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

[alsa-devel] [PATCH 00/15] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 08 Jan '16

08 Jan '16

This patch series adds DP audio and hotplug notification support. On Skylake two DP ports are available and to enable DP on both ports all pins need to be enabled. There is a special vendor widget which need to be programmed to enable all pins and converters. This series adds hotplug notification, read/set constraint based on ELD, enable all pin/cvts, DP1.2, programs the audio infoframe for DP. There is a one to one mapping between converter and stream, so the dais are created based on the no of streams supported on hdmi codec. Even though cvts can be mapped dynamically to the streams, currently it is statically mapped as simultaneous playback on both DP and HDMI is not supported as of now. Pin muxes and controls are created dynamically to map converter to pin widget. So at run time specific pin is mapped to the dai based on the control selected (based on the display type DP/HDMI connected). Finally the DP audio infoframe programming is added to support the DP feature. Also with hotplug notification support, ELD is read and capabilities are set for rate, formats and channels. drm_eld sound/core framework is updated to limit the formats based on ELD. There are few fixes one fixing the static checker warning and other one not to fail if no connection list is found for a pin widget. Pls note, the 12th patch is adding a small macro for getting connection type in drm header, we have CCed drm folks on that and this one. Pls ack so that we can have this series merged thru sound trees Jeeja KP (1): ASoC: hdac_hdmi: Add codec suspend/resume handler Ramesh Babu (1): ASoC: hdac_hdmi: Keep display active while enumerating codec Subhransu S. Prusty (13): ASoC: hdac_hdmi: Fix to check num nodes correctly ASoC: hdac_hdmi: Fix to warn instead of err for no connected nids ASoC: hdac_hdmi - Use list to add pins and converters ALSA: hda - Add helper to read eld data ASoC: hdac_hdmi: Add hotplug notification and read eld ALSA: pcm: Add DRM helper to set constraint for format ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi - create dais based on number of streams ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Assign pin for stream based on dapm connection drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio include/drm/drm_edid.h | 10 + include/sound/hdaudio.h | 3 + sound/core/pcm_drm_eld.c | 42 +- sound/hda/Makefile | 2 +- sound/hda/hdac_eld.c | 95 +++++ sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 908 ++++++++++++++++++++++++++++++++++++------- 7 files changed, 908 insertions(+), 153 deletions(-) create mode 100644 sound/hda/hdac_eld.c -- 1.9.1

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[alsa-devel] [PATCH v3 00/15] ASoC: hdac_hdmi: Add DP & notification support
by Subhransu S. Prusty 08 Jan '16

08 Jan '16

This patch series adds DP audio and hotplug notification support. On Skylake two DP ports are available and to enable DP on both ports all pins need to be enabled. There is a special vendor widget which need to be programmed to enable all pins and converters. This series adds hotplug notification, read/set constraint based on ELD, enable all pin/cvts, DP1.2, programs the audio infoframe for DP. There is a one to one mapping between converter and stream, so the dais are created based on the no of streams supported on hdmi codec. Even though cvts can be mapped dynamically to the streams, currently it is statically mapped as simultaneous playback on both DP and HDMI is not supported as of now. Pin muxes and controls are created dynamically to map converter to pin widget. So at run time specific pin is mapped to the dai based on the control selected (based on the display type DP/HDMI connected). Finally the DP audio infoframe programming is added to support the DP feature. Also with hotplug notification support, ELD is read and capabilities are set for rate, formats and channels. drm_eld sound/core framework is updated to limit the formats based on ELD. There are few fixes one fixing the static checker warning and other one not to fail if no connection list is found for a pin widget. Pls note, the 10th patch is adding a small macro for getting connection type in drm header, we have CCed drm folks on that and this one. Pls ack so that we can have this series merged thru sound trees changes in v3: - Addressed review comments - Use of list_for_each_entry_safe for node deletion. - Moved the format constraint based on ELD to driver - Fix constant as per kernel doc for connection type in drm helper - Fix dp infoframe - Add hotplug event reporting to user space - Move device configuration from dai startup to hw_params as the user space can keep the device open always. Jeeja KP (2): ASoC: hdac_hdmi: Add codec suspend/resume handler ASoC: hdac_hdmi: Fix to enable device configuration in hw_params Ramesh Babu (1): ASoC: hdac_hdmi: Fix to keep display active while enumerating codec Subhransu S. Prusty (12): ASoC: hdac_hdmi: Fix to check num nodes correctly ASoC: hdac_hdmi: Fix to warn instead of err for no connected nids ASoC: hdac_hdmi: Use list to add pins and converters ASoC: hdac_hdmi: Add hotplug notification and read eld ASoC: hdac_hdmi: Apply constraints based on ELD ASoC: hdac_hdmi: Enable DP1.2 and all converters/pins ASoC: hdac_hdmi: create dais based on number of streams ASoC: hdac_hdmi: Create widget/route based on nodes enumerated ASoC: hdac_hdmi: Assign pin for stream based on dapm connection drm/edid: Add API to help find connection type ASoC: hdac_hdmi: Add infoframe support for dp audio ASoC: hdac_hdmi: Add jack reporting for user space include/drm/drm_edid.h | 12 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/hdac_hdmi.c | 1101 +++++++++++++++++++++++++++++++++++------- 3 files changed, 936 insertions(+), 178 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 0/4] ALSA: hda - hdmi jack support for dynamic pcm assignment
by libin.yang＠linux.intel.com 08 Jan '16

08 Jan '16

From: Libin Yang <libin.yang(a)linux.intel.com> The patchset adds hdmi jack support for dynamic pcm assignment Libin Yang (4): ALSA: Add documentation about HD-audio DP MST ALSA: hda - add hdmi_pcm to manage hdmi pcm related features ALSA: hda - hdmi jack created based on pcm ALSA: hda - hdmi monitor hotplug support for dynamic pcm assignment Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt | 75 +++++++ sound/pci/hda/patch_hdmi.c | 229 +++++++++++++++------ 2 files changed, 243 insertions(+), 61 deletions(-) create mode 100644 Documentation/sound/alsa/HD-Audio-DP-MST-audio.txt -- 1.9.1

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[alsa-devel] [PATCH 00/12] dw-hdmi development
by Russell King - ARM Linux 07 Jan '16

07 Jan '16

This sub-series is a mixture of development: * Removing the incorrect pixel repetition configuration code * Preventing pixel-doubled modes from being used * Adding interlaced video support * Implementing the sink_is_hdmi/sink_has_audio flags I suggested a few months ago * Only enabling audio support if the sink indicates it has audio * Avoiding double-enabling the HDMI interface * Fixing the mis-leading name of "dw_hdmi_phy_enable_power" * Adding connector mode forcing (important if your monitor bounces RXSENSE and HPD signals while in low-power mode.) * Improving the HDMI enable/disabling on sink status For review (and testing if people feel like it). Acks/tested-bys etc welcome. It applies on top of my drm-dwhdmi-devel branch, which is waiting for David Airlie to pull (see pull request on dri-devel, 15th July.) drivers/gpu/drm/bridge/dw_hdmi.c | 275 ++++++++++++++++++++++++++++++--------- drivers/gpu/ipu-v3/ipu-dc.c | 18 ++- drivers/gpu/ipu-v3/ipu-di.c | 129 +++++++++--------- 3 files changed, 291 insertions(+), 131 deletions(-) -- FTTC broadband for 0.8mile line: currently at 10.5Mbps down 400kbps up according to speedtest.net.

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[alsa-devel] [PATCH 0/4 v5] ALSA: dice: improve card registration processing
by Takashi Sakamoto 07 Jan '16

07 Jan '16

Hi, This patchset updates my previous one: [alsa-devel] [PATCH 0/5 v4] ALSA: dice: improve card registration processing http://mailman.alsa-project.org/pipermail/alsa-devel/2015-December/102504.h… Changes: * Merge patch 02 and 03 in v4. * Allocate sound card instance in sound card registration work, independently of dice instance, allocated in .probe callback and set as driver_data of device structure. * Use __be32 type for __be32 data. In this version, sound card instance is not allocated in .probe callback. In the callback, dice instance is allocated independently of the card instance. Thus, in .probe callback, this driver reserve no slot of sound card. The allocation of sound card instance is performed in a scheduled work. When processing in the work fails, the work can be re-scheduled again according to bus reset. Takashi Sakamoto (4): ALSA: dice: split subaddress check from category check ALSA: dice: postpone card registration ALSA: dice: purge transaction initialization at timeout of Dice notification ALSA: dice: expand timeout to wait for Dice notification sound/firewire/dice/dice-transaction.c | 123 +++++++++++------- sound/firewire/dice/dice.c | 227 +++++++++++++++++---------------- sound/firewire/dice/dice.h | 3 + 3 files changed, 202 insertions(+), 151 deletions(-) -- 2.5.0

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[alsa-devel] [PATCH v3 0/8] Add support for voice control on Arizona ADSP
by Charles Keepax 06 Jan '16

06 Jan '16

This series add support for voice control on wm8280/5110. This is done by opening a compressed record channel, upon which data will be available once the voice control functionality has been triggered on the DSP. Changes since v2: - Fix specified using in printf for size_t in wm_adsp_compr_read, as reported by the kbuild test robot. Thanks, Charles Charles Keepax (8): ASoC: wm5110: Provide basic hookup for voice control ASoC: wm_adsp: Factor out finding the location of an algorithm region ALSA: compress: Add SND_AUDIOCODEC_BESPOKE ASoC: wm_adsp: Add support for opening a compressed stream ASoC: wm_adsp: Add code to locate and initialise compressed buffer ASoC: wm_adsp: Attach buffers and streams together ASoC: wm_adsp: Add a handler for the compressed IRQ ASoC: wm_adsp: Pull data through compressed read include/uapi/sound/compress_params.h | 5 +- sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/arizona.h | 2 +- sound/soc/codecs/wm5110.c | 95 +++- sound/soc/codecs/wm_adsp.c | 907 ++++++++++++++++++++++++++++++++++- sound/soc/codecs/wm_adsp.h | 21 + 6 files changed, 1010 insertions(+), 21 deletions(-) -- 2.1.4

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[alsa-devel] [PATCH 00/19] Baytrail/Cherrytrail cleanups and fixes
by Pierre-Louis Bossart 05 Jan '16

05 Jan '16

Set of patches to provide better upstream support for BYT/CHT platforms (24 bit, Deep-buffer, DPCM). This was tested on Asus T100 and Minnowboard. A set of UCM files is available at https://github.com/plbossart/UCM.git for better integration with Ubuntu/pulseaudio (no need for asound.state or asound.conf) Acknowledgements: Keyon Jie for his MinnowBoard validation Sebastien Guiriec for pointing out the switch logic issue and fixing SSP configs Vinod Koul for providing codec/mixer configurations Francisco Mendez for his help with the MCLK and jack detection Jack Yu and Bard Liao for their codec support Known issues: noise on startup when using speakers Once this batch is merged, the following updates will be provided: MCLK support to fix noise issues BYT-CR fixes jack detection Jack Yu (1): ASoC: rt5640: add ASRC support Jorge Fernandez Monteagudo (1): ASoC: Intel: Atom: Add support for HP ElitePad 1000 G2 Pierre-Louis Bossart (16): ASoC: Intel: fix ACPI probe regression with Atom DPCM driver ASoC: Intel: boards: align pin names between byt-rt5640 drivers ASoC: Intel: boards: start merging byt-rt5640 drivers ASoC: Intel: boards: merge DMI-based quirks in bytcr-rt5640 driver ASoC: Intel: tag byt-rt5640 machine driver as deprecated ASoc: Intel: Atom: add deep buffer definitions for atom platforms ASoC: Intel: boards: add DEEP_BUFFER support for BYT/CHT/BSW ASoC: Intel: Atom: add 24-bit support for media playback and capture ASoC: Intel: Atom: clean-up compressed DAI definition ASoC: Intel: Atom: flip logic for gain Switch ASoC: codec: rt5651: add ACPI and OF support ASoC: Intel: add bytct-rt5651 machine driver ASoC: intel: boards: add card for MinnowBoard I2S access ASoC: Intel: bytcr-rt5640: enable ASRC ASoC: Intel: bytcr_rt5640: fixup DAI codec_name with HID ASoC: Intel: Atom: add support for RT5642 Sebastien Guiriec (1): ASoC: Intel: bytcr_rt5640: set SSP to I2S mode 2ch arch/x86/include/asm/platform_sst_audio.h | 1 + sound/soc/codecs/rt5640.c | 103 ++++++++ sound/soc/codecs/rt5640.h | 17 ++ sound/soc/codecs/rt5651.c | 19 ++ sound/soc/intel/Kconfig | 35 ++- sound/soc/intel/atom/sst-atom-controls.c | 5 +- sound/soc/intel/atom/sst-atom-controls.h | 1 + sound/soc/intel/atom/sst-mfld-platform-pcm.c | 20 +- sound/soc/intel/atom/sst/sst_acpi.c | 25 +- sound/soc/intel/boards/Makefile | 4 + sound/soc/intel/boards/bytcr_nocodec.c | 233 ++++++++++++++++++ sound/soc/intel/boards/bytcr_rt5640.c | 266 ++++++++++++++++----- sound/soc/intel/boards/bytcr_rt5651.c | 338 +++++++++++++++++++++++++++ sound/soc/intel/boards/cht_bsw_max98090_ti.c | 12 + sound/soc/intel/boards/cht_bsw_rt5645.c | 12 + sound/soc/intel/boards/cht_bsw_rt5672.c | 12 + sound/soc/intel/common/Makefile | 5 + 17 files changed, 1036 insertions(+), 72 deletions(-) create mode 100644 sound/soc/intel/boards/bytcr_nocodec.c create mode 100644 sound/soc/intel/boards/bytcr_rt5651.c -- 1.9.1

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[alsa-devel] [PATCH v2] ASoC: fsl_asrc: old version asrc only support even numbered channels
by Zidan Wang 05 Jan '16

05 Jan '16

The latest asrc's channel bits is 4 bit, the old asrc's channel bits is 3. For the old version asrc, channel should be a even number. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- v1->v2: 1. add an extra else if for the new condition. sound/soc/fsl/fsl_asrc.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/fsl/fsl_asrc.c b/sound/soc/fsl/fsl_asrc.c index be78dc9..d7d81cf 100644 --- a/sound/soc/fsl/fsl_asrc.c +++ b/sound/soc/fsl/fsl_asrc.c @@ -116,6 +116,9 @@ static int fsl_asrc_request_pair(int channels, struct fsl_asrc_pair *pair) } else if (asrc_priv->channel_avail < channels) { dev_err(dev, "can't afford required channels: %d\n", channels); ret = -EINVAL; + } else if (asrc_priv->channel_bits < 4 && channels % 2 != 0) { + dev_err(dev, "old version asrc can't support an odd channel number"); + ret = -EINVAL; } else { asrc_priv->channel_avail -= channels; asrc_priv->pair[index] = pair; -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: imx-wm8958: add imx-wm8958 machine driver
by Zidan Wang 05 Jan '16

05 Jan '16

This is the initial imx-wm8958 device-tree-only machine driver working with fsl_sai driver. This sound card has three dai link, hifi, voice and bt. Hifi dai link will support codec dai master and slave mode. Voice an bt dai link have dummy cpu dai, and just support codec dai master mode. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- .../devicetree/bindings/sound/imx-audio-wm8958.txt | 74 ++++ sound/soc/fsl/Kconfig | 14 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/imx-wm8958.c | 418 +++++++++++++++++++++ 4 files changed, 508 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt create mode 100644 sound/soc/fsl/imx-wm8958.c diff --git a/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt new file mode 100644 index 0000000..81775e4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/imx-audio-wm8958.txt @@ -0,0 +1,74 @@ +Freescale i.MX audio complex with WM8958 codec + +Required properties: + + - compatible : "fsl,imx-audio-wm8958" + + - model : The user-visible name of this sound complex + + - cpu-dai : The phandle of an CPU DAI controller + + - audio-codec : The phandle of the WM8958 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, WM8958 pins, and the jacks on the board: + + Power supplies: + * MICBIAS1 + * MICBIAS2 + + Board connectors: + * Headphone Jack + * Ext Spk + + - aifx-format : set format for aifx, such as "i2s", "left_j", + "dsp_a", "dsp_b". + + - aifx-bitclock-master : If present, aifx wroks as bit clock master. + Otherwise, aifx works as bit clock slave. + + - aifx-frame-master : If present, aifx wroks as frame clock master. + Otherwise, aifx works as frame clock slave. + + - aifx-bitclock-inversion : If present, bit clock will invert polarity. + + - aifx-frame-inversion : If present, frame clock will invert polarity. + + - aifx-continuous-clock : If present, bit clock will be continuous. + Otherwise, dai bit clocks can be be gated + (disabled) when the dai is not sending or + receiving PCM data in a frame. + +Example: + +sound { + compatible = "fsl,imx6ul-evk-wm8958", + "fsl,imx-audio-wm8958"; + model = "wm8960-audio"; + cpu-dai = <&sai1>; + audio-codec = <&codec>; + audio-routing = + "Headphone Jack", "HPOUT1L", + "Headphone Jack", "HPOUT1R", + "Ext Spk", "SPKOUTLP", + "Ext Spk", "SPKOUTLN", + "Ext Spk", "SPKOUTRP", + "Ext Spk", "SPKOUTRN", + "IN1LN", "MICBIAS2"; + + /* set aif1 dai format */ + aif1-format = "i2s"; + aif1-bitclock-master; + aif1-frame-master; + /* set aif2 dai format */ + aif2-format = "i2s"; + aif2-bitclock-master; + aif2-frame-master; + /* set aif3 dai format */ + aif3-format = "i2s"; + aif3-bitclock-master; + aif3-frame-master; +}; diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig index 14dfdee..53041cc 100644 --- a/sound/soc/fsl/Kconfig +++ b/sound/soc/fsl/Kconfig @@ -229,6 +229,20 @@ config SND_SOC_EUKREA_TLV320 Enable I2S based access to the TLV320AIC23B codec attached to the SSI interface +config SND_SOC_IMX_WM8958 + tristate "SoC Audio support for i.MX boards with wm8958" + depends on OF && I2C + select MFD_WM8994 + select SND_SOC_WM8994 + select SND_SOC_IMX_PCM_DMA + select SND_SOC_FSL_SAI + select SND_SOC_FSL_UTILS + select SND_KCTL_JACK + help + SoC Audio support for i.MX boards with WM8958 + Say Y if you want to add support for SoC audio on an i.MX board with + a wm8958 codec. + config SND_SOC_IMX_WM8962 tristate "SoC Audio support for i.MX boards with wm8962" depends on OF && I2C && INPUT diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile index d28dc25..2a781c0 100644 --- a/sound/soc/fsl/Makefile +++ b/sound/soc/fsl/Makefile @@ -54,6 +54,7 @@ snd-soc-mx27vis-aic32x4-objs := mx27vis-aic32x4.o snd-soc-wm1133-ev1-objs := wm1133-ev1.o snd-soc-imx-es8328-objs := imx-es8328.o snd-soc-imx-sgtl5000-objs := imx-sgtl5000.o +snd-soc-imx-wm8958-objs := imx-wm8958.o snd-soc-imx-wm8962-objs := imx-wm8962.o snd-soc-imx-spdif-objs := imx-spdif.o snd-soc-imx-mc13783-objs := imx-mc13783.o @@ -64,6 +65,7 @@ obj-$(CONFIG_SND_SOC_MX27VIS_AIC32X4) += snd-soc-mx27vis-aic32x4.o obj-$(CONFIG_SND_MXC_SOC_WM1133_EV1) += snd-soc-wm1133-ev1.o obj-$(CONFIG_SND_SOC_IMX_ES8328) += snd-soc-imx-es8328.o obj-$(CONFIG_SND_SOC_IMX_SGTL5000) += snd-soc-imx-sgtl5000.o +obj-$(CONFIG_SND_SOC_IMX_WM8958) += snd-soc-imx-wm8958.o obj-$(CONFIG_SND_SOC_IMX_WM8962) += snd-soc-imx-wm8962.o obj-$(CONFIG_SND_SOC_IMX_SPDIF) += snd-soc-imx-spdif.o obj-$(CONFIG_SND_SOC_IMX_MC13783) += snd-soc-imx-mc13783.o diff --git a/sound/soc/fsl/imx-wm8958.c b/sound/soc/fsl/imx-wm8958.c new file mode 100644 index 0000000..920bc9e --- /dev/null +++ b/sound/soc/fsl/imx-wm8958.c @@ -0,0 +1,418 @@ +/* + * Copyright (C) 2015 Freescale Semiconductor, Inc. + * + * 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_platform.h> +#include <linux/i2c.h> +#include <linux/of_gpio.h> +#include <linux/gpio.h> +#include <linux/clk.h> +#include <sound/soc.h> +#include <sound/jack.h> +#include <sound/control.h> +#include <sound/pcm_params.h> +#include <sound/soc-dapm.h> +#include <linux/mfd/wm8994/registers.h> +#include "../fsl/fsl_sai.h" +#include "../codecs/wm8994.h" + +#define DAI_NAME_SIZE 32 + +#define DAI_LINK_NUM (3) +#define HIFI_DAI (0) +#define VOICE_DAI (1) +#define BT_DAI (2) + +#define WM8958_MCLK_MAX (2) + +#define WM8994_FLL(id) (id == HIFI_DAI ? WM8994_FLL1 : WM8994_FLL2) +#define WM8994_SYSCLK_FLL(id) (id == HIFI_DAI ? WM8994_SYSCLK_FLL1 : WM8994_SYSCLK_FLL2) +#define WM8994_FLL_SRC_MCLK(id) (id == HIFI_DAI ? WM8994_FLL_SRC_MCLK1: WM8994_FLL_SRC_MCLK2) + +struct imx_wm8958_data { + struct snd_soc_dai_link *dai_link;; + struct snd_soc_card card; + struct clk *mclk[WM8958_MCLK_MAX]; + u32 mclk_freq[WM8958_MCLK_MAX]; + bool is_hifi_dai_master; + bool is_stream_in_use[DAI_LINK_NUM][2]; +}; + +static const struct snd_soc_dapm_widget imx_wm8958_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_SPK("Ext Spk", NULL), +}; + +static int imx_wm8958_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 snd_soc_dai *cpu_dai = rtd->cpu_dai; + struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_card *card = rtd->card; + struct device *dev = card->dev; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + bool hifi_dai_sysclk_dir = SND_SOC_CLOCK_OUT; + u32 mclk_id, id = codec_dai->id - 1; + u32 pll_out; + int ret; + + data->is_stream_in_use[id][tx] = true; + + if (data->mclk_freq[id]) + mclk_id = WM8994_FLL_SRC_MCLK(id); + else if (id == HIFI_DAI) + mclk_id = WM8994_FLL_SRC_MCLK2; + else + mclk_id = WM8994_FLL_SRC_MCLK1; + + if (id == HIFI_DAI) { + /* + * Set GPIO1 pin function to reserve, so that DAC1 and ADC1 + * using shared LRCLK from DACLRCK1. + */ + snd_soc_update_bits(codec, WM8994_GPIO_1, 0x1f, 0x2); + + if (!data->is_hifi_dai_master) + hifi_dai_sysclk_dir = SND_SOC_CLOCK_IN; + + ret = snd_soc_dai_set_sysclk(cpu_dai, 0, 0, !hifi_dai_sysclk_dir); + + if (ret) { + dev_err(dev, "failed to set cpu sysclk: %d\n", ret); + return ret; + } + + if (!data->is_hifi_dai_master) { + ret = snd_soc_dai_set_sysclk(codec_dai, mclk_id, + data->mclk_freq[mclk_id - 1], + hifi_dai_sysclk_dir); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; + } + } else if (id == VOICE_DAI) { + /* + * Set GPIO6 pin function to reserve, so that DAC2 and ADC2 + * using shared LRCLK from DACLRCK2. + */ + snd_soc_update_bits(codec, WM8994_GPIO_6, 0x1f, 0x2); + } + + if (params_width(params) == 24) + pll_out = params_rate(params) * 384; + else + pll_out = params_rate(params) * 256; + + ret = snd_soc_dai_set_pll(codec_dai, + WM8994_FLL(id), + mclk_id, + data->mclk_freq[mclk_id - 1], + pll_out); + if (ret) { + dev_err(dev, "failed to set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, + WM8994_SYSCLK_FLL(id), + pll_out, + SND_SOC_CLOCK_OUT); + if (ret) { + dev_err(dev, "failed to set codec sysclk: %d\n", ret); + return ret; + } + + return 0; +} + +static int imx_wm8958_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 snd_soc_card *card = rtd->card; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int id = codec_dai->id - 1; + + data->is_stream_in_use[id][tx] = false; + + if (id == HIFI_DAI && !data->is_hifi_dai_master) + return 0; + + if (!data->is_stream_in_use[id][!tx]) { + /* + * We should connect AIFxCLK source to FLL after enable FLL, and + * disconnet AIF1CLK source to FLL before disable FLL, otherwise + * FLL worked abnormal. + */ + snd_soc_dai_set_sysclk(codec_dai, WM8994_FLL_SRC_MCLK(id), + data->mclk_freq[id], SND_SOC_CLOCK_OUT); + + /* Disable FLL1 after all stream finished. */ + snd_soc_dai_set_pll(codec_dai, WM8994_FLL(id), 0, 0, 0); + } + + return 0; +} + +static int imx_wm8958_set_bias_level(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + int ret, i; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_STANDBY: + if (card->dapm.bias_level == SND_SOC_BIAS_OFF) { + /* need to enable mclk to write/read wm8958 register */ + for (i = 0; i < WM8958_MCLK_MAX; i++) { + if (!IS_ERR(data->mclk[i])) { + ret = clk_prepare_enable(data->mclk[i]); + if (ret) + dev_warn(card->dev, "Failed to enable MCLK%d: %d\n", i + 1, ret); + } + } + } + break; + default: + break; + } + + return 0; +} + +static int imx_wm8958_set_bias_level_post(struct snd_soc_card *card, + struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level) +{ + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *codec_dai; + struct imx_wm8958_data *data = snd_soc_card_get_drvdata(card); + int i; + + rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); + codec_dai = rtd->codec_dai; + + if (dapm->dev != codec_dai->dev) + return 0; + + switch (level) { + case SND_SOC_BIAS_OFF: + if (card->dapm.bias_level == SND_SOC_BIAS_STANDBY) + for (i = 0; i < WM8958_MCLK_MAX; i++) { + if (!IS_ERR(data->mclk[i])) + clk_disable_unprepare(data->mclk[i]); + } + break; + default: + break; + } + + card->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_ops imx_hifi_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_ops imx_voice_ops = { + .hw_params = imx_wm8958_hw_params, + .hw_free = imx_wm8958_hw_free, +}; + +static struct snd_soc_dai_link imx_wm8958_dai_link[] = { + [HIFI_DAI] = { + .name = "HiFi", + .stream_name = "HiFi", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif1", + .ops = &imx_hifi_ops, + }, + [VOICE_DAI] = { + .name = "Voice", + .stream_name = "Voice", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif2", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + .ops = &imx_voice_ops, + }, + [BT_DAI] = { + .name = "Bluetooth", + .stream_name = "Bluetooth", + .cpu_dai_name = "snd-soc-dummy-dai", + .codec_name = "wm8994-codec", + .codec_dai_name = "wm8994-aif3", + .platform_name = "snd-soc-dummy", + .ignore_pmdown_time = 1, + }, +}; + +static int imx_wm8958_probe(struct platform_device *pdev) +{ + struct device_node *cpu_np, *codec_np; + struct device_node *np = pdev->dev.of_node; + struct platform_device *cpu_pdev; + struct i2c_client *codec_dev; + struct imx_wm8958_data *data; + char tmp[8]; + u32 dai_fmt; + int ret, i; + + cpu_np = of_parse_phandle(np, "cpu-dai", 0); + if (!cpu_np) { + dev_err(&pdev->dev, "cpu dai phandle missing or invalid\n"); + return -EINVAL; + } + + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (!codec_np) { + dev_err(&pdev->dev, "phandle missing or invalid\n"); + ret = -EINVAL; + goto fail; + } + + cpu_pdev = of_find_device_by_node(cpu_np); + if (!cpu_pdev) { + dev_err(&pdev->dev, "failed to find cpu dai platform device\n"); + ret = -EINVAL; + goto fail; + } + + codec_dev = of_find_i2c_device_by_node(codec_np); + if (!codec_dev || !codec_dev->dev.driver) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto fail; + } + + data = devm_kzalloc(&pdev->dev, sizeof(*data), GFP_KERNEL); + if (!data) + return -ENOMEM; + + data->dai_link = imx_wm8958_dai_link; + + /* + * AIF1 support codec master and slave mode + * AIF2 and AIF3 just support codec master mode + */ + for (i = 0; i < DAI_LINK_NUM; i++) { + sprintf(tmp, "aif%d-", i + 1); + dai_fmt = snd_soc_of_parse_daifmt(np, tmp, NULL, NULL); + + if (!(dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK)) { + dev_warn(&pdev->dev, + "failed to get format, default to i2s mode\n"); + dai_fmt |= SND_SOC_DAIFMT_I2S; + } + + if (i == HIFI_DAI) { + switch (dai_fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + data->is_hifi_dai_master = false; + break; + default: + data->is_hifi_dai_master = true; + break; + } + } else { + dai_fmt &= ~SND_SOC_DAIFMT_MASTER_MASK; + dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } + data->dai_link[i].dai_fmt = dai_fmt; + } + + for (i = 0; i < WM8958_MCLK_MAX; i++) { + sprintf(tmp, "mclk%d", i + 1); + data->mclk[i] = devm_clk_get(&codec_dev->dev, tmp); + if (IS_ERR(data->mclk[i])) { + ret = PTR_ERR(data->mclk[i]); + dev_err(&pdev->dev, "failed to get mclk%d clock: %d\n", + i + 1, ret); + } else { + data->mclk_freq[i] = clk_get_rate(data->mclk[i]); + } + } + + data->dai_link[HIFI_DAI].cpu_dai_name = dev_name(&cpu_pdev->dev); + data->dai_link[HIFI_DAI].platform_of_node = cpu_np; + + data->card.set_bias_level = imx_wm8958_set_bias_level; + data->card.set_bias_level_post = imx_wm8958_set_bias_level_post; + data->card.dev = &pdev->dev; + ret = snd_soc_of_parse_card_name(&data->card, "model"); + if (ret) + goto fail; + + data->card.num_links = DAI_LINK_NUM; + data->card.dai_link = data->dai_link; + data->card.dapm_widgets = imx_wm8958_dapm_widgets; + data->card.num_dapm_widgets = ARRAY_SIZE(imx_wm8958_dapm_widgets); + + ret = snd_soc_of_parse_audio_routing(&data->card, "audio-routing"); + if (ret) + goto fail; + + platform_set_drvdata(pdev, &data->card); + snd_soc_card_set_drvdata(&data->card, data); + + ret = devm_snd_soc_register_card(&pdev->dev, &data->card); + if (ret) { + dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); + goto fail; + } + +fail: + of_node_put(cpu_np); + of_node_put(codec_np); + + return ret; +} + +static const struct of_device_id imx_wm8958_dt_ids[] = { + { .compatible = "fsl,imx-audio-wm8958", }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, imx_wm8958_dt_ids); + +static struct platform_driver imx_wm8958_driver = { + .driver = { + .name = "imx-wm8958", + .pm = &snd_soc_pm_ops, + .of_match_table = imx_wm8958_dt_ids, + }, + .probe = imx_wm8958_probe, +}; +module_platform_driver(imx_wm8958_driver); + +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_DESCRIPTION("Freescale i.MX WM8958 ASoC machine driver"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:imx-wm8958"); -- 1.9.1

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[alsa-devel] [PATCH 1/4] ALSA: core: use dev_to_snd_card
by Geliang Tang 04 Jan '16

04 Jan '16

Use dev_to_snd_card() instead of open-coding it. Signed-off-by: Geliang Tang <geliangtang(a)163.com> --- sound/core/init.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/init.c b/sound/core/init.c index 20f37fb..fcb84e1 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -639,7 +639,7 @@ static ssize_t card_id_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_card *card = container_of(dev, struct snd_card, card_dev); + struct snd_card *card = dev_to_snd_card(dev); return snprintf(buf, PAGE_SIZE, "%s\n", card->id); } @@ -647,7 +647,7 @@ static ssize_t card_id_store_attr(struct device *dev, struct device_attribute *attr, const char *buf, size_t count) { - struct snd_card *card = container_of(dev, struct snd_card, card_dev); + struct snd_card *card = dev_to_snd_card(dev); char buf1[sizeof(card->id)]; size_t copy = count > sizeof(card->id) - 1 ? sizeof(card->id) - 1 : count; @@ -679,7 +679,7 @@ static ssize_t card_number_show_attr(struct device *dev, struct device_attribute *attr, char *buf) { - struct snd_card *card = container_of(dev, struct snd_card, card_dev); + struct snd_card *card = dev_to_snd_card(dev); return snprintf(buf, PAGE_SIZE, "%i\n", card->number); } -- 2.5.0

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