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authorLinus Torvalds <torvalds@linux-foundation.org>2012-10-09 07:07:14 +0900
committerLinus Torvalds <torvalds@linux-foundation.org>2012-10-09 07:07:14 +0900
commitf5a246eab9a268f51ba8189ea5b098a1bfff200e (patch)
treea6ff7169e0bcaca498d9aec8b0624de1b74eaecb /include
parentd5bbd43d5f450c3fca058f5b85f3dfb4e8cc88c9 (diff)
parent7ff34ad80b7080fafaac8efa9ef0061708eddd51 (diff)
downloadblackbird-op-linux-f5a246eab9a268f51ba8189ea5b098a1bfff200e.tar.gz
blackbird-op-linux-f5a246eab9a268f51ba8189ea5b098a1bfff200e.zip
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 ...
Diffstat (limited to 'include')
-rw-r--r--include/linux/dmaengine.h7
-rw-r--r--include/linux/i2c/twl.h2
-rw-r--r--include/linux/mfd/wm8994/pdata.h4
-rw-r--r--include/linux/platform_data/asoc-mx27vis.h11
-rw-r--r--include/linux/platform_data/asoc-ti-mcbsp.h2
-rw-r--r--include/linux/platform_data/davinci_asp.h105
-rw-r--r--include/linux/platform_data/omap-twl4030.h32
-rw-r--r--include/sound/ac97_codec.h3
-rw-r--r--include/sound/ad1816a.h9
-rw-r--r--include/sound/asound.h39
-rw-r--r--include/sound/compress_driver.h1
-rw-r--r--include/sound/compress_params.h1
-rw-r--r--include/sound/da9055.h33
-rw-r--r--include/sound/emu10k1.h4
-rw-r--r--include/sound/initval.h14
-rw-r--r--include/sound/memalloc.h27
-rw-r--r--include/sound/pcm.h87
-rw-r--r--include/sound/soc-dai.h3
-rw-r--r--include/sound/soc-dapm.h10
-rw-r--r--include/sound/soc.h20
-rw-r--r--include/sound/tegra_wm8903.h26
-rw-r--r--include/sound/tlv.h8
-rw-r--r--include/sound/version.h3
-rw-r--r--include/sound/wm0010.h27
-rw-r--r--include/sound/wm8960.h2
-rw-r--r--include/sound/wm8993.h4
26 files changed, 439 insertions, 45 deletions
diff --git a/include/linux/dmaengine.h b/include/linux/dmaengine.h
index 9c02a4508b25..d3201e438d16 100644
--- a/include/linux/dmaengine.h
+++ b/include/linux/dmaengine.h
@@ -591,7 +591,7 @@ struct dma_device {
struct dma_async_tx_descriptor *(*device_prep_dma_cyclic)(
struct dma_chan *chan, dma_addr_t buf_addr, size_t buf_len,
size_t period_len, enum dma_transfer_direction direction,
- void *context);
+ unsigned long flags, void *context);
struct dma_async_tx_descriptor *(*device_prep_interleaved_dma)(
struct dma_chan *chan, struct dma_interleaved_template *xt,
unsigned long flags);
@@ -653,10 +653,11 @@ static inline struct dma_async_tx_descriptor *dmaengine_prep_rio_sg(
static inline struct dma_async_tx_descriptor *dmaengine_prep_dma_cyclic(
struct dma_chan *chan, dma_addr_t buf_addr, size_t buf_len,
- size_t period_len, enum dma_transfer_direction dir)
+ size_t period_len, enum dma_transfer_direction dir,
+ unsigned long flags)
{
return chan->device->device_prep_dma_cyclic(chan, buf_addr, buf_len,
- period_len, dir, NULL);
+ period_len, dir, flags, NULL);
}
static inline int dmaengine_terminate_all(struct dma_chan *chan)
diff --git a/include/linux/i2c/twl.h b/include/linux/i2c/twl.h
index 1faa58f9b85e..9a5e28462324 100644
--- a/include/linux/i2c/twl.h
+++ b/include/linux/i2c/twl.h
@@ -664,7 +664,7 @@ struct twl4030_codec_data {
unsigned int check_defaults:1;
unsigned int reset_registers:1;
unsigned int hs_extmute:1;
- void (*set_hs_extmute)(int mute);
+ int hs_extmute_gpio;
};
struct twl4030_vibra_data {
diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h
index f0361c031927..fc87be4fdc25 100644
--- a/include/linux/mfd/wm8994/pdata.h
+++ b/include/linux/mfd/wm8994/pdata.h
@@ -164,6 +164,10 @@ struct wm8994_pdata {
int num_micd_rates;
struct wm8958_micd_rate *micd_rates;
+ /* Power up delays to add after microphone bias power up (ms) */
+ int micb1_delay;
+ int micb2_delay;
+
/* LINEOUT can be differential or single ended */
unsigned int lineout1_diff:1;
unsigned int lineout2_diff:1;
diff --git a/include/linux/platform_data/asoc-mx27vis.h b/include/linux/platform_data/asoc-mx27vis.h
new file mode 100644
index 000000000000..409adcd04d04
--- /dev/null
+++ b/include/linux/platform_data/asoc-mx27vis.h
@@ -0,0 +1,11 @@
+#ifndef __PLATFORM_DATA_ASOC_MX27VIS_H
+#define __PLATFORM_DATA_ASOC_MX27VIS_H
+
+struct snd_mx27vis_platform_data {
+ int amp_gain0_gpio;
+ int amp_gain1_gpio;
+ int amp_mutel_gpio;
+ int amp_muter_gpio;
+};
+
+#endif /* __PLATFORM_DATA_ASOC_MX27VIS_H */
diff --git a/include/linux/platform_data/asoc-ti-mcbsp.h b/include/linux/platform_data/asoc-ti-mcbsp.h
index 18814127809a..c78d90b28b19 100644
--- a/include/linux/platform_data/asoc-ti-mcbsp.h
+++ b/include/linux/platform_data/asoc-ti-mcbsp.h
@@ -47,8 +47,6 @@ struct omap_mcbsp_platform_data {
bool has_wakeup; /* Wakeup capability */
bool has_ccr; /* Transceiver has configuration control registers */
int (*enable_st_clock)(unsigned int, bool);
- int (*set_clk_src)(struct device *dev, struct clk *clk, const char *src);
- int (*mux_signal)(struct device *dev, const char *signal, const char *src);
};
/**
diff --git a/include/linux/platform_data/davinci_asp.h b/include/linux/platform_data/davinci_asp.h
new file mode 100644
index 000000000000..d0c5825876f8
--- /dev/null
+++ b/include/linux/platform_data/davinci_asp.h
@@ -0,0 +1,105 @@
+/*
+ * TI DaVinci Audio Serial Port support
+ *
+ * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com/
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License as
+ * published by the Free Software Foundation version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any
+ * kind, whether express or implied; without even the implied warranty
+ * of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ */
+
+#ifndef __DAVINCI_ASP_H
+#define __DAVINCI_ASP_H
+
+struct snd_platform_data {
+ u32 tx_dma_offset;
+ u32 rx_dma_offset;
+ int asp_chan_q; /* event queue number for ASP channel */
+ int ram_chan_q; /* event queue number for RAM channel */
+ unsigned int codec_fmt;
+ /*
+ * Allowing this is more efficient and eliminates left and right swaps
+ * caused by underruns, but will swap the left and right channels
+ * when compared to previous behavior.
+ */
+ unsigned enable_channel_combine:1;
+ unsigned sram_size_playback;
+ unsigned sram_size_capture;
+
+ /*
+ * If McBSP peripheral gets the clock from an external pin,
+ * there are three chooses, that are MCBSP_CLKX, MCBSP_CLKR
+ * and MCBSP_CLKS.
+ * Depending on different hardware connections it is possible
+ * to use this setting to change the behaviour of McBSP
+ * driver.
+ */
+ int clk_input_pin;
+
+ /*
+ * This flag works when both clock and FS are outputs for the cpu
+ * and makes clock more accurate (FS is not symmetrical and the
+ * clock is very fast.
+ * The clock becoming faster is named
+ * i2s continuous serial clock (I2S_SCK) and it is an externally
+ * visible bit clock.
+ *
+ * first line : WordSelect
+ * second line : ContinuousSerialClock
+ * third line: SerialData
+ *
+ * SYMMETRICAL APPROACH:
+ * _______________________ LEFT
+ * _| RIGHT |______________________|
+ * _ _ _ _ _ _ _ _
+ * _| |_| |_ x16 _| |_| |_| |_| |_ x16 _| |_| |_
+ * _ _ _ _ _ _ _ _
+ * _/ \_/ \_ ... _/ \_/ \_/ \_/ \_ ... _/ \_/ \_
+ * \_/ \_/ \_/ \_/ \_/ \_/ \_/ \_/
+ *
+ * ACCURATE CLOCK APPROACH:
+ * ______________ LEFT
+ * _| RIGHT |_______________________________|
+ * _ _ _ _ _ _ _ _ _
+ * _| |_ x16 _| |_| |_ x16 _| |_| |_| |_| |_| |_| |
+ * _ _ _ _ dummy cycles
+ * _/ \_ ... _/ \_/ \_ ... _/ \__________________
+ * \_/ \_/ \_/ \_/
+ *
+ */
+ bool i2s_accurate_sck;
+
+ /* McASP specific fields */
+ int tdm_slots;
+ u8 op_mode;
+ u8 num_serializer;
+ u8 *serial_dir;
+ u8 version;
+ u8 txnumevt;
+ u8 rxnumevt;
+};
+
+enum {
+ MCASP_VERSION_1 = 0, /* DM646x */
+ MCASP_VERSION_2, /* DA8xx/OMAPL1x */
+ MCASP_VERSION_3, /* TI81xx/AM33xx */
+};
+
+enum mcbsp_clk_input_pin {
+ MCBSP_CLKR = 0, /* as in DM365 */
+ MCBSP_CLKS,
+};
+
+#define INACTIVE_MODE 0
+#define TX_MODE 1
+#define RX_MODE 2
+
+#define DAVINCI_MCASP_IIS_MODE 0
+#define DAVINCI_MCASP_DIT_MODE 1
+
+#endif
diff --git a/include/linux/platform_data/omap-twl4030.h b/include/linux/platform_data/omap-twl4030.h
new file mode 100644
index 000000000000..c7bef788daab
--- /dev/null
+++ b/include/linux/platform_data/omap-twl4030.h
@@ -0,0 +1,32 @@
+/**
+ * omap-twl4030.h - ASoC machine driver for TI SoC based boards with twl4030
+ * codec, header.
+ *
+ * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com
+ * All rights reserved.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef _OMAP_TWL4030_H_
+#define _OMAP_TWL4030_H_
+
+struct omap_tw4030_pdata {
+ const char *card_name;
+};
+
+#endif /* _OMAP_TWL4030_H_ */
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index fdeb8dceec0f..d315a08d6c6d 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -422,6 +422,7 @@
*/
struct snd_ac97;
+struct snd_pcm_chmap;
struct snd_ac97_build_ops {
int (*build_3d) (struct snd_ac97 *ac97);
@@ -528,6 +529,8 @@ struct snd_ac97 {
struct delayed_work power_work;
#endif
struct device dev;
+
+ struct snd_pcm_chmap *chmaps[2]; /* channel-maps (optional) */
};
#define to_ac97_t(d) container_of(d, struct snd_ac97, dev)
diff --git a/include/sound/ad1816a.h b/include/sound/ad1816a.h
index a7d8dc782e7c..abdf609c5918 100644
--- a/include/sound/ad1816a.h
+++ b/include/sound/ad1816a.h
@@ -147,6 +147,9 @@ struct snd_ad1816a {
unsigned int c_dma_size;
struct snd_timer *timer;
+#ifdef CONFIG_PM
+ unsigned short image[48];
+#endif
};
@@ -165,11 +168,15 @@ struct snd_ad1816a {
extern int snd_ad1816a_create(struct snd_card *card, unsigned long port,
int irq, int dma1, int dma2,
- struct snd_ad1816a **chip);
+ struct snd_ad1816a *chip);
extern int snd_ad1816a_pcm(struct snd_ad1816a *chip, int device, struct snd_pcm **rpcm);
extern int snd_ad1816a_mixer(struct snd_ad1816a *chip);
extern int snd_ad1816a_timer(struct snd_ad1816a *chip, int device,
struct snd_timer **rtimer);
+#ifdef CONFIG_PM
+extern void snd_ad1816a_suspend(struct snd_ad1816a *chip);
+extern void snd_ad1816a_resume(struct snd_ad1816a *chip);
+#endif
#endif /* __SOUND_AD1816A_H */
diff --git a/include/sound/asound.h b/include/sound/asound.h
index 0876a1e76aef..dfe7d441748c 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -472,6 +472,45 @@ enum {
SNDRV_PCM_TSTAMP_TYPE_LAST = SNDRV_PCM_TSTAMP_TYPE_MONOTONIC,
};
+/* 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,
+};
+
+#define SNDRV_CHMAP_POSITION_MASK 0xffff
+#define SNDRV_CHMAP_PHASE_INVERSE (0x01 << 16)
+#define SNDRV_CHMAP_DRIVER_SPEC (0x02 << 16)
+
#define SNDRV_PCM_IOCTL_PVERSION _IOR('A', 0x00, int)
#define SNDRV_PCM_IOCTL_INFO _IOR('A', 0x01, struct snd_pcm_info)
#define SNDRV_PCM_IOCTL_TSTAMP _IOW('A', 0x02, int)
diff --git a/include/sound/compress_driver.h b/include/sound/compress_driver.h
index 48f2a1ff2bbc..f2912abacdf3 100644
--- a/include/sound/compress_driver.h
+++ b/include/sound/compress_driver.h
@@ -61,6 +61,7 @@ struct snd_compr_runtime {
u64 total_bytes_available;
u64 total_bytes_transferred;
wait_queue_head_t sleep;
+ void *private_data;
};
/**
diff --git a/include/sound/compress_params.h b/include/sound/compress_params.h
index da4a456de032..602dc6c45d1a 100644
--- a/include/sound/compress_params.h
+++ b/include/sound/compress_params.h
@@ -72,6 +72,7 @@
#define SND_AUDIOCODEC_IEC61937 ((__u32) 0x0000000B)
#define SND_AUDIOCODEC_G723_1 ((__u32) 0x0000000C)
#define SND_AUDIOCODEC_G729 ((__u32) 0x0000000D)
+#define SND_AUDIOCODEC_MAX SND_AUDIOCODEC_G729
/*
* Profile and modes are listed with bit masks. This allows for a
diff --git a/include/sound/da9055.h b/include/sound/da9055.h
new file mode 100644
index 000000000000..cf1241b64d89
--- /dev/null
+++ b/include/sound/da9055.h
@@ -0,0 +1,33 @@
+/*
+ * DA9055 ALSA Soc codec driver
+ *
+ * Copyright (c) 2012 Dialog Semiconductor
+ *
+ * Tested on (Samsung SMDK6410 board + DA9055 EVB) using I2S and I2C
+ * Written by David Chen <david.chen@diasemi.com> and
+ * Ashish Chavan <ashish.chavan@kpitcummins.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef __SOUND_DA9055_H__
+#define __SOUND_DA9055_H__
+
+enum da9055_micbias_voltage {
+ DA9055_MICBIAS_1_6V = 0,
+ DA9055_MICBIAS_1_8V = 1,
+ DA9055_MICBIAS_2_1V = 2,
+ DA9055_MICBIAS_2_2V = 3,
+};
+
+struct da9055_platform_data {
+ /* Selects which of the two MicBias pins acts as the bias source */
+ bool micbias_source;
+ /* Selects the micbias voltage */
+ enum da9055_micbias_voltage micbias;
+};
+
+#endif
diff --git a/include/sound/emu10k1.h b/include/sound/emu10k1.h
index 4f865df42f0f..1a33f48ebe78 100644
--- a/include/sound/emu10k1.h
+++ b/include/sound/emu10k1.h
@@ -1788,7 +1788,7 @@ struct snd_emu10k1 {
unsigned int efx_voices_mask[2];
unsigned int next_free_voice;
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
unsigned int *saved_ptr;
unsigned int *saved_gpr;
unsigned int *tram_val_saved;
@@ -1856,7 +1856,7 @@ unsigned short snd_emu10k1_ac97_read(struct snd_ac97 *ac97, unsigned short reg);
void snd_emu10k1_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short data);
unsigned int snd_emu10k1_rate_to_pitch(unsigned int rate);
-#ifdef CONFIG_PM
+#ifdef CONFIG_PM_SLEEP
void snd_emu10k1_suspend_regs(struct snd_emu10k1 *emu);
void snd_emu10k1_resume_init(struct snd_emu10k1 *emu);
void snd_emu10k1_resume_regs(struct snd_emu10k1 *emu);
diff --git a/include/sound/initval.h b/include/sound/initval.h
index f99a0d2ddfe7..ac62c67e6f42 100644
--- a/include/sound/initval.h
+++ b/include/sound/initval.h
@@ -50,6 +50,20 @@
#define SNDRV_DEFAULT_DMA_SIZE { [0 ... (SNDRV_CARDS-1)] = SNDRV_AUTO_DMA_SIZE }
#define SNDRV_DEFAULT_PTR SNDRV_DEFAULT_STR
+#ifdef SNDRV_LEGACY_FIND_FREE_IOPORT
+static long snd_legacy_find_free_ioport(long *port_table, long size)
+{
+ while (*port_table != -1) {
+ if (request_region(*port_table, size, "ALSA test")) {
+ release_region(*port_table, size);
+ return *port_table;
+ }
+ port_table++;
+ }
+ return -1;
+}
+#endif
+
#ifdef SNDRV_LEGACY_FIND_FREE_IRQ
#include <linux/interrupt.h>
diff --git a/include/sound/memalloc.h b/include/sound/memalloc.h
index c42506212649..844af65af626 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -98,8 +98,10 @@ static inline unsigned int snd_sgbuf_aligned_pages(size_t size)
/*
* return the physical address at the corresponding offset
*/
-static inline dma_addr_t snd_sgbuf_get_addr(struct snd_sg_buf *sgbuf, size_t offset)
+static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
+ size_t offset)
{
+ struct snd_sg_buf *sgbuf = dmab->private_data;
dma_addr_t addr = sgbuf->table[offset >> PAGE_SHIFT].addr;
addr &= PAGE_MASK;
return addr + offset % PAGE_SIZE;
@@ -108,10 +110,31 @@ static inline dma_addr_t snd_sgbuf_get_addr(struct snd_sg_buf *sgbuf, size_t off
/*
* return the virtual address at the corresponding offset
*/
-static inline void *snd_sgbuf_get_ptr(struct snd_sg_buf *sgbuf, size_t offset)
+static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
+ size_t offset)
{
+ struct snd_sg_buf *sgbuf = dmab->private_data;
return sgbuf->table[offset >> PAGE_SHIFT].buf + offset % PAGE_SIZE;
}
+
+unsigned int snd_sgbuf_get_chunk_size(struct snd_dma_buffer *dmab,
+ unsigned int ofs, unsigned int size);
+#else
+/* non-SG versions */
+static inline dma_addr_t snd_sgbuf_get_addr(struct snd_dma_buffer *dmab,
+ size_t offset)
+{
+ return dmab->addr + offset;
+}
+
+static inline void *snd_sgbuf_get_ptr(struct snd_dma_buffer *dmab,
+ size_t offset)
+{
+ return dmab->area + offset;
+}
+
+#define snd_sgbuf_get_chunk_size(dmab, ofs, size) (size)
+
#endif /* CONFIG_SND_DMA_SGBUF */
/* allocate/release a buffer */
diff --git a/include/sound/pcm.h b/include/sound/pcm.h
index d0711bc8c914..6268a4192d5c 100644
--- a/include/sound/pcm.h
+++ b/include/sound/pcm.h
@@ -437,6 +437,7 @@ struct snd_pcm_str {
struct snd_info_entry *proc_xrun_debug_entry;
#endif
#endif
+ struct snd_kcontrol *chmap_kctl; /* channel-mapping controls */
};
struct snd_pcm {
@@ -982,53 +983,42 @@ static int snd_pcm_lib_alloc_vmalloc_32_buffer
_snd_pcm_lib_alloc_vmalloc_buffer \
(subs, size, GFP_KERNEL | GFP_DMA32 | __GFP_ZERO)
+#define snd_pcm_get_dma_buf(substream) ((substream)->runtime->dma_buffer_p)
+
#ifdef CONFIG_SND_DMA_SGBUF
/*
* SG-buffer handling
*/
#define snd_pcm_substream_sgbuf(substream) \
- ((substream)->runtime->dma_buffer_p->private_data)
-
-static inline dma_addr_t
-snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
-{
- struct snd_sg_buf *sg = snd_pcm_substream_sgbuf(substream);
- return snd_sgbuf_get_addr(sg, ofs);
-}
-
-static inline void *
-snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs)
-{
- struct snd_sg_buf *sg = snd_pcm_substream_sgbuf(substream);
- return snd_sgbuf_get_ptr(sg, ofs);
-}
+ snd_pcm_get_dma_buf(substream)->private_data
struct page *snd_pcm_sgbuf_ops_page(struct snd_pcm_substream *substream,
unsigned long offset);
-unsigned int snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
- unsigned int ofs, unsigned int size);
-
#else /* !SND_DMA_SGBUF */
/*
* fake using a continuous buffer
*/
+#define snd_pcm_sgbuf_ops_page NULL
+#endif /* SND_DMA_SGBUF */
+
static inline dma_addr_t
snd_pcm_sgbuf_get_addr(struct snd_pcm_substream *substream, unsigned int ofs)
{
- return substream->runtime->dma_addr + ofs;
+ return snd_sgbuf_get_addr(snd_pcm_get_dma_buf(substream), ofs);
}
static inline void *
snd_pcm_sgbuf_get_ptr(struct snd_pcm_substream *substream, unsigned int ofs)
{
- return substream->runtime->dma_area + ofs;
+ return snd_sgbuf_get_ptr(snd_pcm_get_dma_buf(substream), ofs);
}
-#define snd_pcm_sgbuf_ops_page NULL
-
-#define snd_pcm_sgbuf_get_chunk_size(subs, ofs, size) (size)
-
-#endif /* SND_DMA_SGBUF */
+static inline unsigned int
+snd_pcm_sgbuf_get_chunk_size(struct snd_pcm_substream *substream,
+ unsigned int ofs, unsigned int size)
+{
+ return snd_sgbuf_get_chunk_size(snd_pcm_get_dma_buf(substream), ofs, size);
+}
/* handle mmap counter - PCM mmap callback should handle this counter properly */
static inline void snd_pcm_mmap_data_open(struct vm_area_struct *area)
@@ -1086,4 +1076,51 @@ static inline const char *snd_pcm_stream_str(struct snd_pcm_substream *substream
return "Capture";
}
+/*
+ * PCM channel-mapping control API
+ */
+/* array element of channel maps */
+struct snd_pcm_chmap_elem {
+ unsigned char channels;
+ unsigned char map[15];
+};
+
+/* channel map information; retrieved via snd_kcontrol_chip() */
+struct snd_pcm_chmap {
+ struct snd_pcm *pcm; /* assigned PCM instance */
+ int stream; /* PLAYBACK or CAPTURE */
+ struct snd_kcontrol *kctl;
+ const struct snd_pcm_chmap_elem *chmap;
+ unsigned int max_channels;
+ unsigned int channel_mask; /* optional: active channels bitmask */
+ void *private_data; /* optional: private data pointer */
+};
+
+/* get the PCM substream assigned to the given chmap info */
+static inline struct snd_pcm_substream *
+snd_pcm_chmap_substream(struct snd_pcm_chmap *info, unsigned int idx)
+{
+ struct snd_pcm_substream *s;
+ for (s = info->pcm->streams[info->stream].substream; s; s = s->next)
+ if (s->number == idx)
+ return s;
+ return NULL;
+}
+
+/* ALSA-standard channel maps (RL/RR prior to C/LFE) */
+extern const struct snd_pcm_chmap_elem snd_pcm_std_chmaps[];
+/* Other world's standard channel maps (C/LFE prior to RL/RR) */
+extern const struct snd_pcm_chmap_elem snd_pcm_alt_chmaps[];
+
+/* bit masks to be passed to snd_pcm_chmap.channel_mask field */
+#define SND_PCM_CHMAP_MASK_24 ((1U << 2) | (1U << 4))
+#define SND_PCM_CHMAP_MASK_246 (SND_PCM_CHMAP_MASK_24 | (1U << 6))
+#define SND_PCM_CHMAP_MASK_2468 (SND_PCM_CHMAP_MASK_246 | (1U << 8))
+
+int snd_pcm_add_chmap_ctls(struct snd_pcm *pcm, int stream,
+ const struct snd_pcm_chmap_elem *chmap,
+ int max_channels,
+ unsigned long private_value,
+ struct snd_pcm_chmap **info_ret);
+
#endif /* __SOUND_PCM_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 1f69e0af2941..628db7bca4fd 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -18,6 +18,7 @@
struct snd_pcm_substream;
struct snd_soc_dapm_widget;
+struct snd_compr_stream;
/*
* DAI hardware audio formats.
@@ -205,6 +206,8 @@ struct snd_soc_dai_driver {
int (*remove)(struct snd_soc_dai *dai);
int (*suspend)(struct snd_soc_dai *dai);
int (*resume)(struct snd_soc_dai *dai);
+ /* compress dai */
+ bool compress_dai;
/* ops */
const struct snd_soc_dai_ops *ops;
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index abe373d57adc..e1ef63d4a5c4 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -244,10 +244,11 @@ struct device;
{ .id = snd_soc_dapm_supply, .name = wname, .reg = wreg, \
.shift = wshift, .invert = winvert, .event = wevent, \
.event_flags = wflags}
-#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay) \
+#define SND_SOC_DAPM_REGULATOR_SUPPLY(wname, wdelay, wflags) \
{ .id = snd_soc_dapm_regulator_supply, .name = wname, \
.reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \
- .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD }
+ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \
+ .invert = wflags}
/* dapm kcontrol types */
@@ -319,6 +320,9 @@ struct device;
#define SND_SOC_DAPM_EVENT_OFF(e) \
(e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD))
+/* regulator widget flags */
+#define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */
+
struct snd_soc_dapm_widget;
enum snd_soc_dapm_type;
struct snd_soc_dapm_path;
@@ -412,6 +416,7 @@ void snd_soc_dapm_auto_nc_codec_pins(struct snd_soc_codec *codec);
/* Mostly internal - should not normally be used */
void dapm_mark_dirty(struct snd_soc_dapm_widget *w, const char *reason);
+void dapm_mark_io_dirty(struct snd_soc_dapm_context *dapm);
/* dapm path query */
int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream,
@@ -510,7 +515,6 @@ struct snd_soc_dapm_widget {
/* dapm control */
int reg; /* negative reg = no direct dapm */
unsigned char shift; /* bits to shift */
- unsigned int saved_value; /* widget saved value */
unsigned int value; /* widget current value */
unsigned int mask; /* non-shifted mask */
unsigned int on_val; /* on state value */
diff --git a/include/sound/soc.h b/include/sound/soc.h
index e063380f63a2..91244a096c19 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -20,8 +20,10 @@
#include <linux/interrupt.h>
#include <linux/kernel.h>
#include <linux/regmap.h>
+#include <linux/log2.h>
#include <sound/core.h>
#include <sound/pcm.h>
+#include <sound/compress_driver.h>
#include <sound/control.h>
#include <sound/ac97_codec.h>
@@ -159,7 +161,8 @@
.platform_max = xmax} }
#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \
{ .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \
- .max = xmax, .texts = xtexts }
+ .max = xmax, .texts = xtexts, \
+ .mask = xmax ? roundup_pow_of_two(xmax) - 1 : 0}
#define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \
SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts)
#define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \
@@ -399,6 +402,7 @@ int snd_soc_platform_read(struct snd_soc_platform *platform,
int snd_soc_platform_write(struct snd_soc_platform *platform,
unsigned int reg, unsigned int val);
int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num);
+int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num);
struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card,
const char *dai_link, int stream);
@@ -632,6 +636,13 @@ struct snd_soc_ops {
int (*trigger)(struct snd_pcm_substream *, int);
};
+struct snd_soc_compr_ops {
+ int (*startup)(struct snd_compr_stream *);
+ void (*shutdown)(struct snd_compr_stream *);
+ int (*set_params)(struct snd_compr_stream *);
+ int (*trigger)(struct snd_compr_stream *);
+};
+
/* SoC cache ops */
struct snd_soc_cache_ops {
const char *name;
@@ -787,9 +798,12 @@ struct snd_soc_platform_driver {
snd_pcm_sframes_t (*delay)(struct snd_pcm_substream *,
struct snd_soc_dai *);
- /* platform stream ops */
+ /* platform stream pcm ops */
struct snd_pcm_ops *ops;
+ /* platform stream compress ops */
+ struct snd_compr_ops *compr_ops;
+
/* platform stream completion event */
int (*stream_event)(struct snd_soc_dapm_context *dapm, int event);
@@ -891,6 +905,7 @@ struct snd_soc_dai_link {
/* machine stream operations */
struct snd_soc_ops *ops;
+ struct snd_soc_compr_ops *compr_ops;
};
struct snd_soc_codec_conf {
@@ -1027,6 +1042,7 @@ struct snd_soc_pcm_runtime {
/* runtime devices */
struct snd_pcm *pcm;
+ struct snd_compr *compr;
struct snd_soc_codec *codec;
struct snd_soc_platform *platform;
struct snd_soc_dai *codec_dai;
diff --git a/include/sound/tegra_wm8903.h b/include/sound/tegra_wm8903.h
new file mode 100644
index 000000000000..57b202ee97c3
--- /dev/null
+++ b/include/sound/tegra_wm8903.h
@@ -0,0 +1,26 @@
+/*
+ * Copyright 2011 NVIDIA, Inc.
+ *
+ * This software is licensed under the terms of the GNU General Public
+ * License version 2, as published by the Free Software Foundation, and
+ * may be copied, distributed, and modified under those terms.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ */
+
+#ifndef __SOUND_TEGRA_WM38903_H
+#define __SOUND_TEGRA_WM38903_H
+
+struct tegra_wm8903_platform_data {
+ int gpio_spkr_en;
+ int gpio_hp_det;
+ int gpio_hp_mute;
+ int gpio_int_mic_en;
+ int gpio_ext_mic_en;
+};
+
+#endif
diff --git a/include/sound/tlv.h b/include/sound/tlv.h
index a64d8fe3f855..28c65e1ada21 100644
--- a/include/sound/tlv.h
+++ b/include/sound/tlv.h
@@ -86,4 +86,12 @@
#define TLV_DB_GAIN_MUTE -9999999
+/*
+ * channel-mapping TLV items
+ * TLV length must match with num_channels
+ */
+#define SNDRV_CTL_TLVT_CHMAP_FIXED 0x101 /* fixed channel position */
+#define SNDRV_CTL_TLVT_CHMAP_VAR 0x102 /* channels freely swappable */
+#define SNDRV_CTL_TLVT_CHMAP_PAIRED 0x103 /* pair-wise swappable */
+
#endif /* __SOUND_TLV_H */
diff --git a/include/sound/version.h b/include/sound/version.h
deleted file mode 100644
index cc75024c1089..000000000000
--- a/include/sound/version.h
+++ /dev/null
@@ -1,3 +0,0 @@
-/* include/version.h */
-#define CONFIG_SND_VERSION "1.0.25"
-#define CONFIG_SND_DATE ""
diff --git a/include/sound/wm0010.h b/include/sound/wm0010.h
new file mode 100644
index 000000000000..3261e90815af
--- /dev/null
+++ b/include/sound/wm0010.h
@@ -0,0 +1,27 @@
+/*
+ * wm0010.h -- Platform data for WM0010 DSP Driver
+ *
+ * Copyright 2012 Wolfson Microelectronics PLC.
+ *
+ * Author: Dimitris Papastamos <dp@opensource.wolfsonmicro.com>
+ *
+ * This program is free software; you can redistribute it and/or modify it
+ * under the terms of the GNU General Public License as published by the
+ * Free Software Foundation; either version 2 of the License, or (at your
+ * option) any later version.
+ */
+
+#ifndef WM0010_PDATA_H
+#define WM0010_PDATA_H
+
+struct wm0010_pdata {
+ int gpio_reset;
+
+ /* Set if there is an inverter between the GPIO controlling
+ * the reset signal and the device.
+ */
+ int reset_active_high;
+ int irq_flags;
+};
+
+#endif
diff --git a/include/sound/wm8960.h b/include/sound/wm8960.h
index 74e9a95529c5..e8ce8ee7d62d 100644
--- a/include/sound/wm8960.h
+++ b/include/sound/wm8960.h
@@ -18,7 +18,7 @@
struct wm8960_data {
bool capless; /* Headphone outputs configured in capless mode */
- int dres; /* Discharge resistance for headphone outputs */
+ bool shared_lrclk; /* DAC and ADC LRCLKs are wired together */
};
#endif
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
index eee19f63c0d8..8016fd826f5a 100644
--- a/include/sound/wm8993.h
+++ b/include/sound/wm8993.h
@@ -32,6 +32,10 @@ struct wm8993_platform_data {
unsigned int lineout1fb:1;
unsigned int lineout2fb:1;
+ /* Delay to add for microphones to stabalise after power up */
+ int micbias1_delay;
+ int micbias2_delay;
+
/* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
unsigned int micbias1_lvl:1;
unsigned int micbias2_lvl:1;
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