Commit be3bfbba authored by Linus Torvalds's avatar Linus Torvalds

Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
  ALSA: ASoC codec: remove unused #include <version.h>
  ALSA: ASoC: update email address for Liam Girdwood
  ALSA: hda: corrected invalid mixer values
  ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
  ALSA: ASoC: Add destination and source port for DMA on OMAP1
  ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
  ALSA: ASoC: Fix build of GTA01 audio driver
  ALSA: ASoC: Add widgets before setting endpoints on GTA01
  ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
  ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
  ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
  ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
  ALSA: ASoC: Make TLV320AIC26 user-visible
  ALSA: ASoC - clean up Kconfig for TLV320AIC2
  ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
  ALSA: ASoC: Implement WM8510 bias level control
  ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
  ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
  ALSA: ASoC: Add WM8510 SPI support
  ALSA: ASoC: Add WM8753 SPI support
  ...
parents 20272c89 7dc85076
......@@ -240,6 +240,7 @@ int snd_soc_dapm_sys_add(struct device *dev);
/* dapm audio pin control and status */
int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
int snd_soc_dapm_sync(struct snd_soc_codec *codec);
......
......@@ -30,7 +30,7 @@
**************************************************************************
*
* History
* May 02, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* May 02, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
* Removed non existant WM9700
* Added support for WM9705, WM9708, WM9709, WM9710, WM9711
* WM9712 and WM9717
......
......@@ -476,7 +476,7 @@ static int patch_yamaha_ymf753(struct snd_ac97 * ac97)
}
/*
* May 2, 2003 Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* May 2, 2003 Liam Girdwood <lrg@slimlogic.co.uk>
* removed broken wolfson00 patch.
* added support for WM9705,WM9708,WM9709,WM9710,WM9711,WM9712 and WM9717.
*/
......
......@@ -322,8 +322,8 @@ static hda_nid_t stac92hd71bxx_mux_nids[2] = {
0x1a, 0x1b
};
static hda_nid_t stac92hd71bxx_dmux_nids[1] = {
0x1c,
static hda_nid_t stac92hd71bxx_dmux_nids[2] = {
0x1c, 0x1d,
};
static hda_nid_t stac92hd71bxx_smux_nids[2] = {
......@@ -861,20 +861,18 @@ static struct hda_verb stac92hd71bxx_core_init[] = {
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
/* connect headphone jack to dac1 */
{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
/* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */
{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
};
#define HD_DISABLE_PORTF 3
#define HD_DISABLE_PORTF 2
static struct hda_verb stac92hd71bxx_analog_core_init[] = {
/* start of config #1 */
/* connect port 0f to audio mixer */
{ 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2},
{ 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */
/* unmute right and left channels for node 0x0f */
{ 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
/* start of config #2 */
......@@ -883,10 +881,6 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = {
{ 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff},
/* connect headphone jack to dac1 */
{ 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01},
/* connect port 0d to audio mixer */
{ 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2},
/* unmute dac0 input in audio mixer */
{ 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f},
/* unmute right and left channels for nodes 0x0a, 0xd */
{ 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
{ 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)},
......@@ -1107,6 +1101,7 @@ static struct snd_kcontrol_new stac92hd83xxx_mixer[] = {
static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
STAC_INPUT_SOURCE(2),
STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2),
HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT),
HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT),
......@@ -1119,8 +1114,17 @@ static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = {
HDA_CODEC_MUTE("PC Beep Switch", 0x17, 0x2, HDA_INPUT),
*/
HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT),
HDA_CODEC_MUTE("Import0 Mux Capture Switch", 0x17, 0x0, HDA_INPUT),
HDA_CODEC_VOLUME("Import0 Mux Capture Volume", 0x17, 0x0, HDA_INPUT),
HDA_CODEC_MUTE("Import1 Mux Capture Switch", 0x17, 0x1, HDA_INPUT),
HDA_CODEC_VOLUME("Import1 Mux Capture Volume", 0x17, 0x1, HDA_INPUT),
HDA_CODEC_MUTE("DAC0 Capture Switch", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_VOLUME("DAC0 Capture Volume", 0x17, 0x3, HDA_INPUT),
HDA_CODEC_MUTE("DAC1 Capture Switch", 0x17, 0x4, HDA_INPUT),
HDA_CODEC_VOLUME("DAC1 Capture Volume", 0x17, 0x4, HDA_INPUT),
{ } /* end */
};
......@@ -1649,7 +1653,7 @@ static struct snd_pci_quirk stac92hd83xxx_cfg_tbl[] = {
static unsigned int ref92hd71bxx_pin_configs[11] = {
0x02214030, 0x02a19040, 0x01a19020, 0x01014010,
0x0181302e, 0x01114010, 0x01019020, 0x90a000f0,
0x0181302e, 0x01014010, 0x01019020, 0x90a000f0,
0x90a000f0, 0x01452050, 0x01452050,
};
......@@ -3000,7 +3004,7 @@ static int stac92xx_auto_create_mono_output_ctls(struct hda_codec *codec)
/* labels for amp mux outputs */
static const char *stac92xx_amp_labels[3] = {
"Front Microphone", "Microphone", "Line In"
"Front Microphone", "Microphone", "Line In",
};
/* create amp out controls mux on capable codecs */
......@@ -4327,6 +4331,16 @@ static struct hda_codec_ops stac92hd71bxx_patch_ops = {
#endif
};
static struct hda_input_mux stac92hd71bxx_dmux = {
.num_items = 4,
.items = {
{ "Analog Inputs", 0x00 },
{ "Mixer", 0x01 },
{ "Digital Mic 1", 0x02 },
{ "Digital Mic 2", 0x03 },
}
};
static int patch_stac92hd71bxx(struct hda_codec *codec)
{
struct sigmatel_spec *spec;
......@@ -4341,6 +4355,8 @@ static int patch_stac92hd71bxx(struct hda_codec *codec)
spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids);
spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids);
spec->pin_nids = stac92hd71bxx_pin_nids;
memcpy(&spec->private_dimux, &stac92hd71bxx_dmux,
sizeof(stac92hd71bxx_dmux));
spec->board_config = snd_hda_check_board_config(codec,
STAC_92HD71BXX_MODELS,
stac92hd71bxx_models,
......@@ -4392,6 +4408,7 @@ again:
/* no output amps */
spec->num_pwrs = 0;
spec->mixer = stac92hd71bxx_analog_mixer;
spec->dinput_mux = &spec->private_dimux;
/* disable VSW */
spec->init = &stac92hd71bxx_analog_core_init[HD_DISABLE_PORTF];
......@@ -4409,12 +4426,13 @@ again:
spec->num_pwrs = 0;
/* fallthru */
default:
spec->dinput_mux = &spec->private_dimux;
spec->mixer = stac92hd71bxx_analog_mixer;
spec->init = stac92hd71bxx_analog_core_init;
codec->slave_dig_outs = stac92hd71bxx_slave_dig_outs;
}
spec->aloopback_mask = 0x20;
spec->aloopback_mask = 0x50;
spec->aloopback_shift = 0;
if (spec->board_config > STAC_92HD71BXX_REF) {
......@@ -4456,6 +4474,10 @@ again:
spec->multiout.num_dacs = 1;
spec->multiout.hp_nid = 0x11;
spec->multiout.dac_nids = stac92hd71bxx_dac_nids;
if (spec->dinput_mux)
spec->private_dimux.num_items +=
spec->num_dmics -
(ARRAY_SIZE(stac92hd71bxx_dmic_nids) - 1);
err = stac92xx_parse_auto_config(codec, 0x21, 0x23);
if (!err) {
......
......@@ -8,20 +8,3 @@ config SND_AT91_SOC
config SND_AT91_SOC_SSC
tristate
config SND_AT91_SOC_ETI_B1_WM8731
tristate "SoC Audio support for WM8731-based Endrelia ETI-B1 boards"
depends on SND_AT91_SOC && (MACH_ETI_B1 || MACH_ETI_C1)
select SND_AT91_SOC_SSC
select SND_SOC_WM8731
help
Say Y if you want to add support for SoC audio on WM8731-based
Endrelia Technologies Inc ETI-B1 or ETI-C1 boards.
config SND_AT91_SOC_ETI_SLAVE
bool "Run codec in slave Mode on Endrelia boards"
depends on SND_AT91_SOC_ETI_B1_WM8731
default n
help
Say Y if you want to run with the AT91 SSC generating the BCLK
and LRC signals on Endrelia boards.
......@@ -4,8 +4,3 @@ snd-soc-at91-ssc-objs := at91-ssc.o
obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o
# AT91 Machine Support
snd-soc-eti-b1-wm8731-objs := eti_b1_wm8731.o
obj-$(CONFIG_SND_AT91_SOC_ETI_B1_WM8731) += snd-soc-eti-b1-wm8731.o
......@@ -5,7 +5,7 @@
* Endrelia Technologies Inc.
*
* Based on pxa2xx Platform drivers by
* Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Liam Girdwood <lrg@slimlogic.co.uk>
*
* 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
......
/*
* eti_b1_wm8731 -- SoC audio for AT91RM9200-based Endrelia ETI_B1 board.
*
* Author: Frank Mandarino <fmandarino@endrelia.com>
* Endrelia Technologies Inc.
* Created: Mar 29, 2006
*
* Based on corgi.c by:
*
* Copyright 2005 Wolfson Microelectronics PLC.
* Copyright 2005 Openedhand Ltd.
*
* Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
* Richard Purdie <richard@openedhand.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.
*
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/kernel.h>
#include <linux/clk.h>
#include <linux/timer.h>
#include <linux/interrupt.h>
#include <linux/platform_device.h>
#include <sound/core.h>
#include <sound/pcm.h>
#include <sound/soc.h>
#include <sound/soc-dapm.h>
#include <mach/hardware.h>
#include <mach/gpio.h>
#include "../codecs/wm8731.h"
#include "at91-pcm.h"
#include "at91-ssc.h"
#if 0
#define DBG(x...) printk(KERN_INFO "eti_b1_wm8731: " x)
#else
#define DBG(x...)
#endif
static struct clk *pck1_clk;
static struct clk *pllb_clk;
static int eti_b1_startup(struct snd_pcm_substream *substream)
{
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
int ret;
/* cpu clock is the AT91 master clock sent to the SSC */
ret = snd_soc_dai_set_sysclk(cpu_dai, AT91_SYSCLK_MCK,
60000000, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* codec system clock is supplied by PCK1, set to 12MHz */
ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
12000000, SND_SOC_CLOCK_IN);
if (ret < 0)
return ret;
/* Start PCK1 clock. */
clk_enable(pck1_clk);
DBG("pck1 started\n");
return 0;
}
static void eti_b1_shutdown(struct snd_pcm_substream *substream)
{
/* Stop PCK1 clock. */
clk_disable(pck1_clk);
DBG("pck1 stopped\n");
}
static int eti_b1_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->dai->codec_dai;
struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
int ret;
#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
unsigned int rate;
int cmr_div, period;
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
if (ret < 0)
return ret;
/*
* The SSC clock dividers depend on the sample rate. The CMR.DIV
* field divides the system master clock MCK to drive the SSC TK
* signal which provides the codec BCLK. The TCMR.PERIOD and
* RCMR.PERIOD fields further divide the BCLK signal to drive
* the SSC TF and RF signals which provide the codec DACLRC and
* ADCLRC clocks.
*
* The dividers were determined through trial and error, where a
* CMR.DIV value is chosen such that the resulting BCLK value is
* divisible, or almost divisible, by (2 * sample rate), and then
* the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
*/
rate = params_rate(params);
switch (rate) {
case 8000:
cmr_div = 25; /* BCLK = 60MHz/(2*25) = 1.2MHz */
period = 74; /* LRC = BCLK/(2*(74+1)) = 8000Hz */
break;
case 32000:
cmr_div = 7; /* BCLK = 60MHz/(2*7) ~= 4.28571428MHz */
period = 66; /* LRC = BCLK/(2*(66+1)) = 31982.942Hz */
break;
case 48000:
cmr_div = 13; /* BCLK = 60MHz/(2*13) ~= 2.3076923MHz */
period = 23; /* LRC = BCLK/(2*(23+1)) = 48076.923Hz */
break;
default:
printk(KERN_WARNING "unsupported rate %d on ETI-B1 board\n", rate);
return -EINVAL;
}
/* set the MCK divider for BCLK */
ret = snd_soc_dai_set_clkdiv(cpu_dai, AT91SSC_CMR_DIV, cmr_div);
if (ret < 0)
return ret;
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
/* set the BCLK divider for DACLRC */
ret = snd_soc_dai_set_clkdiv(cpu_dai,
AT91SSC_TCMR_PERIOD, period);
} else {
/* set the BCLK divider for ADCLRC */
ret = snd_soc_dai_set_clkdiv(cpu_dai,
AT91SSC_RCMR_PERIOD, period);
}
if (ret < 0)
return ret;
#else /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
/*
* Codec in Master Mode.
*/
/* set codec DAI configuration */
ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
/* set cpu DAI configuration */
ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
#endif /* CONFIG_SND_AT91_SOC_ETI_SLAVE */
return 0;
}
static struct snd_soc_ops eti_b1_ops = {
.startup = eti_b1_startup,
.hw_params = eti_b1_hw_params,
.shutdown = eti_b1_shutdown,
};
static const struct snd_soc_dapm_widget eti_b1_dapm_widgets[] = {
SND_SOC_DAPM_MIC("Int Mic", NULL),
SND_SOC_DAPM_SPK("Ext Spk", NULL),
};
static const struct snd_soc_dapm_route intercon[] = {
/* speaker connected to LHPOUT */
{"Ext Spk", NULL, "LHPOUT"},
/* mic is connected to Mic Jack, with WM8731 Mic Bias */
{"MICIN", NULL, "Mic Bias"},
{"Mic Bias", NULL, "Int Mic"},
};
/*
* Logic for a wm8731 as connected on a Endrelia ETI-B1 board.
*/
static int eti_b1_wm8731_init(struct snd_soc_codec *codec)
{
DBG("eti_b1_wm8731_init() called\n");
/* Add specific widgets */
snd_soc_dapm_new_controls(codec, eti_b1_dapm_widgets,
ARRAY_SIZE(eti_b1_dapm_widgets));
/* Set up specific audio path interconnects */
snd_soc_dapm_add_route(codec, intercon, ARRAY_SIZE(intercon));
/* not connected */
snd_soc_dapm_disable_pin(codec, "RLINEIN");
snd_soc_dapm_disable_pin(codec, "LLINEIN");
/* always connected */
snd_soc_dapm_enable_pin(codec, "Int Mic");
snd_soc_dapm_enable_pin(codec, "Ext Spk");
snd_soc_dapm_sync(codec);
return 0;
}
static struct snd_soc_dai_link eti_b1_dai = {
.name = "WM8731",
.stream_name = "WM8731 PCM",
.cpu_dai = &at91_ssc_dai[1],
.codec_dai = &wm8731_dai,
.init = eti_b1_wm8731_init,
.ops = &eti_b1_ops,
};
static struct snd_soc_machine snd_soc_machine_eti_b1 = {
.name = "ETI_B1_WM8731",
.dai_link = &eti_b1_dai,
.num_links = 1,
};
static struct wm8731_setup_data eti_b1_wm8731_setup = {
.i2c_bus = 0,
.i2c_address = 0x1a,
};
static struct snd_soc_device eti_b1_snd_devdata = {
.machine = &snd_soc_machine_eti_b1,
.platform = &at91_soc_platform,
.codec_dev = &soc_codec_dev_wm8731,
.codec_data = &eti_b1_wm8731_setup,
};
static struct platform_device *eti_b1_snd_device;
static int __init eti_b1_init(void)
{
int ret;
struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
if (!request_mem_region(AT91RM9200_BASE_SSC1, SZ_16K, "soc-audio")) {
DBG("SSC1 memory region is busy\n");
return -EBUSY;
}
ssc->base = ioremap(AT91RM9200_BASE_SSC1, SZ_16K);
if (!ssc->base) {
DBG("SSC1 memory ioremap failed\n");
ret = -ENOMEM;
goto fail_release_mem;
}
ssc->pid = AT91RM9200_ID_SSC1;
eti_b1_snd_device = platform_device_alloc("soc-audio", -1);
if (!eti_b1_snd_device) {
DBG("platform device allocation failed\n");
ret = -ENOMEM;
goto fail_io_unmap;
}
platform_set_drvdata(eti_b1_snd_device, &eti_b1_snd_devdata);
eti_b1_snd_devdata.dev = &eti_b1_snd_device->dev;
ret = platform_device_add(eti_b1_snd_device);
if (ret) {
DBG("platform device add failed\n");
platform_device_put(eti_b1_snd_device);
goto fail_io_unmap;
}
at91_set_A_periph(AT91_PIN_PB6, 0); /* TF1 */
at91_set_A_periph(AT91_PIN_PB7, 0); /* TK1 */
at91_set_A_periph(AT91_PIN_PB8, 0); /* TD1 */
at91_set_A_periph(AT91_PIN_PB9, 0); /* RD1 */
/* at91_set_A_periph(AT91_PIN_PB10, 0);*/ /* RK1 */
at91_set_A_periph(AT91_PIN_PB11, 0); /* RF1 */
/*
* Set PCK1 parent to PLLB and its rate to 12 Mhz.
*/
pllb_clk = clk_get(NULL, "pllb");
pck1_clk = clk_get(NULL, "pck1");
clk_set_parent(pck1_clk, pllb_clk);
clk_set_rate(pck1_clk, 12000000);
DBG("MCLK rate %luHz\n", clk_get_rate(pck1_clk));
/* assign the GPIO pin to PCK1 */
at91_set_B_periph(AT91_PIN_PA24, 0);
#ifdef CONFIG_SND_AT91_SOC_ETI_SLAVE
printk(KERN_INFO "eti_b1_wm8731: Codec in Slave Mode\n");
#else
printk(KERN_INFO "eti_b1_wm8731: Codec in Master Mode\n");
#endif
return ret;
fail_io_unmap:
iounmap(ssc->base);
fail_release_mem:
release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
return ret;
}
static void __exit eti_b1_exit(void)
{
struct at91_ssc_periph *ssc = eti_b1_dai.cpu_dai->private_data;
clk_put(pck1_clk);
clk_put(pllb_clk);
platform_device_unregister(eti_b1_snd_device);
iounmap(ssc->base);
release_mem_region(AT91RM9200_BASE_SSC1, SZ_16K);
}
module_init(eti_b1_init);
module_exit(eti_b1_exit);
/* Module information */
MODULE_AUTHOR("Frank Mandarino <fmandarino@endrelia.com>");
MODULE_DESCRIPTION("ALSA SoC ETI-B1-WM8731");
MODULE_LICENSE("GPL");
......@@ -17,6 +17,22 @@ config SND_BF5XX_SOC_SSM2602
help
Say Y if you want to add support for SoC audio on BF527-EZKIT.
config SND_BF5XX_SOC_AD73311
tristate "SoC AD73311 Audio support for Blackfin"
depends on SND_BF5XX_I2S
select SND_BF5XX_SOC_I2S
select SND_SOC_AD73311
help
Say Y if you want to add support for AD73311 codec on Blackfin.
config SND_BFIN_AD73311_SE
int "PF pin for AD73311L Chip Select"
depends on SND_BF5XX_SOC_AD73311
default 4
help
Enter the GPIO used to control AD73311's SE pin. Acceptable
values are 0 to 7
config SND_BF5XX_AC97
tristate "SoC AC97 Audio for the ADI BF5xx chip"
depends on BLACKFIN && SND_SOC
......
......@@ -14,7 +14,8 @@ obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o
# Blackfin Machine Support
snd-ad1980-objs := bf5xx-ad1980.o
snd-ssm2602-objs := bf5xx-ssm2602.o
snd-ad73311-objs := bf5xx-ad73311.o
obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o
obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o
obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o
......@@ -56,6 +56,7 @@ static void bf5xx_mmap_copy(struct snd_pcm_substream *substream,
sport->tx_pos += runtime->period_size;
if (sport->tx_pos >= runtime->buffer_size)
sport->tx_pos %= runtime->buffer_size;
sport->tx_delay_pos = sport->tx_pos;