ +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 7860cc2..aa43854 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -168,6 +168,18 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8962 + tristate "SoC Audio support for i.MX boards with wm8962" + depends on OF && I2C + select SND_SOC_WM8962 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_IMX_AUDMUX + select SND_SOC_FSL_SSI + select SND_SOC_FSL_UTILS + help + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8962 codec. + config SND_SOC_IMX_SGTL5000 tristate "SoC Audio support for i.MX boards with sgtl5000" depends on OF && I2C diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index 91883f8..d4b4aa8 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -42,6 +42,7 @@ snd-soc-phycore-ac97-objs := phycore-ac97.o snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-mc13783-objs := imx-mc13783.o obj-$(CONFIG_SND_SOC_EUKREA_TLV320) += snd-soc-eukrea-tlv320.o @@ -49,4 +50,5 @@ obj-$(CONFIG_SND_SOC_PHYCORE_AC97) += snd-soc-phycore-ac97.o obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8962.c b/sound/soc/fsl/imx-wm8962.c new file mode 100644 index 0000000..7aa42d8 --- /dev/null +++ b/sound/soc/fsl/imx-wm8962.c @@ -0,0 +1,387 @@ +/* + * Copyright 2013 Freescale Semiconductor, Inc. + * + * Based on imx-sgtl5000.c + * Copyright 2012 Freescale Semiconductor, Inc. + * Copyright 2012 Linaro Ltd. + * + * The code contained herein is licensed under the GNU General Public + * License. You may obtain a copy of the GNU General Public License + * Version 2 or later at the following locations: + * + * http://www.opensource.org/licenses/gpl-license.html + * http://www.gnu.org/copyleft/gpl.html + */ + +#include <linux/module.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_i2c.h> +#include <linux/of_gpio.h> +#include <linux/slab.h> +#include <linux/clk.h> +#include <linux/irq.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/pinctrl/consumer.h> + +#include "../codecs/wm8962.h" +#include "imx-audmux.h" + +#define DAI_NAME_SIZE 32 + +struct imx_wm8962_data { + struct snd_soc_dai_link dai; + struct snd_soc_card card; + char codec_dai_name[DAI_NAME_SIZE]; + char platform_name[DAI_NAME_SIZE]; + struct clk *codec_clk; + unsigned int clk_frequency; +}; + +struct imx_priv { + int hp_gpio; + int hp_active_low; + int hp_status; + int amic_gpio; + int amic_active_low; + int amic_status; + struct platform_device *pdev; + struct snd_pcm_substream *first_stream; + struct snd_pcm_substream *second_stream; +}; +static struct imx_priv card_priv; + +static const struct snd_soc_dapm_widget imx_wm8962_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), + SND_SOC_DAPM_MIC("AMIC", NULL), + SND_SOC_DAPM_MIC("DMIC", NULL), +}; + +static int check_hw_params(struct snd_pcm_substream *substream, + snd_pcm_format_t sample_format, unsigned int sample_rate) +{ + struct imx_priv *priv = &card_priv; + struct device *dev = substream->pcm->card->dev; + + substream->runtime->sample_bits = + snd_pcm_format_physical_width(sample_format); + substream->runtime->rate = sample_rate; + substream->runtime->format = sample_format; + + if (!priv->first_stream) { + priv->first_stream = substream; + } else { + priv->second_stream = substream; + + if (priv->first_stream->runtime->rate != + priv->second_stream->runtime->rate) { + dev_err(dev, "KEEP THE SAME SAMPLE RATE: %d!\n", + priv->first_stream->runtime->rate); + return -EINVAL; + } + if (priv->first_stream->runtime->sample_bits != + priv->second_stream->runtime->sample_bits) { + snd_pcm_format_t first_format = + priv->first_stream->runtime->format; + + dev_err(dev, "KEEP THE SAME FORMAT: %s!\n", + snd_pcm_format_name(first_format)); + return -EINVAL; + } + } + + return 0; +} + +static int imx_hifi_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 *codec_dai = rtd->codec_dai; + struct device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + unsigned int sample_rate = params_rate(params); + unsigned int sample_format = params_format(params); + int ret = 0; + unsigned int pll_out; + + /* + * WM8962 doesn't support two substreams in different sample rates or + * sample formats. So check the two parameters of two substreams' when + * there're two substreams of playback and of capture in the same time. + */ + ret = check_hw_params(substream, sample_format, sample_rate); + if (ret < 0) { + dev_err(dev, "Failed to match hw params: %d\n", ret); + return ret; + } + + if (sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = sample_rate * 384; + else + pll_out = sample_rate * 256; + + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, WM8962_FLL_MCLK, + data->clk_frequency, pll_out); + if (ret < 0) { + dev_err(dev, "Failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_FLL, + pll_out, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_hifi_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + struct device *dev = substream->pcm->card->dev; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + /* We don't need to handle anything if there's no substream running */ + if (!priv->first_stream) + return 0; + + if (priv->first_stream == substream) + priv->first_stream = priv->second_stream; + priv->second_stream = NULL; + + if (!priv->first_stream) { + /* + * SYSCLK of wm8962's not disabled yet. If we wanna close FLL, + * we need to keep SYSCLK alive by changing its source to MCLK. + */ + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(dev, "Failed to set SYSCLK: %d\n", ret); + return ret; + } + + /* Disable FLL and let codec do pm_runtime_put() */ + ret = snd_soc_dai_set_pll(codec_dai, WM8962_FLL, 0, 0, 0); + if (ret < 0) { + dev_err(dev, "Failed to disable FLL: %d\n", ret); + return ret; + } + } + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_hifi_hw_params, + .hw_free = imx_hifi_hw_free, +}; + +static int imx_wm8962_late_probe(struct snd_soc_card *card) +{ + struct snd_soc_dai *codec_dai = card->rtd[0].codec_dai; + struct imx_priv *priv = &card_priv; + struct imx_wm8962_data *data = platform_get_drvdata(priv->pdev); + int ret; + + ret = snd_soc_dai_set_sysclk(codec_dai, WM8962_SYSCLK_MCLK, + data->clk_frequency, SND_SOC_CLOCK_IN); + if (ret < 0) + dev_err(card->dev, "Failed to set sysclk in %s\n", __func__); + + return ret; +} + +static int imx_wm8962_probe(struct platform_device *pdev) +{ + struct device_node *np = pdev->dev.of_node; + struct device_node *ssi_np, *codec_np; + struct platform_device *ssi_pdev; + struct imx_priv *priv = &card_priv; + struct i2c_client *codec_dev; + struct imx_wm8962_data *data; + int int_port, ext_port; + int ret; + + priv->pdev = pdev; + + ret = of_property_read_u32(np, "mux-int-port", &int_port); + if (ret) { + dev_err(&pdev->dev, "mux-int-port missing or invalid\n"); + return ret; + } + ret = of_property_read_u32(np, "mux-ext-port", &ext_port); + if (ret) { + dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); + return ret; + } + + /* + * The port numbering in the hardware manual starts at 1, while + * the audmux API expects it starts at 0. + */ + int_port--; + ext_port--; + ret = imx_audmux_v2_configure_port(int_port, + IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR | + IMX_AUDMUX_V2_PTCR_TCLKDIR, + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); + if (ret) { + dev_err(&pdev->dev, "audmux internal port setup failed\n"); + return ret; + } + imx_audmux_v2_configure_port(ext_port, + IMX_AUDMUX_V2_PTCR_SYN, + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); + if (ret) { + dev_err(&pdev->dev, "audmux external port setup failed\n"); + return ret; + } + + ssi_np = of_parse_phandle(pdev->dev.of_node, "ssi-controller", 0); + codec_np = of_parse_phandle(pdev->dev.of_node, "audio-codec", 0); + if (!ssi_np || !codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + ssi_pdev = of_find_device_by_node(ssi_np); + if (!ssi_pdev) { + dev_err(&pdev->dev, "failed to find SSI platform device\n"); + ret = -EINVAL; + goto fail; + } + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + return -EINVAL; + } + + priv->hp_gpio = of_get_named_gpio_flags(np, "hp-det-gpios", 0, + (enum of_gpio_flags *)&priv->hp_active_low); + priv->amic_gpio = of_get_named_gpio_flags(np, "mic-det-gpios", 0, + (enum of_gpio_flags *)&priv->amic_active_low); + + priv->first_stream = NULL; + priv->second_stream = NULL; + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) { + ret = -ENOMEM; + goto fail; + } + + data->codec_clk = clk_get(&codec_dev->dev, NULL); + if (IS_ERR(data->codec_clk)) { + /* assuming clock enabled by default */ + data->codec_clk = NULL; + ret = of_property_read_u32(codec_np, "clock-frequency", + &data->clk_frequency); + if (ret) { + dev_err(&codec_dev->dev, + "clock-frequency missing or invalid\n"); + goto fail; + } + } else { + data->clk_frequency = clk_get_rate(data->codec_clk); + clk_prepare_enable(data->codec_clk); + } + + data->dai.name = "HiFi"; + data->dai.stream_name = "HiFi"; + data->dai.codec_dai_name = "wm8962"; + data->dai.codec_of_node = codec_np; + data->dai.cpu_dai_name = dev_name(&ssi_pdev->dev); + data->dai.platform_of_node = ssi_np; + data->dai.ops = &imx_hifi_ops, + data->dai.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM; + + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto clk_fail; + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto clk_fail; + data->card.num_links = 1; + data->card.dai_link = &data->dai; + data->card.dapm_widgets = imx_wm8962_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8962_dapm_widgets); + + data->card.late_probe = imx_wm8962_late_probe, + + ret = snd_soc_register_card(&data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto clk_fail; + } + + platform_set_drvdata(pdev, data); + of_node_put(ssi_np); + of_node_put(codec_np); + + return 0; + +clk_fail: + if (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(data->codec_clk); + } +fail: + if (ssi_np) + of_node_put(ssi_np); + if (codec_np) + of_node_put(codec_np); + + return ret; +} + +static int imx_wm8962_remove(struct platform_device *pdev) +{ + struct imx_wm8962_data *data = platform_get_drvdata(pdev); + + if (data->codec_clk) { + clk_disable_unprepare(data->codec_clk); + clk_put(data->codec_clk); + } + snd_soc_unregister_card(&data->card); + + return 0; +} + +static const struct of_device_id imx_wm8962_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8962", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8962_dt_ids); + +static struct platform_driver imx_wm8962_driver = { + .driver = { + .name = "imx-wm8962", + .owner = THIS_MODULE, + .of_match_table = imx_wm8962_dt_ids, + }, + .probe = imx_wm8962_probe, + .remove = imx_wm8962_remove, +}; +module_platform_driver(imx_wm8962_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8962 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8962"); -- 1.7.1

1 0

Re: [alsa-devel] USB audio Gadget - code 10 error
by anish singh 05 Jun '13

05 Jun '13

adding alsa-devel On Tue, Jun 4, 2013 at 10:35 PM, Eamon Hayden <eamon.hayden(a)cloudiumsystems.com> wrote: > I’m trying to use the linux usb audio gadget but I get the error : “This > device cannot start (code 10) when I connect to a windows 7/8 box. I don’t > have this problem when I connect to ubunto 12.0.4 > > > > I’m current using kernel 3.6.9, but have also had the issue with an older > kernel (2.6.39.4). > > > > I’m not sure if this is an appropriate issue for this mail list, but I need > to get a resolution to this... > > > > Thanks for your help. > > Eamon. > > > _______________________________________________ > Kernelnewbies mailing list > Kernelnewbies(a)kernelnewbies.org > http://lists.kernelnewbies.org/mailman/listinfo/kernelnewbies >

2 1

[alsa-devel] [PATCH] ASoC: tegra: add runtime PM to resume functions
by Stephen Warren 04 Jun '13

04 Jun '13

From: Stephen Warren <swarren(a)nvidia.com> Tegra HW needs clocks etc. active when touching registers. Make sure they are active during resume, by calling pm_runtime_get_sync() before touching HW. Signed-off-by: Stephen Warren <swarren(a)nvidia.com> --- sound/soc/tegra/tegra30_ahub.c | 4 ++++ sound/soc/tegra/tegra30_i2s.c | 9 ++++++++- 2 files changed, 12 insertions(+), 1 deletion(-) diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 0f4787c..d554d46 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -640,8 +640,12 @@ static int tegra30_ahub_resume(struct device *dev) { int ret; + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; ret = regcache_sync(ahub->regmap_ahub); ret |= regcache_sync(ahub->regmap_apbif); + pm_runtime_put(dev); return ret; } diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index bd0ebc0..d04146c 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -527,8 +527,15 @@ static int tegra30_i2s_suspend(struct device *dev) static int tegra30_i2s_resume(struct device *dev) { struct tegra30_i2s *i2s = dev_get_drvdata(dev); + int ret; - return regcache_sync(i2s->regmap); + ret = pm_runtime_get_sync(dev); + if (ret < 0) + return ret; + ret = regcache_sync(i2s->regmap); + pm_runtime_put(dev); + + return ret; } #endif -- 1.8.1.5

2 1

[alsa-devel] [PATCH V2] ASoC: sgtl5000: Fix VAG_POWER enabling/disabling order
by Marek Vasut 04 Jun '13

04 Jun '13

The VAG_POWER must be enabled after all other bits in CHIP_ANA_POWER and disabled before any other bit in CHIP_ANA_POWER. See the SGTL5000 datasheet (Table 31, BIT 7, page 42-43). Failing to follow this order will result in ugly loud "POP" noise at the end of playback. To achieve such order, use the _PRE and _POST DAPM widgets to trigger the power_vag_event, where the event type check has to be fixed accordingly as well. Signed-off-by: Marek Vasut <marex(a)denx.de> Cc: Dong Aisheng <dong.aisheng(a)linaro.org> Cc: Eric Nelson <eric.nelson(a)boundarydevices.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/sgtl5000.c | 14 +++++--------- 1 file changed, 5 insertions(+), 9 deletions(-) V2: Remove that mdelay/msleep regression diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 92bbfec..1994f01 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -151,12 +151,12 @@ static int power_vag_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { switch (event) { - case SND_SOC_DAPM_PRE_PMU: + case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, SGTL5000_VAG_POWERUP); break; - case SND_SOC_DAPM_POST_PMD: + case SND_SOC_DAPM_PRE_PMD: snd_soc_update_bits(w->codec, SGTL5000_CHIP_ANA_POWER, SGTL5000_VAG_POWERUP, 0); msleep(400); @@ -217,12 +217,11 @@ static const struct snd_soc_dapm_widget sgtl5000_dapm_widgets[] = { 0, SGTL5000_CHIP_DIG_POWER, 1, 0), - SND_SOC_DAPM_SUPPLY("VAG_POWER", SGTL5000_CHIP_ANA_POWER, 7, 0, - power_vag_event, - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_ADC("ADC", "Capture", SGTL5000_CHIP_ANA_POWER, 1, 0), SND_SOC_DAPM_DAC("DAC", "Playback", SGTL5000_CHIP_ANA_POWER, 3, 0), + + SND_SOC_DAPM_PRE("VAG_POWER_PRE", power_vag_event), + SND_SOC_DAPM_POST("VAG_POWER_POST", power_vag_event), }; /* routes for sgtl5000 */ @@ -230,16 +229,13 @@ static const struct snd_soc_dapm_route sgtl5000_dapm_routes[] = { {"Capture Mux", "LINE_IN", "LINE_IN"}, /* line_in --> adc_mux */ {"Capture Mux", "MIC_IN", "MIC_IN"}, /* mic_in --> adc_mux */ - {"ADC", NULL, "VAG_POWER"}, {"ADC", NULL, "Capture Mux"}, /* adc_mux --> adc */ {"AIFOUT", NULL, "ADC"}, /* adc --> i2s_out */ - {"DAC", NULL, "VAG_POWER"}, {"DAC", NULL, "AIFIN"}, /* i2s-->dac,skip audio mux */ {"Headphone Mux", "DAC", "DAC"}, /* dac --> hp_mux */ {"LO", NULL, "DAC"}, /* dac --> line_out */ - {"LINE_IN", NULL, "VAG_POWER"}, {"Headphone Mux", "LINE_IN", "LINE_IN"},/* line_in --> hp_mux */ {"HP", NULL, "Headphone Mux"}, /* hp_mux --> hp */ -- 1.7.10.4

2 1

[alsa-devel] [PATCH] ASoC: omap-hdmi-codec: make the driver common for other SoCs
by Barry Song 04 Jun '13

04 Jun '13

to support HDMI on CSR SiRFprimaII and atlasVI, we need one more HDMI pseudo codec, rather than add a new driver, we can make omap HDMI codec common for other SoCs as well. then the omap-hdmi codec becomes a generic HDMI pseudo- codec as HDMI audio features depend on HDMI specification not on SoCs. Cc: Ricardo Neri <ricardo.neri(a)ti.com> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{omap-hdmi.c => hdmi.c} | 26 +++++++++++++------------- sound/soc/omap/omap-hdmi-card.c | 2 +- 4 files changed, 18 insertions(+), 18 deletions(-) rename sound/soc/codecs/{omap-hdmi.c => hdmi.c} (69%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 2f45f00..d8c4f3d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -53,7 +53,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C - select SND_SOC_OMAP_HDMI_CODEC if OMAP4_DSS_HDMI + select SND_SOC_HDMI_CODEC select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_SGTL5000 if I2C @@ -287,7 +287,7 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_OMAP_HDMI_CODEC +config SND_SOC_HDMI_CODEC tristate config SND_SOC_PCM3008 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index b9e41c9..49ff127 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -41,7 +41,7 @@ snd-soc-max98095-objs := max98095.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o -snd-soc-omap-hdmi-codec-objs := omap-hdmi.o +snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -168,7 +168,7 @@ obj-$(CONFIG_SND_SOC_MAX98095) += snd-soc-max98095.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o -obj-$(CONFIG_SND_SOC_OMAP_HDMI_CODEC) += snd-soc-omap-hdmi-codec.o +obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/omap-hdmi.c b/sound/soc/codecs/hdmi.c similarity index 69% rename from sound/soc/codecs/omap-hdmi.c rename to sound/soc/codecs/hdmi.c index 529d064..2bcae2b 100644 --- a/sound/soc/codecs/omap-hdmi.c +++ b/sound/soc/codecs/hdmi.c @@ -1,5 +1,5 @@ /* - * ALSA SoC codec driver for HDMI audio on OMAP processors. + * ALSA SoC codec driver for HDMI audio codecs. * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/ * Author: Ricardo Neri <ricardo.neri(a)ti.com> * @@ -23,10 +23,10 @@ #define DRV_NAME "hdmi-audio-codec" -static struct snd_soc_codec_driver omap_hdmi_codec; +static struct snd_soc_codec_driver hdmi_codec; -static struct snd_soc_dai_driver omap_hdmi_codec_dai = { - .name = "omap-hdmi-hifi", +static struct snd_soc_dai_driver hdmi_codec_dai = { + .name = "hdmi-hifi", .playback = { .channels_min = 2, .channels_max = 8, @@ -39,31 +39,31 @@ static struct snd_soc_dai_driver omap_hdmi_codec_dai = { }, }; -static int omap_hdmi_codec_probe(struct platform_device *pdev) +static int hdmi_codec_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &omap_hdmi_codec, - &omap_hdmi_codec_dai, 1); + return snd_soc_register_codec(&pdev->dev, &hdmi_codec, + &hdmi_codec_dai, 1); } -static int omap_hdmi_codec_remove(struct platform_device *pdev) +static int hdmi_codec_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver omap_hdmi_codec_driver = { +static struct platform_driver hdmi_codec_driver = { .driver = { .name = DRV_NAME, .owner = THIS_MODULE, }, - .probe = omap_hdmi_codec_probe, - .remove = omap_hdmi_codec_remove, + .probe = hdmi_codec_probe, + .remove = hdmi_codec_remove, }; -module_platform_driver(omap_hdmi_codec_driver); +module_platform_driver(hdmi_codec_driver); MODULE_AUTHOR("Ricardo Neri <ricardo.neri(a)ti.com>"); -MODULE_DESCRIPTION("ASoC OMAP HDMI codec driver"); +MODULE_DESCRIPTION("ASoC generic HDMI codec driver"); MODULE_LICENSE("GPL"); MODULE_ALIAS("platform:" DRV_NAME); diff --git a/sound/soc/omap/omap-hdmi-card.c b/sound/soc/omap/omap-hdmi-card.c index d4eaa92..7e66e9c 100644 --- a/sound/soc/omap/omap-hdmi-card.c +++ b/sound/soc/omap/omap-hdmi-card.c @@ -35,7 +35,7 @@ static struct snd_soc_dai_link omap_hdmi_dai = { .cpu_dai_name = "omap-hdmi-audio-dai", .platform_name = "omap-pcm-audio", .codec_name = "hdmi-audio-codec", - .codec_dai_name = "omap-hdmi-hifi", + .codec_dai_name = "hdmi-hifi", }; static struct snd_soc_card snd_soc_omap_hdmi = { -- 1.8.2.3

2 1

[alsa-devel] [RFC PATCH] ALSA: hda - Ignore default pin config on Haswell HDMI codecs
by David Henningsson 04 Jun '13

04 Jun '13

According to Daniel Vetter, these pin configs should be ignored, because the graphics driver/userspace could route audio to these pins even if they are disabled by BIOS. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- sound/pci/hda/patch_hdmi.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) It would be good to have this acked by someone at Intel before committing. diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index e12f7a0..8c1bede 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1307,12 +1307,14 @@ static int hdmi_add_pin(struct hda_codec *codec, hda_nid_t pin_nid) if (!(caps & (AC_PINCAP_HDMI | AC_PINCAP_DP))) return 0; - config = snd_hda_codec_get_pincfg(codec, pin_nid); - if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) - return 0; - if (codec->vendor_id == 0x80862807) intel_haswell_fixup_connect_list(codec, pin_nid); + else { + /* On Haswell, default pin config should be ignored */ + config = snd_hda_codec_get_pincfg(codec, pin_nid); + if (get_defcfg_connect(config) == AC_JACK_PORT_NONE) + return 0; + } pin_idx = spec->num_pins; per_pin = snd_array_new(&spec->pins); -- 1.7.9.5

3 4

[alsa-devel] dsnoop memory leak
by Иван Больсунов 04 Jun '13

04 Jun '13

Since alsa bug tracker does not work for me for weeks, I have to report bug here. Sorry if it a wrong place. So, to reproduce memory leak (Alsa lib used v.1.0.27), you have to use next asound.conf: pcm_slave.myslave { pcm "hw:0,0" } pcm.mysnoop { type dsnoop ipc_key 1024 slave myslave } Then you simply do: snd_pcm_open(&pcm_handle, "mysnoop", SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK); snd_pcm_close(pcm_handle); snd_config_update_free_global(); Valgrind will show something like this: ==7889== ==7889== HEAP SUMMARY: ==7889== in use at exit: 15,854 bytes in 871 blocks ==7889== total heap usage: 333,212 allocs, 332,341 frees, 11,695,490 bytes allocated ==7889== ==7889== 15,834 (6,264 direct, 9,570 indirect) bytes in 174 blocks are definitely lost in loss record 6 of 6 ==7889== at 0x4027CBF: calloc (vg_replace_malloc.c:566) ==7889== by 0x406B416: _snd_config_make (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406C69C: snd_config_make (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406C861: snd_config_make_compound (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406D00C: _snd_config_copy (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406F112: snd_config_walk (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x406FE04: snd_config_copy (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x4070370: snd_config_expand (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x407107E: snd_config_search_definition (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D2C22: _snd_pcm_direct_get_slave_ipc_offset (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D68A2: snd1_pcm_direct_parse_open_conf (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40D28CA: _snd_pcm_dsnoop_open (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x40909C4: snd_pcm_open_conf (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x4091036: snd_pcm_open_noupdate (in /usr/lib/libasound.so.2.0.0) ==7889== by 0x804873D: main (in /home/blah/al/a.out) ==7889== ==7889== LEAK SUMMARY: ==7889== definitely lost: 6,264 bytes in 174 blocks ==7889== indirectly lost: 9,570 bytes in 696 blocks ==7889== possibly lost: 0 bytes in 0 blocks ==7889== still reachable: 20 bytes in 1 blocks ==7889== suppressed: 0 bytes in 0 blocks ==7889== Reachable blocks (those to which a pointer was found) are not shown. ==7889== To see them, rerun with: --leak-check=full --show-reachable=yes ==7889== ==7889== For counts of detected and suppressed errors, rerun with: -v ==7889== ERROR SUMMARY: 1 errors from 1 contexts (suppressed: 0 from 0)

2 1

[alsa-devel] dsnoop memory leak
by Ivan Bolsunov 04 Jun '13

04 Jun '13

Hi guys, on may 31 I sent message, can you at least confirm what bug exist? Thanks.

1 0