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Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (526 commits)
ASoC: twl6040 - Add method to query optimum PDM_DL1 gain
ALSA: hda - Fix the lost power-setup of seconary pins after PM resume
ALSA: usb-audio: add Yamaha MOX6/MOX8 support
ALSA: virtuoso: add S/PDIF input support for all Xonars
ALSA: ice1724 - Support for ooAoo SQ210a
ALSA: ice1724 - Allow card info based on model only
ALSA: ice1724 - Create capture pcm only for ADC-enabled configurations
ALSA: hdspm - Provide unique driver id based on card serial
ASoC: Dynamically allocate the rtd device for a non-empty release()
ASoC: Fix recursive dependency due to select ATMEL_SSC in SND_ATMEL_SOC_SSC
ALSA: hda - Fix the detection of "Loopback Mixing" control for VIA codecs
ALSA: hda - Return the error from get_wcaps_type() for invalid NIDs
ALSA: hda - Use auto-parser for HP laptops with cx20459 codec
ALSA: asihpi - Fix potential Oops in snd_asihpi_cmode_info()
ALSA: hdsp - Fix potential Oops in snd_hdsp_info_pref_sync_ref()
ALSA: hda/cirrus - support for iMac12,2 model
ASoC: cx20442: add bias control over a platform provided regulator
ALSA: usb-audio - Avoid flood of frame-active debug messages
ALSA: snd-usb-us122l: Delete calls to preempt_disable
mfd: Put WM8994 into cache only mode when suspending
...
Fix up trivial conflicts in:
- arch/arm/mach-s3c64xx/mach-crag6410.c:
renamed speyside_wm8962 to tobermory, added littlemill right
next to it
- drivers/base/regmap/{regcache.c,regmap.c}:
duplicate diff that had already come in with other changes in
the regmap tree
This commit is contained in:
Linus Torvalds
@@ -404,7 +404,7 @@
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||||
/* SNDRV_CARDS: maximum number of cards supported by this module */
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static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
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static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
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static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
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static bool enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
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/* definition of the chip-specific record */
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struct mychip {
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@@ -0,0 +1,71 @@
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NVIDIA Tegra audio complex
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Required properties:
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- compatible : "nvidia,tegra-audio-wm8903"
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- nvidia,model : The user-visible name of this sound complex.
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- nvidia,audio-routing : A list of the connections between audio components.
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Each entry is a pair of strings, the first being the connection's sink,
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the second being the connection's source. Valid names for sources and
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sinks are the WM8903's pins, and the jacks on the board:
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WM8903 pins:
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* IN1L
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* IN1R
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* IN2L
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* IN2R
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* IN3L
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* IN3R
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* DMICDAT
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* HPOUTL
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* HPOUTR
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* LINEOUTL
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* LINEOUTR
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* LOP
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* LON
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* ROP
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* RON
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* MICBIAS
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Board connectors:
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* Headphone Jack
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* Int Spk
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* Mic Jack
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- nvidia,i2s-controller : The phandle of the Tegra I2S1 controller
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- nvidia,audio-codec : The phandle of the WM8903 audio codec
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Optional properties:
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- nvidia,spkr-en-gpios : The GPIO that enables the speakers
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- nvidia,hp-mute-gpios : The GPIO that mutes the headphones
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- nvidia,hp-det-gpios : The GPIO that detect headphones are plugged in
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- nvidia,int-mic-en-gpios : The GPIO that enables the internal microphone
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- nvidia,ext-mic-en-gpios : The GPIO that enables the external microphone
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Example:
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sound {
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compatible = "nvidia,tegra-audio-wm8903-harmony",
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"nvidia,tegra-audio-wm8903"
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nvidia,model = "tegra-wm8903-harmony";
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nvidia,audio-routing =
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"Headphone Jack", "HPOUTR",
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"Headphone Jack", "HPOUTL",
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"Int Spk", "ROP",
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"Int Spk", "RON",
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"Int Spk", "LOP",
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"Int Spk", "LON",
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"Mic Jack", "MICBIAS",
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"IN1L", "Mic Jack";
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nvidia,i2s-controller = <&i2s1>;
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nvidia,audio-codec = <&wm8903>;
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nvidia,spkr-en-gpios = <&codec 2 0>;
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nvidia,hp-det-gpios = <&gpio 178 0>; /* gpio PW2 */
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nvidia,int-mic-en-gpios = <&gpio 184 0>; /*gpio PX0 */
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nvidia,ext-mic-en-gpios = <&gpio 185 0>; /* gpio PX1 */
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};
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@@ -0,0 +1,12 @@
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NVIDIA Tegra 20 DAS (Digital Audio Switch) controller
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Required properties:
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- compatible : "nvidia,tegra20-das"
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- reg : Should contain DAS registers location and length
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Example:
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das@70000c00 {
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compatible = "nvidia,tegra20-das";
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reg = <0x70000c00 0x80>;
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};
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@@ -0,0 +1,17 @@
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NVIDIA Tegra 20 I2S controller
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Required properties:
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- compatible : "nvidia,tegra20-i2s"
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- reg : Should contain I2S registers location and length
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- interrupts : Should contain I2S interrupt
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- nvidia,dma-request-selector : The Tegra DMA controller's phandle and
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request selector for this I2S controller
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Example:
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i2s@70002800 {
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compatible = "nvidia,tegra20-i2s";
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reg = <0x70002800 0x200>;
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interrupts = < 45 >;
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nvidia,dma-request-selector = < &apbdma 2 >;
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};
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@@ -0,0 +1,50 @@
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WM8903 audio CODEC
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This device supports I2C only.
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Required properties:
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- compatible : "wlf,wm8903"
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- reg : the I2C address of the device.
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- gpio-controller : Indicates this device is a GPIO controller.
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- #gpio-cells : Should be two. The first cell is the pin number and the
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second cell is used to specify optional parameters (currently unused).
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Optional properties:
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- interrupts : The interrupt line the codec is connected to.
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- micdet-cfg : Default register value for R6 (Mic Bias). If absent, the
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default is 0.
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- micdet-delay : The debounce delay for microphone detection in mS. If
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absent, the default is 100.
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- gpio-cfg : A list of GPIO configuration register values. The list must
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be 5 entries long. If absent, no configuration of these registers is
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performed. If any entry has the value 0xffffffff, that GPIO's
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configuration will not be modified.
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Example:
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codec: wm8903@1a {
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compatible = "wlf,wm8903";
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reg = <0x1a>;
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interrupts = < 347 >;
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gpio-controller;
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#gpio-cells = <2>;
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micdet-cfg = <0>;
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micdet-delay = <100>;
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gpio-cfg = <
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0x0600 /* DMIC_LR, output */
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0x0680 /* DMIC_DAT, input */
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0x0000 /* GPIO, output, low */
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0x0200 /* Interrupt, output */
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0x01a0 /* BCLK, input, active high */
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>;
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};
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@@ -0,0 +1,18 @@
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WM1811/WM8994/WM8958 audio CODEC
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These devices support both I2C and SPI (configured with pin strapping
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on the board).
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Required properties:
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- compatible : "wlf,wm1811", "wlf,wm8994", "wlf,wm8958"
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- reg : the I2C address of the device for I2C, the chip select
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number for SPI.
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Example:
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codec: wm8994@1a {
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compatible = "wlf,wm8994";
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reg = <0x1a>;
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};
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@@ -42,4 +42,5 @@ sirf SiRF Technology, Inc.
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st STMicroelectronics
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stericsson ST-Ericsson
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ti Texas Instruments
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wlf Wolfson Microelectronics
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xlnx Xilinx
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@@ -42,19 +42,7 @@ ALC260
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ALC262
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======
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fujitsu Fujitsu Laptop
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benq Benq ED8
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benq-t31 Benq T31
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hippo Hippo (ATI) with jack detection, Sony UX-90s
|
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hippo_1 Hippo (Benq) with jack detection
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toshiba-s06 Toshiba S06
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toshiba-rx1 Toshiba RX1
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tyan Tyan Thunder n6650W (S2915-E)
|
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ultra Samsung Q1 Ultra Vista model
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lenovo-3000 Lenovo 3000 y410
|
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nec NEC Versa S9100
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basic fixed pin assignment w/o SPDIF
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auto auto-config reading BIOS (default)
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N/A
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ALC267/268
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==========
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@@ -350,7 +338,6 @@ STAC92HD83*
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mic-ref Reference board with power management for ports
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dell-s14 Dell laptop
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dell-vostro-3500 Dell Vostro 3500 laptop
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hp HP laptops with (inverted) mute-LED
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hp-dv7-4000 HP dv-7 4000
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auto BIOS setup (default)
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@@ -0,0 +1,188 @@
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compress_offload.txt
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=====================
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Pierre-Louis.Bossart <pierre-louis.bossart@linux.intel.com>
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Vinod Koul <vinod.koul@linux.intel.com>
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Overview
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Since its early days, the ALSA API was defined with PCM support or
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constant bitrates payloads such as IEC61937 in mind. Arguments and
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returned values in frames are the norm, making it a challenge to
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extend the existing API to compressed data streams.
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In recent years, audio digital signal processors (DSP) were integrated
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in system-on-chip designs, and DSPs are also integrated in audio
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codecs. Processing compressed data on such DSPs results in a dramatic
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reduction of power consumption compared to host-based
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processing. Support for such hardware has not been very good in Linux,
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mostly because of a lack of a generic API available in the mainline
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kernel.
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Rather than requiring a compability break with an API change of the
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ALSA PCM interface, a new 'Compressed Data' API is introduced to
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provide a control and data-streaming interface for audio DSPs.
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The design of this API was inspired by the 2-year experience with the
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Intel Moorestown SOC, with many corrections required to upstream the
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API in the mainline kernel instead of the staging tree and make it
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usable by others.
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Requirements
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The main requirements are:
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- separation between byte counts and time. Compressed formats may have
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a header per file, per frame, or no header at all. The payload size
|
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may vary from frame-to-frame. As a result, it is not possible to
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estimate reliably the duration of audio buffers when handling
|
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compressed data. Dedicated mechanisms are required to allow for
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reliable audio-video synchronization, which requires precise
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reporting of the number of samples rendered at any given time.
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- Handling of multiple formats. PCM data only requires a specification
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of the sampling rate, number of channels and bits per sample. In
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contrast, compressed data comes in a variety of formats. Audio DSPs
|
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may also provide support for a limited number of audio encoders and
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decoders embedded in firmware, or may support more choices through
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dynamic download of libraries.
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- Focus on main formats. This API provides support for the most
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popular formats used for audio and video capture and playback. It is
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likely that as audio compression technology advances, new formats
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will be added.
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- Handling of multiple configurations. Even for a given format like
|
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AAC, some implementations may support AAC multichannel but HE-AAC
|
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stereo. Likewise WMA10 level M3 may require too much memory and cpu
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cycles. The new API needs to provide a generic way of listing these
|
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formats.
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- Rendering/Grabbing only. This API does not provide any means of
|
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hardware acceleration, where PCM samples are provided back to
|
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user-space for additional processing. This API focuses instead on
|
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streaming compressed data to a DSP, with the assumption that the
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decoded samples are routed to a physical output or logical back-end.
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- Complexity hiding. Existing user-space multimedia frameworks all
|
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have existing enums/structures for each compressed format. This new
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API assumes the existence of a platform-specific compatibility layer
|
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to expose, translate and make use of the capabilities of the audio
|
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DSP, eg. Android HAL or PulseAudio sinks. By construction, regular
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applications are not supposed to make use of this API.
|
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Design
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The new API shares a number of concepts with with the PCM API for flow
|
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control. Start, pause, resume, drain and stop commands have the same
|
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semantics no matter what the content is.
|
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|
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The concept of memory ring buffer divided in a set of fragments is
|
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borrowed from the ALSA PCM API. However, only sizes in bytes can be
|
||||
specified.
|
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|
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Seeks/trick modes are assumed to be handled by the host.
|
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|
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The notion of rewinds/forwards is not supported. Data committed to the
|
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ring buffer cannot be invalidated, except when dropping all buffers.
|
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|
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The Compressed Data API does not make any assumptions on how the data
|
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is transmitted to the audio DSP. DMA transfers from main memory to an
|
||||
embedded audio cluster or to a SPI interface for external DSPs are
|
||||
possible. As in the ALSA PCM case, a core set of routines is exposed;
|
||||
each driver implementer will have to write support for a set of
|
||||
mandatory routines and possibly make use of optional ones.
|
||||
|
||||
The main additions are
|
||||
|
||||
- get_caps
|
||||
This routine returns the list of audio formats supported. Querying the
|
||||
codecs on a capture stream will return encoders, decoders will be
|
||||
listed for playback streams.
|
||||
|
||||
- get_codec_caps For each codec, this routine returns a list of
|
||||
capabilities. The intent is to make sure all the capabilities
|
||||
correspond to valid settings, and to minimize the risks of
|
||||
configuration failures. For example, for a complex codec such as AAC,
|
||||
the number of channels supported may depend on a specific profile. If
|
||||
the capabilities were exposed with a single descriptor, it may happen
|
||||
that a specific combination of profiles/channels/formats may not be
|
||||
supported. Likewise, embedded DSPs have limited memory and cpu cycles,
|
||||
it is likely that some implementations make the list of capabilities
|
||||
dynamic and dependent on existing workloads. In addition to codec
|
||||
settings, this routine returns the minimum buffer size handled by the
|
||||
implementation. This information can be a function of the DMA buffer
|
||||
sizes, the number of bytes required to synchronize, etc, and can be
|
||||
used by userspace to define how much needs to be written in the ring
|
||||
buffer before playback can start.
|
||||
|
||||
- set_params
|
||||
This routine sets the configuration chosen for a specific codec. The
|
||||
most important field in the parameters is the codec type; in most
|
||||
cases decoders will ignore other fields, while encoders will strictly
|
||||
comply to the settings
|
||||
|
||||
- get_params
|
||||
This routines returns the actual settings used by the DSP. Changes to
|
||||
the settings should remain the exception.
|
||||
|
||||
- get_timestamp
|
||||
The timestamp becomes a multiple field structure. It lists the number
|
||||
of bytes transferred, the number of samples processed and the number
|
||||
of samples rendered/grabbed. All these values can be used to determine
|
||||
the avarage bitrate, figure out if the ring buffer needs to be
|
||||
refilled or the delay due to decoding/encoding/io on the DSP.
|
||||
|
||||
Note that the list of codecs/profiles/modes was derived from the
|
||||
OpenMAX AL specification instead of reinventing the wheel.
|
||||
Modifications include:
|
||||
- Addition of FLAC and IEC formats
|
||||
- Merge of encoder/decoder capabilities
|
||||
- Profiles/modes listed as bitmasks to make descriptors more compact
|
||||
- Addition of set_params for decoders (missing in OpenMAX AL)
|
||||
- Addition of AMR/AMR-WB encoding modes (missing in OpenMAX AL)
|
||||
- Addition of format information for WMA
|
||||
- Addition of encoding options when required (derived from OpenMAX IL)
|
||||
- Addition of rateControlSupported (missing in OpenMAX AL)
|
||||
|
||||
Not supported:
|
||||
|
||||
- Support for VoIP/circuit-switched calls is not the target of this
|
||||
API. Support for dynamic bit-rate changes would require a tight
|
||||
coupling between the DSP and the host stack, limiting power savings.
|
||||
|
||||
- Packet-loss concealment is not supported. This would require an
|
||||
additional interface to let the decoder synthesize data when frames
|
||||
are lost during transmission. This may be added in the future.
|
||||
|
||||
- Volume control/routing is not handled by this API. Devices exposing a
|
||||
compressed data interface will be considered as regular ALSA devices;
|
||||
volume changes and routing information will be provided with regular
|
||||
ALSA kcontrols.
|
||||
|
||||
- Embedded audio effects. Such effects should be enabled in the same
|
||||
manner, no matter if the input was PCM or compressed.
|
||||
|
||||
- multichannel IEC encoding. Unclear if this is required.
|
||||
|
||||
- Encoding/decoding acceleration is not supported as mentioned
|
||||
above. It is possible to route the output of a decoder to a capture
|
||||
stream, or even implement transcoding capabilities. This routing
|
||||
would be enabled with ALSA kcontrols.
|
||||
|
||||
- Audio policy/resource management. This API does not provide any
|
||||
hooks to query the utilization of the audio DSP, nor any premption
|
||||
mechanisms.
|
||||
|
||||
- No notion of underun/overrun. Since the bytes written are compressed
|
||||
in nature and data written/read doesn't translate directly to
|
||||
rendered output in time, this does not deal with underrun/overun and
|
||||
maybe dealt in user-library
|
||||
|
||||
Credits:
|
||||
- Mark Brown and Liam Girdwood for discussions on the need for this API
|
||||
- Harsha Priya for her work on intel_sst compressed API
|
||||
- Rakesh Ughreja for valuable feedback
|
||||
- Sing Nallasellan, Sikkandar Madar and Prasanna Samaga for
|
||||
demonstrating and quantifying the benefits of audio offload on a
|
||||
real platform.
|
||||
@@ -537,6 +537,7 @@ F: sound/soc/codecs/adau*
|
||||
F: sound/soc/codecs/adav*
|
||||
F: sound/soc/codecs/ad1*
|
||||
F: sound/soc/codecs/ssm*
|
||||
F: sound/soc/codecs/sigmadsp.*
|
||||
|
||||
ANALOG DEVICES INC ASOC DRIVERS
|
||||
L: uclinux-dist-devel@blackfin.uclinux.org
|
||||
|
||||
@@ -156,6 +156,9 @@ static int omap_init_mcbsp(struct omap_hwmod *oh, void *unused)
|
||||
else
|
||||
/* The FIFO has 128 locations */
|
||||
pdata->buffer_size = 0x80;
|
||||
} else if (oh->class->rev == MCBSP_CONFIG_TYPE4) {
|
||||
/* The FIFO has 128 locations for all instances */
|
||||
pdata->buffer_size = 0x80;
|
||||
}
|
||||
|
||||
if (oh->class->rev >= MCBSP_CONFIG_TYPE3)
|
||||
|
||||
@@ -435,6 +435,14 @@ static struct platform_device corgiled_device = {
|
||||
},
|
||||
};
|
||||
|
||||
/*
|
||||
* Corgi Audio
|
||||
*/
|
||||
static struct platform_device corgi_audio_device = {
|
||||
.name = "corgi-audio",
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
/*
|
||||
* MMC/SD Device
|
||||
*
|
||||
@@ -641,6 +649,7 @@ static struct platform_device *devices[] __initdata = {
|
||||
&corgifb_device,
|
||||
&corgikbd_device,
|
||||
&corgiled_device,
|
||||
&corgi_audio_device,
|
||||
&sharpsl_nand_device,
|
||||
&sharpsl_rom_device,
|
||||
};
|
||||
|
||||
@@ -528,12 +528,18 @@ static struct platform_device e740_t7l66xb_device = {
|
||||
.resource = eseries_tmio_resources,
|
||||
};
|
||||
|
||||
static struct platform_device e740_audio_device = {
|
||||
.name = "e740-audio",
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
/* ----------------------------------------------------------------------- */
|
||||
|
||||
static struct platform_device *e740_devices[] __initdata = {
|
||||
&e740_fb_device,
|
||||
&e740_t7l66xb_device,
|
||||
&e7xx_gpio_vbus,
|
||||
&e740_audio_device,
|
||||
};
|
||||
|
||||
static void __init e740_init(void)
|
||||
@@ -722,12 +728,18 @@ static struct platform_device e750_tc6393xb_device = {
|
||||
.resource = eseries_tmio_resources,
|
||||
};
|
||||
|
||||
static struct platform_device e750_audio_device = {
|
||||
.name = "e750-audio",
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
/* ------------------------------------------------------------- */
|
||||
|
||||
static struct platform_device *e750_devices[] __initdata = {
|
||||
&e750_fb_device,
|
||||
&e750_tc6393xb_device,
|
||||
&e7xx_gpio_vbus,
|
||||
&e750_audio_device,
|
||||
};
|
||||
|
||||
static void __init e750_init(void)
|
||||
@@ -929,12 +941,18 @@ static struct platform_device e800_tc6393xb_device = {
|
||||
.resource = eseries_tmio_resources,
|
||||
};
|
||||
|
||||
static struct platform_device e800_audio_device = {
|
||||
.name = "e800-audio",
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
/* ----------------------------------------------------------------------- */
|
||||
|
||||
static struct platform_device *e800_devices[] __initdata = {
|
||||
&e800_fb_device,
|
||||
&e800_tc6393xb_device,
|
||||
&e800_gpio_vbus,
|
||||
&e800_audio_device,
|
||||
};
|
||||
|
||||
static void __init e800_init(void)
|
||||
|
||||
@@ -158,6 +158,11 @@ static struct scoop_pcmcia_config poodle_pcmcia_config = {
|
||||
EXPORT_SYMBOL(poodle_scoop_device);
|
||||
|
||||
|
||||
static struct platform_device poodle_audio_device = {
|
||||
.name = "poodle-audio",
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
/* LoCoMo device */
|
||||
static struct resource locomo_resources[] = {
|
||||
[0] = {
|
||||
@@ -407,6 +412,7 @@ static struct platform_device sharpsl_rom_device = {
|
||||
static struct platform_device *devices[] __initdata = {
|
||||
&poodle_locomo_device,
|
||||
&poodle_scoop_device,
|
||||
&poodle_audio_device,
|
||||
&sharpsl_nand_device,
|
||||
&sharpsl_rom_device,
|
||||
};
|
||||
|
||||
@@ -593,10 +593,16 @@ static struct pxa2xx_udc_mach_info imote2_udc_info __initdata = {
|
||||
.udc_command = sg2_udc_command,
|
||||
};
|
||||
|
||||
static struct platform_device imote2_audio_device = {
|
||||
.name = "imote2-audio",
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
static struct platform_device *imote2_devices[] = {
|
||||
&stargate2_flash_device,
|
||||
&imote2_leds,
|
||||
&sht15,
|
||||
&imote2_audio_device,
|
||||
};
|
||||
|
||||
static void __init imote2_init(void)
|
||||
|
||||
@@ -889,6 +889,11 @@ static struct platform_device wm9712_device = {
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
static struct platform_device tosa_audio_device = {
|
||||
.name = "tosa-audio",
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
static struct platform_device *devices[] __initdata = {
|
||||
&tosascoop_device,
|
||||
&tosascoop_jc_device,
|
||||
@@ -901,6 +906,7 @@ static struct platform_device *devices[] __initdata = {
|
||||
&sharpsl_rom_device,
|
||||
&wm9712_device,
|
||||
&tosa_gpio_vbus,
|
||||
&tosa_audio_device,
|
||||
};
|
||||
|
||||
static void tosa_poweroff(void)
|
||||
|
||||
@@ -286,8 +286,8 @@ static struct platform_device lowland_device = {
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
static struct platform_device speyside_wm8962_device = {
|
||||
.name = "speyside-wm8962",
|
||||
static struct platform_device tobermory_device = {
|
||||
.name = "tobermory",
|
||||
.id = -1,
|
||||
};
|
||||
|
||||
@@ -347,7 +347,7 @@ static struct platform_device *crag6410_devices[] __initdata = {
|
||||
&crag6410_lcd_powerdev,
|
||||
&crag6410_backlight_device,
|
||||
&speyside_device,
|
||||
&speyside_wm8962_device,
|
||||
&tobermory_device,
|
||||
&littlemill_device,
|
||||
&lowland_device,
|
||||
&wallvdd_device,
|
||||
|
||||
@@ -762,9 +762,22 @@ static struct platform_device fsi_device = {
|
||||
},
|
||||
};
|
||||
|
||||
static struct platform_device fsi_ak4643_device = {
|
||||
.name = "sh_fsi2_a_ak4643",
|
||||
static struct fsi_ak4642_info fsi2_ak4643_info = {
|
||||
.name = "AK4643",
|
||||
.card = "FSI2A-AK4643",
|
||||
.cpu_dai = "fsia-dai",
|
||||
.codec = "ak4642-codec.0-0013",
|
||||
.platform = "sh_fsi2",
|
||||
.id = FSI_PORT_A,
|
||||
};
|
||||
|
||||
static struct platform_device fsi_ak4643_device = {
|
||||
.name = "fsi-ak4642-audio",
|
||||
.dev = {
|
||||
.platform_data = &fsi_info,
|
||||
},
|
||||
};
|
||||
|
||||
static struct sh_mobile_meram_cfg hdmi_meram_cfg = {
|
||||
.icb[0] = {
|
||||
.marker_icb = 30,
|
||||
|
||||
@@ -990,8 +990,20 @@ static struct platform_device fsi_device = {
|
||||
},
|
||||
};
|
||||
|
||||
static struct fsi_ak4642_info fsi2_ak4643_info = {
|
||||
.name = "AK4643",
|
||||
.card = "FSI2A-AK4643",
|
||||
.cpu_dai = "fsia-dai",
|
||||
.codec = "ak4642-codec.0-0013",
|
||||
.platform = "sh_fsi2",
|
||||
.id = FSI_PORT_A,
|
||||
};
|
||||
|
||||
static struct platform_device fsi_ak4643_device = {
|
||||
.name = "sh_fsi2_a_ak4643",
|
||||
.name = "fsi-ak4642-audio",
|
||||
.dev = {
|
||||
.platform_data = &fsi2_ak4643_info,
|
||||
},
|
||||
};
|
||||
|
||||
/*
|
||||
|
||||
@@ -90,11 +90,11 @@ static struct wm8903_platform_data harmony_wm8903_pdata = {
|
||||
.micdet_delay = 100,
|
||||
.gpio_base = HARMONY_GPIO_WM8903(0),
|
||||
.gpio_cfg = {
|
||||
WM8903_GPIO_NO_CONFIG,
|
||||
WM8903_GPIO_NO_CONFIG,
|
||||
0,
|
||||
WM8903_GPIO_NO_CONFIG,
|
||||
WM8903_GPIO_NO_CONFIG,
|
||||
0,
|
||||
WM8903_GPIO_CONFIG_ZERO,
|
||||
0,
|
||||
0,
|
||||
},
|
||||
};
|
||||
|
||||
|
||||
Some files were not shown because too many files have changed in this diff Show More
Reference in New Issue
Block a user