Commit f5a246ea authored by Linus Torvalds's avatar Linus Torvalds
Browse files

Merge tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound

Pull sound updates from Takashi Iwai:
 "This contains pretty many small commits covering fairly large range of
  files in sound/ directory.  Partly because of additional API support
  and partly because of constantly developed ASoC and ARM stuff.

  Some highlights:

   - Introduced the helper function and documentation for exposing the
     channel map via control API, as discussed in Plumbers; most of PCI
     drivers are covered, will follow more drivers later

   - Most of drivers have been replaced with the new PM callbacks (if
     the bus is supported)

   - HD-audio controller got the support of runtime PM and the support
     of D3 clock-stop.  Also changing the power_save option in sysfs
     kicks off immediately to enable / disable the power-save mode.

   - Another significant code change in HD-audio is the rewrite of
     firmware loading code.  Other than that, most of changes in
     HD-audio are continued cleanups and standardization for the generic
     auto parser and bug fixes (HBR, device-specific fixups), in
     addition to the support of channel-map API.

   - Addition of ASoC bindings for the compressed API, used by the
     mid-x86 drivers.

   - Lots of cleanups and API refreshes for ASoC codec drivers and
     DaVinci.

   - Conversion of OMAP to dmaengine.

   - New machine driver for Wolfson Microelectronics Bells.

   - New CODEC driver for Wolfson Microelectronics WM0010.

   - Enhancements to the ux500 and wm2000 drivers

   - A new driver for DA9055 and the support for regulator bypass mode."

Fix up various arm soc header file reorg conflicts.

* tag 'sound-3.7' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (339 commits)
  ALSA: hda - Add new codec ALC283 ALC290 support
  ALSA: hda - avoid unneccesary indices on "Headphone Jack" controls
  ALSA: hda - fix indices on boost volume on Conexant
  ALSA: aloop - add locking to timer access
  ALSA: hda - Fix hang caused by race during suspend.
  sound: Remove unnecessary semicolon
  ALSA: hda/realtek - Fix detection of ALC271X codec
  ALSA: hda - Add inverted internal mic quirk for Lenovo IdeaPad U310
  ALSA: hda - make Realtek/Sigmatel/Conexant use the generic unsol event
  ALSA: hda - make a generic unsol event handler
  ASoC: codecs: Add DA9055 codec driver
  ASoC: eukrea-tlv320: Convert it to platform driver
  ALSA: ASoC: add DT bindings for CS4271
  ASoC: wm_hubs: Ensure volume updates are handled during class W startup
  ASoC: wm5110: Adding missing volume update bits
  ASoC: wm5110: Add OUT3R support
  ASoC: wm5110: Add AEC loopback support
  ASoC: wm5110: Rename EPOUT to HPOUT3
  ASoC: arizona: Add more clock rates
  ASoC: arizona: Add more DSP options for mixer input muxes
  ...
parents d5bbd43d 7ff34ad8
CS4270 audio CODEC
The driver for this device currently only supports I2C.
Required properties:
- compatible : "cirrus,cs4270"
- reg : the I2C address of the device for I2C
Optional properties:
- reset-gpio : a GPIO spec for the reset pin. If specified, it will be
deasserted before communication to the codec starts.
Example:
codec: cs4270@48 {
compatible = "cirrus,cs4270";
reg = <0x48>;
};
Cirrus Logic CS4271 DT bindings
This driver supports both the I2C and the SPI bus.
Required properties:
- compatible: "cirrus,cs4271"
For required properties on SPI, please consult
Documentation/devicetree/bindings/spi/spi-bus.txt
Required properties on I2C:
- reg: the i2c address
Optional properties:
- reset-gpio: a GPIO spec to define which pin is connected to the chip's
!RESET pin
Examples:
codec_i2c: cs4271@10 {
compatible = "cirrus,cs4271";
reg = <0x10>;
reset-gpio = <&gpio 23 0>;
};
codec_spi: cs4271@0 {
compatible = "cirrus,cs4271";
reg = <0x0>;
reset-gpio = <&gpio 23 0>;
spi-max-frequency = <6000000>;
};
Texas Instruments McASP controller
Required properties:
- compatible :
"ti,dm646x-mcasp-audio" : for DM646x platforms
"ti,da830-mcasp-audio" : for both DA830 & DA850 platforms
"ti,omap2-mcasp-audio" : for OMAP2 platforms (TI81xx, AM33xx)
- reg : Should contain McASP registers offset and length
- interrupts : Interrupt number for McASP
- op-mode : I2S/DIT ops mode.
- tdm-slots : Slots for TDM operation.
- num-serializer : Serializers used by McASP.
- serial-dir : A list of serializer pin mode. The list number should be equal
to "num-serializer" parameter. Each entry is a number indication
serializer pin direction. (0 - INACTIVE, 1 - TX, 2 - RX)
Optional properties:
- ti,hwmods : Must be "mcasp<n>", n is controller instance starting 0
- tx-num-evt : FIFO levels.
- rx-num-evt : FIFO levels.
- sram-size-playback : size of sram to be allocated during playback
- sram-size-capture : size of sram to be allocated during capture
Example:
mcasp0: mcasp0@1d00000 {
compatible = "ti,da830-mcasp-audio";
#address-cells = <1>;
#size-cells = <0>;
reg = <0x100000 0x3000>;
interrupts = <82 83>;
op-mode = <0>; /* MCASP_IIS_MODE */
tdm-slots = <2>;
num-serializer = <16>;
serial-dir = <
0 0 0 0 /* 0: INACTIVE, 1: TX, 2: RX */
0 0 0 0
0 0 0 1
2 0 0 0 >;
tx-num-evt = <1>;
rx-num-evt = <1>;
};
* Texas Instruments OMAP4+ and twl6040 based audio setups
Required properties:
- compatible: "ti,abe-twl6040"
- ti,model: Name of the sound card ( for example "SDP4430")
- ti,mclk-freq: MCLK frequency for HPPLL operation
- ti,mcpdm: phandle for the McPDM node
- ti,twl6040: phandle for the twl6040 core node
- ti,audio-routing: List of 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.
Optional properties:
- ti,dmic: phandle for the OMAP dmic node if the machine have it connected
- ti,jack_detection: Need to be set to <1> if the board capable to detect jack
insertion, removal.
Available audio endpoints for the audio-routing table:
Board connectors:
* Headset Stereophone
* Earphone Spk
* Ext Spk
* Line Out
* Vibrator
* Headset Mic
* Main Handset Mic
* Sub Handset Mic
* Line In
* Digital Mic
twl6040 pins:
* HSOL
* HSOR
* EP
* HFL
* HFR
* AUXL
* AUXR
* VIBRAL
* VIBRAR
* HSMIC
* MAINMIC
* SUBMIC
* AFML
* AFMR
* Headset Mic Bias
* Main Mic Bias
* Digital Mic1 Bias
* Digital Mic2 Bias
Digital mic pins:
* DMic
Example:
sound {
compatible = "ti,abe-twl6040";
ti,model = "SDP4430";
ti,jack-detection = <1>;
ti,mclk-freq = <38400000>;
ti,mcpdm = <&mcpdm>;
ti,dmic = <&dmic>;
ti,twl6040 = <&twl6040>;
/* Audio routing */
ti,audio-routing =
"Headset Stereophone", "HSOL",
"Headset Stereophone", "HSOR",
"Earphone Spk", "EP",
"Ext Spk", "HFL",
"Ext Spk", "HFR",
"Line Out", "AUXL",
"Line Out", "AUXR",
"Vibrator", "VIBRAL",
"Vibrator", "VIBRAR",
"HSMIC", "Headset Mic",
"Headset Mic", "Headset Mic Bias",
"MAINMIC", "Main Handset Mic",
"Main Handset Mic", "Main Mic Bias",
"SUBMIC", "Sub Handset Mic",
"Sub Handset Mic", "Main Mic Bias",
"AFML", "Line In",
"AFMR", "Line In",
"DMic", "Digital Mic",
"Digital Mic", "Digital Mic1 Bias";
};
* Texas Instruments OMAP2+ McBSP module
Required properties:
- compatible: "ti,omap2420-mcbsp" for McBSP on OMAP2420
"ti,omap2430-mcbsp" for McBSP on OMAP2430
"ti,omap3-mcbsp" for McBSP on OMAP3
"ti,omap4-mcbsp" for McBSP on OMAP4 and newer SoC
- reg: Register location and size, for OMAP4+ as an array:
<MPU access base address, size>,
<L3 interconnect address, size>;
- reg-names: Array of strings associated with the address space
- interrupts: Interrupt numbers for the McBSP port, as an array in case the
McBSP IP have more interrupt lines:
<OCP compliant irq>,
<TX irq>,
<RX irq>;
- interrupt-names: Array of strings associated with the interrupt numbers
- interrupt-parent: The parent interrupt controller
- ti,buffer-size: Size of the FIFO on the port (OMAP2430 and newer SoC)
- ti,hwmods: Name of the hwmod associated to the McBSP port
Example:
mcbsp2: mcbsp@49022000 {
compatible = "ti,omap3-mcbsp";
reg = <0x49022000 0xff>,
<0x49028000 0xff>;
reg-names = "mpu", "sidetone";
interrupts = <0 17 0x4>, /* OCP compliant interrupt */
<0 62 0x4>, /* TX interrupt */
<0 63 0x4>, /* RX interrupt */
<0 4 0x4>; /* Sidetone */
interrupt-names = "common", "tx", "rx", "sidetone";
interrupt-parent = <&intc>;
ti,buffer-size = <1280>;
ti,hwmods = "mcbsp2";
};
* Texas Instruments SoC with twl4030 based audio setups
Required properties:
- compatible: "ti,omap-twl4030"
- ti,model: Name of the sound card (for example "omap3beagle")
- ti,mcbsp: phandle for the McBSP node
- ti,codec: phandle for the twl4030 audio node
Example:
sound {
compatible = "ti,omap-twl4030";
ti,model = "omap3beagle";
ti,mcbsp = <&mcbsp2>;
ti,codec = <&twl_audio>;
};
Texas Instruments - tlv320aic3x Codec module
The tlv320aic3x serial control bus communicates through I2C protocols
Required properties:
- compatible - "string" - "ti,tlv320aic3x"
- reg - <int> - I2C slave address
Optional properties:
- gpio-reset - gpio pin number used for codec reset
- ai3x-gpio-func - <array of 2 int> - AIC3X_GPIO1 & AIC3X_GPIO2 Functionality
Example:
tlv320aic3x: tlv320aic3x@1b {
compatible = "ti,tlv320aic3x";
reg = <0x1b>;
};
......@@ -860,8 +860,14 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
[Multiple options for each card instance]
model - force the model name
position_fix - Fix DMA pointer (0 = auto, 1 = use LPIB, 2 = POSBUF,
3 = VIACOMBO, 4 = COMBO)
position_fix - Fix DMA pointer
-1 = system default: choose appropriate one per controller
hardware
0 = auto: falls back to LPIB when POSBUF doesn't work
1 = use LPIB
2 = POSBUF: use position buffer
3 = VIACOMBO: VIA-specific workaround for capture
4 = COMBO: use LPIB for playback, auto for capture stream
probe_mask - Bitmask to probe codecs (default = -1, meaning all slots)
When the bit 8 (0x100) is set, the lower 8 bits are used
as the "fixed" codec slots; i.e. the driver probes the
......
ALSA PCM channel-mapping API
============================
Takashi Iwai <tiwai@suse.de>
GENERAL
-------
The channel mapping API allows user to query the possible channel maps
and the current channel map, also optionally to modify the channel map
of the current stream.
A channel map is an array of position for each PCM channel.
Typically, a stereo PCM stream has a channel map of
{ front_left, front_right }
while a 4.0 surround PCM stream has a channel map of
{ front left, front right, rear left, rear right }.
The problem, so far, was that we had no standard channel map
explicitly, and applications had no way to know which channel
corresponds to which (speaker) position. Thus, applications applied
wrong channels for 5.1 outputs, and you hear suddenly strange sound
from rear. Or, some devices secretly assume that center/LFE is the
third/fourth channels while others that C/LFE as 5th/6th channels.
Also, some devices such as HDMI are configurable for different speaker
positions even with the same number of total channels. However, there
was no way to specify this because of lack of channel map
specification. These are the main motivations for the new channel
mapping API.
DESIGN
------
Actually, "the channel mapping API" doesn't introduce anything new in
the kernel/user-space ABI perspective. It uses only the existing
control element features.
As a ground design, each PCM substream may contain a control element
providing the channel mapping information and configuration. This
element is specified by:
iface = SNDRV_CTL_ELEM_IFACE_PCM
name = "Playback Channel Map" or "Capture Channel Map"
device = the same device number for the assigned PCM substream
index = the same index number for the assigned PCM substream
Note the name is different depending on the PCM substream direction.
Each control element provides at least the TLV read operation and the
read operation. Optionally, the write operation can be provided to
allow user to change the channel map dynamically.
* TLV
The TLV operation gives the list of available channel
maps. A list item of a channel map is usually a TLV of
type data-bytes ch0 ch1 ch2...
where type is the TLV type value, the second argument is the total
bytes (not the numbers) of channel values, and the rest are the
position value for each channel.
As a TLV type, either SNDRV_CTL_TLVT_CHMAP_FIXED,
SNDRV_CTL_TLV_CHMAP_VAR or SNDRV_CTL_TLVT_CHMAP_PAIRED can be used.
The _FIXED type is for a channel map with the fixed channel position
while the latter two are for flexible channel positions. _VAR type is
for a channel map where all channels are freely swappable and _PAIRED
type is where pair-wise channels are swappable. For example, when you
have {FL/FR/RL/RR} channel map, _PAIRED type would allow you to swap
only {RL/RR/FL/FR} while _VAR type would allow even swapping FL and
RR.
These new TLV types are defined in sound/tlv.h.
The available channel position values are defined in sound/asound.h,
here is a cut:
/* channel positions */
enum {
SNDRV_CHMAP_UNKNOWN = 0,
SNDRV_CHMAP_NA, /* N/A, silent */
SNDRV_CHMAP_MONO, /* mono stream */
/* this follows the alsa-lib mixer channel value + 3 */
SNDRV_CHMAP_FL, /* front left */
SNDRV_CHMAP_FR, /* front right */
SNDRV_CHMAP_RL, /* rear left */
SNDRV_CHMAP_RR, /* rear right */
SNDRV_CHMAP_FC, /* front center */
SNDRV_CHMAP_LFE, /* LFE */
SNDRV_CHMAP_SL, /* side left */
SNDRV_CHMAP_SR, /* side right */
SNDRV_CHMAP_RC, /* rear center */
/* new definitions */
SNDRV_CHMAP_FLC, /* front left center */
SNDRV_CHMAP_FRC, /* front right center */
SNDRV_CHMAP_RLC, /* rear left center */
SNDRV_CHMAP_RRC, /* rear right center */
SNDRV_CHMAP_FLW, /* front left wide */
SNDRV_CHMAP_FRW, /* front right wide */
SNDRV_CHMAP_FLH, /* front left high */
SNDRV_CHMAP_FCH, /* front center high */
SNDRV_CHMAP_FRH, /* front right high */
SNDRV_CHMAP_TC, /* top center */
SNDRV_CHMAP_TFL, /* top front left */
SNDRV_CHMAP_TFR, /* top front right */
SNDRV_CHMAP_TFC, /* top front center */
SNDRV_CHMAP_TRL, /* top rear left */
SNDRV_CHMAP_TRR, /* top rear right */
SNDRV_CHMAP_TRC, /* top rear center */
SNDRV_CHMAP_LAST = SNDRV_CHMAP_TRC,
};
When a PCM stream can provide more than one channel map, you can
provide multiple channel maps in a TLV container type. The TLV data
to be returned will contain such as:
SNDRV_CTL_TLVT_CONTAINER 96
SNDRV_CTL_TLVT_CHMAP_FIXED 4 SNDRV_CHMAP_FC
SNDRV_CTL_TLVT_CHMAP_FIXED 8 SNDRV_CHMAP_FL SNDRV_CHMAP_FR
SNDRV_CTL_TLVT_CHMAP_FIXED 16 NDRV_CHMAP_FL SNDRV_CHMAP_FR \
SNDRV_CHMAP_RL SNDRV_CHMAP_RR
The channel position is provided in LSB 16bits. The upper bits are
used for bit flags.
#define SNDRV_CHMAP_POSITION_MASK 0xffff
#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16)
#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16)
SNDRV_CHMAP_PHASE_INVERSE indicates the channel is phase inverted,
(thus summing left and right channels would result in almost silence).
Some digital mic devices have this.
When SNDRV_CHMAP_DRIVER_SPEC is set, all the channel position values
don't follow the standard definition above but driver-specific.
* READ OPERATION
The control read operation is for providing the current channel map of
the given stream. The control element returns an integer array
containing the position of each channel.
When this is performed before the number of the channel is specified
(i.e. hw_params is set), it should return all channels set to
UNKNOWN.
* WRITE OPERATION
The control write operation is optional, and only for devices that can
change the channel configuration on the fly, such as HDMI. User needs
to pass an integer value containing the valid channel positions for
all channels of the assigned PCM substream.
This operation is allowed only at PCM PREPARED state. When called in
other states, it shall return an error.
......@@ -74,7 +74,8 @@ CMI9880
AD1882 / AD1882A
================
3stack 3-stack mode (default)
3stack 3-stack mode
3stack-automute 3-stack with automute front HP (default)
6stack 6-stack mode
AD1884A / AD1883 / AD1984A / AD1984B
......
/*
* TI DaVinci Audio definitions
*/
#ifndef __ASM_ARCH_DAVINCI_ASP_H
#define __ASM_ARCH_DAVINCI_ASP_H
/* Bases of dm644x and dm355 register banks */
#define DAVINCI_ASP0_BASE 0x01E02000
#define DAVINCI_ASP1_BASE 0x01E04000
/* Bases of dm365 register banks */
#define DAVINCI_DM365_ASP0_BASE 0x01D02000
/* Bases of dm646x register banks */
#define DAVINCI_DM646X_MCASP0_REG_BASE 0x01D01000
#define DAVINCI_DM646X_MCASP1_REG_BASE 0x01D01800
/* Bases of da850/da830 McASP0 register banks */
#define DAVINCI_DA8XX_MCASP0_REG_BASE 0x01D00000
/* Bases of da830 McASP1 register banks */
#define DAVINCI_DA830_MCASP1_REG_BASE 0x01D04000
/* EDMA channels of dm644x and dm355 */
#define DAVINCI_DMA_ASP0_TX 2
#define DAVINCI_DMA_ASP0_RX 3
#define DAVINCI_DMA_ASP1_TX 8
#define DAVINCI_DMA_ASP1_RX 9
/* EDMA channels of dm646x */
#define DAVINCI_DM646X_DMA_MCASP0_AXEVT0 6
#define DAVINCI_DM646X_DMA_MCASP0_AREVT0 9
#define DAVINCI_DM646X_DMA_MCASP1_AXEVT1 12
/* EDMA channels of da850/da830 McASP0 */
#define DAVINCI_DA8XX_DMA_MCASP0_AREVT 0
#define DAVINCI_DA8XX_DMA_MCASP0_AXEVT 1
/* EDMA channels of da830 McASP1 */
#define DAVINCI_DA830_DMA_MCASP1_AREVT 2
#define DAVINCI_DA830_DMA_MCASP1_AXEVT 3
/* Interrupts */
#define DAVINCI_ASP0_RX_INT IRQ_MBRINT
#define DAVINCI_ASP0_TX_INT IRQ_MBXINT
#define DAVINCI_ASP1_RX_INT IRQ_MBRINT
#define DAVINCI_ASP1_TX_INT IRQ_MBXINT
#endif /* __ASM_ARCH_DAVINCI_ASP_H */
......@@ -22,10 +22,10 @@
#include <linux/davinci_emac.h>
#include <linux/platform_device.h>
#include <linux/spi/spi.h>
#include <mach/asp.h>
#include <linux/platform_data/davinci_asp.h>
#include <linux/platform_data/keyscan-davinci.h>
#include <mach/hardware.h>
#include <mach/edma.h>
#include <media/davinci/vpfe_capture.h>
#include <media/davinci/vpif_types.h>
......
......@@ -24,6 +24,7 @@
#include <mach/cpuidle.h>
#include "clock.h"
#include "asp.h"
#define DA8XX_TPCC_BASE 0x01c00000
#define DA8XX_TPTC0_BASE 0x01c08000
......@@ -505,15 +506,8 @@ static struct platform_device da850_mcasp_device = {
.resource = da850_mcasp_resources,
};
static struct platform_device davinci_pcm_device = {
.name = "davinci-pcm-audio",
.id = -1,
};
void __init da8xx_register_mcasp(int id, struct snd_platform_data *pdata)
{
platform_device_register(&davinci_pcm_device);
/* DA830/OMAP-L137 has 3 instances of McASP */
if (cpu_is_davinci_da830() && id == 1) {
da830_mcasp1_device.dev.platform_data = pdata;
......
......@@ -313,16 +313,6 @@ static void davinci_init_wdt(void)
/*-------------------------------------------------------------------------*/
static struct platform_device davinci_pcm_device = {
.name = "davinci-pcm-audio",
.id = -1,
};
static void davinci_init_pcm(void)
{
platform_device_register(&davinci_pcm_device);
}
/*-------------------------------------------------------------------------*/
struct davinci_timer_instance davinci_timer_instance[2] = {
......@@ -345,7 +335,6 @@ static int __init davinci_init_devices(void)
/* please keep these calls, and their implementations above,
* in alphabetical order so they're easier to sort through.
*/
davinci_init_pcm();
davinci_init_wdt();
return 0;
......
......@@ -26,13 +26,13 @@
#include <mach/time.h>
#include <mach/serial.h>
#include <mach/common.h>
#include <mach/asp.h>
#include <linux/platform_data/spi-davinci.h>
#include <mach/gpio-davinci.h>
#include "davinci.h"
#include "clock.h"
#include "mux.h"
#include "asp.h"
#define DM355_UART2_BASE (IO_PHYS + 0x206000)
......
......@@ -29,7 +29,6 @@
#include <mach/time.h>
#include <mach/serial.h>
#include <mach/common.h>
#include <mach/asp.h>
#include <linux/platform_data/keyscan-davinci.h>
#include <linux/platform_data/spi-davinci.h>
#include <mach/gpio-davinci.h>
......@@ -37,6 +36,7 @@
#include "davinci.h"
#include "clock.h"
#include "mux.h"
#include "asp.h"
#define DM365_REF_FREQ 24000000 /* 24 MHz on the DM365 EVM */
......
......@@ -23,12 +23,12 @@
#include <mach/time.h>
#include <mach/serial.h>
#include <mach/common.h>
#include <mach/asp.h>
#include <mach/gpio-davinci.h>
#include "davinci.h"
#include "clock.h"
#include "mux.h"
#include "asp.h"
/*
* Device specific clocks
......