[alsa-devel] Alsa-devel Digest, Vol 27, Issue 124
Demian Martin
demianm at attglobal.net
Sun May 24 01:27:11 CEST 2009
Demian Martin
PDS
209 613 6990
-----Original Message-----
From: alsa-devel-request at alsa-project.org
Date: Sun, 24 May 2009 01:13:19
To: <alsa-devel at alsa-project.org>
Subject: Alsa-devel Digest, Vol 27, Issue 124
Send Alsa-devel mailing list submissions to
alsa-devel at alsa-project.org
To subscribe or unsubscribe via the World Wide Web, visit
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
or, via email, send a message with subject or body 'help' to
alsa-devel-request at alsa-project.org
You can reach the person managing the list at
alsa-devel-owner at alsa-project.org
When replying, please edit your Subject line so it is more specific
than "Re: Contents of Alsa-devel digest..."
Today's Topics:
1. [PATCH V2 4/9] Add a few more mpc5200 PSC defines (Jon Smirl)
2. [PATCH V2 6/9] Codec for STAC9766 used on the Efika (Jon Smirl)
3. [PATCH V2 7/9] AC97 driver for mpc5200 (Jon Smirl)
----------------------------------------------------------------------
Message: 1
Date: Sat, 23 May 2009 19:13:03 -0400
From: Jon Smirl <jonsmirl at gmail.com>
Subject: [alsa-devel] [PATCH V2 4/9] Add a few more mpc5200 PSC
defines
To: grant.likely at secretlab.ca, linuxppc-dev at ozlabs.org,
alsa-devel at alsa-project.org, broonie at sirena.org.uk
Message-ID: <20090523231303.17919.35877.stgit at terra>
Content-Type: text/plain; charset="utf-8"
Add a few more mpc5200 PSC defines. More bit fields defines for mpc5200 PSC registers. This patch is going in via Grant's tree.
Signed-off-by: Jon Smirl <jonsmirl at gmail.com>
---
0 files changed, 0 insertions(+), 0 deletions(-)
diff --git a/arch/powerpc/include/asm/mpc52xx_psc.h b/arch/powerpc/include/asm/mpc52xx_psc.h
index a218da6..fb84120 100644
--- a/arch/powerpc/include/asm/mpc52xx_psc.h
+++ b/arch/powerpc/include/asm/mpc52xx_psc.h
@@ -28,6 +28,10 @@
#define MPC52xx_PSC_MAXNUM 6
/* Programmable Serial Controller (PSC) status register bits */
+#define MPC52xx_PSC_SR_UNEX_RX 0x0001
+#define MPC52xx_PSC_SR_DATA_VAL 0x0002
+#define MPC52xx_PSC_SR_DATA_OVR 0x0004
+#define MPC52xx_PSC_SR_CMDSEND 0x0008
#define MPC52xx_PSC_SR_CDE 0x0080
#define MPC52xx_PSC_SR_RXRDY 0x0100
#define MPC52xx_PSC_SR_RXFULL 0x0200
@@ -61,6 +65,12 @@
#define MPC52xx_PSC_RXTX_FIFO_EMPTY 0x0001
/* PSC interrupt status/mask bits */
+#define MPC52xx_PSC_IMR_UNEX_RX_SLOT 0x0001
+#define MPC52xx_PSC_IMR_DATA_VALID 0x0002
+#define MPC52xx_PSC_IMR_DATA_OVR 0x0004
+#define MPC52xx_PSC_IMR_CMD_SEND 0x0008
+#define MPC52xx_PSC_IMR_ERROR 0x0040
+#define MPC52xx_PSC_IMR_DEOF 0x0080
#define MPC52xx_PSC_IMR_TXRDY 0x0100
#define MPC52xx_PSC_IMR_RXRDY 0x0200
#define MPC52xx_PSC_IMR_DB 0x0400
@@ -117,6 +127,7 @@
#define MPC52xx_PSC_SICR_SIM_FIR (0x6 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_24 (0x7 << 24)
#define MPC52xx_PSC_SICR_SIM_CODEC_32 (0xf << 24)
+#define MPC52xx_PSC_SICR_AWR (1 << 30)
#define MPC52xx_PSC_SICR_GENCLK (1 << 23)
#define MPC52xx_PSC_SICR_I2S (1 << 22)
#define MPC52xx_PSC_SICR_CLKPOL (1 << 21)
------------------------------
Message: 2
Date: Sat, 23 May 2009 19:13:07 -0400
From: Jon Smirl <jonsmirl at gmail.com>
Subject: [alsa-devel] [PATCH V2 6/9] Codec for STAC9766 used on the
Efika
To: grant.likely at secretlab.ca, linuxppc-dev at ozlabs.org,
alsa-devel at alsa-project.org, broonie at sirena.org.uk
Message-ID: <20090523231307.17919.5277.stgit at terra>
Content-Type: text/plain; charset="utf-8"
AC97 codec for STAC9766 used on the Efika.
Datasheet: http://www.idt.com/products/getDoc.cfm?docID=13134007
Signed-off-by: Jon Smirl <jonsmirl at gmail.com>
---
0 files changed, 0 insertions(+), 0 deletions(-)
diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index 7f78b65..cb07d9b 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS
select SND_SOC_CS4270 if I2C
select SND_SOC_PCM3008
select SND_SOC_SSM2602 if I2C
+ select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
select SND_SOC_TLV320AIC23 if I2C
select SND_SOC_TLV320AIC26 if SPI_MASTER
select SND_SOC_TLV320AIC3X if I2C
@@ -93,6 +94,9 @@ config SND_SOC_PCM3008
config SND_SOC_SSM2602
tristate
+config SND_SOC_STAC9766
+ tristate
+
config SND_SOC_TLV320AIC23
tristate
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 70c55fa..46c007c 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -7,6 +7,7 @@ snd-soc-cs4270-objs := cs4270.o
snd-soc-l3-objs := l3.o
snd-soc-pcm3008-objs := pcm3008.o
snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-stac9766-objs := stac9766.o
snd-soc-tlv320aic23-objs := tlv320aic23.o
snd-soc-tlv320aic26-objs := tlv320aic26.o
snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -42,6 +43,7 @@ obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o
obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o
obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o
obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o
obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o
obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o
obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
new file mode 100644
index 0000000..7740cd5
--- /dev/null
+++ b/sound/soc/codecs/stac9766.c
@@ -0,0 +1,470 @@
+/*
+ * stac9766.c -- ALSA SoC STAC9766 codec support
+ *
+ * Copyright 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl at gmail.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.
+ *
+ * Features:-
+ *
+ * o Support for AC97 Codec, S/PDIF
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/soc-of-simple.h>
+
+#include "stac9766.h"
+
+#define STAC9766_VERSION "0.10"
+
+/*
+ * STAC9766 register cache
+ */
+static const u16 stac9766_reg[] = {
+ 0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
+ 0x0000, 0x0000, 0x8008, 0x8008, /* e */
+ 0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
+ 0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
+ 0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+ 0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
+ 0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+ 0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
+ 0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
+ 0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
+ 0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
+ 0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+ 0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+ 0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
+};
+
+static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"};
+static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
+static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
+static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
+static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
+static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"};
+static const char *stac9766_boost1[] = {"0dB", "10dB"};
+static const char *stac9766_boost2[] = {"0dB", "20dB"};
+static const char *stac9766_stereo_mic[] = {"Off", "On"};
+
+static const struct soc_enum stac9766_record_enum =
+ SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
+static const struct soc_enum stac9766_mono_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
+static const struct soc_enum stac9766_mic_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
+static const struct soc_enum stac9766_SPDIF_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
+static const struct soc_enum stac9766_popbypass_enum =
+ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
+static const struct soc_enum stac9766_record_all_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux);
+static const struct soc_enum stac9766_boost1_enum =
+ SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
+static const struct soc_enum stac9766_boost2_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
+static const struct soc_enum stac9766_stereo_mic_enum =
+ SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
+
+static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
+static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
+static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
+static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
+
+static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
+ SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
+ SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
+ SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1, master_tlv),
+ SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
+ SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1, master_tlv),
+ SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+ SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
+ SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
+
+
+ SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
+ SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
+ SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
+ SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
+
+ SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
+ SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
+ SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+ SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
+
+ SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
+ SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
+ SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
+ SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
+
+ SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
+ SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
+ SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
+ SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
+ SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
+
+ SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
+ SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
+ SOC_ENUM("Record All Mux", stac9766_record_all_enum),
+ SOC_ENUM("Record Mux", stac9766_record_enum),
+ SOC_ENUM("Mono Mux", stac9766_mono_enum),
+ SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
+};
+
+int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+ unsigned int val)
+{
+ u16 *cache = codec->reg_cache;
+
+ if (reg > AC97_STAC_PAGE0) {
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+ return 0;
+ }
+ if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+ return -EIO;
+
+ soc_ac97_ops.write(codec->ac97, reg, val);
+ cache[reg / 2] = val;
+ return 0;
+}
+
+unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg)
+{
+ u16 val = 0, *cache = codec->reg_cache;
+
+ if (reg > AC97_STAC_PAGE0) {
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+ val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
+ stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+ return val;
+ }
+ if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+ return -EIO;
+
+ if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+ reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
+ reg == AC97_VENDOR_ID2) {
+
+ val = soc_ac97_ops.read(codec->ac97, reg);
+ return val;
+ }
+ return cache[reg / 2];
+}
+
+static int ac97_analog_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned short reg, vra;
+
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+
+ vra |= 0x1; /* enable variable rate audio */
+
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ reg = AC97_PCM_FRONT_DAC_RATE;
+ else
+ reg = AC97_PCM_LR_ADC_RATE;
+
+ return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_prepare(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ unsigned short reg, vra;
+
+ stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
+
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+ vra |= 0x5; /* Enable VRA and SPDIF out */
+
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+ reg = AC97_PCM_FRONT_DAC_RATE;
+
+ return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_trigger(struct snd_pcm_substream *substream,
+ int cmd, struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ unsigned short vra;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+ vra &= !0x04;
+ stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+ break;
+ }
+ return 0;
+}
+
+static int stac9766_set_bias_level(struct snd_soc_codec *codec,
+ enum snd_soc_bias_level level)
+{
+ switch (level) {
+ case SND_SOC_BIAS_ON: /* full On */
+ case SND_SOC_BIAS_PREPARE: /* partial On */
+ case SND_SOC_BIAS_STANDBY: /* Off, with power */
+ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000);
+ break;
+ case SND_SOC_BIAS_OFF: /* Off, without power */
+ /* disable everything including AC link */
+ stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff);
+ break;
+ }
+ codec->bias_level = level;
+ return 0;
+}
+
+int stac9766_reset(struct snd_soc_codec *codec, int try_warm)
+{
+ if (try_warm && soc_ac97_ops.warm_reset) {
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (stac9766_ac97_read(codec, 0) == stac9766_reg[0])
+ return 1;
+ }
+
+ soc_ac97_ops.reset(codec->ac97);
+ if (soc_ac97_ops.warm_reset)
+ soc_ac97_ops.warm_reset(codec->ac97);
+ if (stac9766_ac97_read(codec, 0) != stac9766_reg[0])
+ return -EIO;
+ return 0;
+}
+
+static int stac9766_codec_suspend(struct platform_device *pdev,
+ pm_message_t state)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF);
+ return 0;
+}
+
+static int stac9766_codec_resume(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+ u16 id, reset;
+
+ reset = 0;
+ /* give the codec an AC97 warm reset to start the link */
+reset:
+ if (reset > 5) {
+ printk(KERN_ERR "stac9766 failed to resume");
+ return -EIO;
+ }
+ codec->ac97->bus->ops->warm_reset(codec->ac97);
+ id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2);
+ if (id != 0x4c13) {
+ stac9766_reset(codec, 0);
+ reset++;
+ goto reset;
+ }
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ if (codec->suspend_bias_level == SND_SOC_BIAS_ON)
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_ON);
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops stac9766_dai_ops_analog =
+{
+ .prepare = ac97_analog_prepare,
+};
+
+static struct snd_soc_dai_ops stac9766_dai_ops_digital =
+{
+ .prepare = ac97_digital_prepare,
+ .trigger = ac97_digital_trigger,
+};
+
+struct snd_soc_dai stac9766_dai[] = {
+{
+ .name = "stac9766 analog",
+ .id = 0,
+ .ac97_control = 1,
+
+ /* stream cababilities */
+ .playback = {
+ .stream_name = "stac9766 analog",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SND_SOC_STD_AC97_FMTS,
+ },
+ .capture = {
+ .stream_name = "stac9766 analog",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SND_SOC_STD_AC97_FMTS,
+ },
+ /* alsa ops */
+ .ops = &stac9766_dai_ops_analog,
+},
+{
+ .name = "stac9766 IEC958",
+ .id = 1,
+ .ac97_control = 1,
+
+ /* stream cababilities */
+ .playback = {
+ .stream_name = "stac9766 IEC958",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FORMAT_IEC958_SUBFRAME_BE,
+ },
+ /* alsa ops */
+ .ops = &stac9766_dai_ops_digital,
+}};
+EXPORT_SYMBOL_GPL(stac9766_dai);
+
+static int stac9766_codec_probe(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec;
+ int ret = 0;
+
+ printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION);
+
+ socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL);
+ if (socdev->card->codec == NULL)
+ return -ENOMEM;
+ codec = socdev->card->codec;
+ mutex_init(&codec->mutex);
+
+ codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL);
+ if (codec->reg_cache == NULL) {
+ ret = -ENOMEM;
+ goto cache_err;
+ }
+ codec->reg_cache_size = sizeof(stac9766_reg);
+ codec->reg_cache_step = 2;
+
+ codec->name = "STAC9766";
+ codec->owner = THIS_MODULE;
+ codec->dai = stac9766_dai;
+ codec->num_dai = ARRAY_SIZE(stac9766_dai);
+ codec->write = stac9766_ac97_write;
+ codec->read = stac9766_ac97_read;
+ codec->set_bias_level = stac9766_set_bias_level;
+ INIT_LIST_HEAD(&codec->dapm_widgets);
+ INIT_LIST_HEAD(&codec->dapm_paths);
+
+ ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0);
+ if (ret < 0)
+ goto codec_err;
+
+ /* register pcms */
+ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1);
+ if (ret < 0)
+ goto pcm_err;
+
+ /* do a cold reset for the controller and then try
+ * a warm reset followed by an optional cold reset for codec */
+ stac9766_reset(codec, 0);
+ ret = stac9766_reset(codec, 1);
+ if (ret < 0) {
+ printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n");
+ goto reset_err;
+ }
+
+ stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY);
+
+ snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(
+ stac9766_snd_ac97_controls));
+
+ ret = snd_soc_init_card(socdev);
+ if (ret < 0)
+ goto reset_err;
+ return 0;
+
+reset_err:
+ snd_soc_free_pcms(socdev);
+pcm_err:
+ snd_soc_free_ac97_codec(codec);
+codec_err:
+ kfree(codec->private_data);
+cache_err:
+ kfree(socdev->card->codec);
+ socdev->card->codec = NULL;
+ return ret;
+}
+
+static int stac9766_codec_remove(struct platform_device *pdev)
+{
+ struct snd_soc_device *socdev = platform_get_drvdata(pdev);
+ struct snd_soc_codec *codec = socdev->card->codec;
+
+ if (codec == NULL)
+ return 0;
+
+ snd_soc_free_pcms(socdev);
+ snd_soc_free_ac97_codec(codec);
+ kfree(codec->reg_cache);
+ kfree(codec);
+ return 0;
+}
+
+struct snd_soc_codec_device soc_codec_dev_stac9766 =
+{
+ .probe = stac9766_codec_probe,
+ .remove = stac9766_codec_remove,
+ .suspend = stac9766_codec_suspend,
+ .resume = stac9766_codec_resume,
+};
+EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766);
+
+static int __init stac9766_modinit(void)
+{
+ return snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai));
+}
+module_init(stac9766_modinit);
+
+static void __exit stac9766_exit(void)
+{
+ snd_soc_unregister_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai));
+}
+module_exit(stac9766_exit);
+
+MODULE_DESCRIPTION("ASoC stac9766 driver");
+MODULE_AUTHOR("Jon Smirl <jonsmirl at gmail.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/stac9766.h b/sound/soc/codecs/stac9766.h
new file mode 100644
index 0000000..65642eb
--- /dev/null
+++ b/sound/soc/codecs/stac9766.h
@@ -0,0 +1,21 @@
+/*
+ * stac9766.h -- STAC9766 Soc Audio driver
+ */
+
+#ifndef _STAC9766_H
+#define _STAC9766_H
+
+#define AC97_STAC_PAGE0 0x1000
+#define AC97_STAC_DA_CONTROL (AC97_STAC_PAGE0 | 0x6A)
+#define AC97_STAC_ANALOG_SPECIAL (AC97_STAC_PAGE0 | 0x6E)
+#define AC97_STAC_STEREO_MIC 0x78
+
+/* STAC9766 DAI ID's */
+#define STAC9766_DAI_AC97_ANALOG 0
+#define STAC9766_DAI_AC97_DIGITAL 1
+
+extern struct snd_soc_dai stac9766_dai[];
+extern struct snd_soc_codec_device soc_codec_dev_stac9766;
+
+
+#endif
------------------------------
Message: 3
Date: Sat, 23 May 2009 19:13:09 -0400
From: Jon Smirl <jonsmirl at gmail.com>
Subject: [alsa-devel] [PATCH V2 7/9] AC97 driver for mpc5200
To: grant.likely at secretlab.ca, linuxppc-dev at ozlabs.org,
alsa-devel at alsa-project.org, broonie at sirena.org.uk
Message-ID: <20090523231309.17919.83073.stgit at terra>
Content-Type: text/plain; charset="utf-8"
AC97 driver for mpc5200
Signed-off-by: Jon Smirl <jonsmirl at gmail.com>
---
sound/soc/fsl/Kconfig | 11 +
sound/soc/fsl/Makefile | 1
sound/soc/fsl/mpc5200_psc_ac97.c | 394 ++++++++++++++++++++++++++++++++++++++
sound/soc/fsl/mpc5200_psc_ac97.h | 15 +
4 files changed, 421 insertions(+), 0 deletions(-)
create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.c
create mode 100644 sound/soc/fsl/mpc5200_psc_ac97.h
diff --git a/sound/soc/fsl/Kconfig b/sound/soc/fsl/Kconfig
index 1918c78..3bce952 100644
--- a/sound/soc/fsl/Kconfig
+++ b/sound/soc/fsl/Kconfig
@@ -29,3 +29,14 @@ config SND_SOC_MPC5200_I2S
select PPC_BESTCOMM_GEN_BD
help
Say Y here to support the MPC5200 PSCs in I2S mode.
+
+config SND_SOC_MPC5200_AC97
+ tristate "Freescale MPC5200 PSC in AC97 mode driver"
+ depends on PPC_MPC52xx && PPC_BESTCOMM
+ select AC97_BUS
+ select SND_MPC52xx_DMA
+ select PPC_BESTCOMM_GEN_BD
+ help
+ Say Y here to support the MPC5200 PSCs in AC97 mode.
+
+
diff --git a/sound/soc/fsl/Makefile b/sound/soc/fsl/Makefile
index 7731ef2..14631a1 100644
--- a/sound/soc/fsl/Makefile
+++ b/sound/soc/fsl/Makefile
@@ -13,4 +13,5 @@ obj-$(CONFIG_SND_SOC_MPC8610) += snd-soc-fsl-ssi.o snd-soc-fsl-dma.o
# MPC5200 Platform Support
obj-$(CONFIG_SND_MPC52xx_DMA) += mpc5200_dma.o
obj-$(CONFIG_SND_SOC_MPC5200_I2S) += mpc5200_psc_i2s.o
+obj-$(CONFIG_SND_SOC_MPC5200_AC97) += mpc5200_psc_ac97.o
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c
new file mode 100644
index 0000000..fa1bb9a
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.c
@@ -0,0 +1,394 @@
+/*
+ * linux/sound/mpc5200-ac97.c -- AC97 support for the Freescale MPC52xx chip.
+ *
+ * Copyright (C) 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl at gmail.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.
+ */
+
+#include <linux/module.h>
+#include <linux/of_device.h>
+#include <linux/of_platform.h>
+
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include <asm/mpc52xx_psc.h>
+
+#include "mpc5200_dma.h"
+#include "mpc5200_psc_ac97.h"
+
+#define DRV_NAME "mpc5200-psc-ac97"
+
+/* ALSA only supports a single AC97 device so static is recommend here */
+static struct psc_dma *psc_dma;
+
+static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+ int timeout;
+ unsigned int val;
+
+ spin_lock(&psc_dma->lock);
+
+ /* Wait for it to be ready */
+ timeout = 1000;
+ while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_CMDSEND) )
+ udelay(10);
+
+ if (!timeout) {
+ pr_err("timeout on ac97 bus (rdy)\n");
+ return 0xffff;
+ }
+
+ /* Do the read */
+ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24));
+
+ /* Wait for the answer */
+ timeout = 1000;
+ while ((--timeout) && !(in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_DATA_VAL) )
+ udelay(10);
+
+ if (!timeout) {
+ pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status));
+ return 0xffff;
+ }
+
+ /* Get the data */
+ val = in_be32(&psc_dma->psc_regs->ac97_data);
+ if ( ((val>>24) & 0x7f) != reg ) {
+ pr_err("reg echo error on ac97 read\n");
+ return 0xffff;
+ }
+ val = (val >> 8) & 0xffff;
+
+ spin_unlock(&psc_dma->lock);
+ return (unsigned short) val;
+}
+
+static void psc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
+{
+ int timeout;
+
+ spin_lock(&psc_dma->lock);
+
+ /* Wait for it to be ready */
+ timeout = 1000;
+ while ((--timeout) && (in_be16(&psc_dma->psc_regs->sr_csr.status) &
+ MPC52xx_PSC_SR_CMDSEND) )
+ udelay(10);
+
+ if (!timeout) {
+ pr_err("timeout on ac97 write\n");
+ return;
+ }
+
+ /* Write data */
+ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8));
+
+ spin_unlock(&psc_dma->lock);
+}
+
+static void psc_ac97_cold_reset(struct snd_ac97 *ac97)
+{
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ /* Do a cold reset */
+ out_8(®s->op1, MPC52xx_PSC_OP_RES);
+ udelay(10);
+ out_8(®s->op0, MPC52xx_PSC_OP_RES);
+ udelay(50);
+
+ /* PSC recover from cold reset (cfr user manual, not sure if useful) */
+ out_be32(®s->sicr, in_be32(®s->sicr));
+}
+
+static void psc_ac97_warm_reset(struct snd_ac97 *ac97)
+{
+ struct mpc52xx_psc __iomem *regs = psc_dma->psc_regs;
+
+ out_be32(®s->sicr, psc_dma->sicr | MPC52xx_PSC_SICR_AWR);
+ udelay(3);
+ out_be32(®s->sicr, psc_dma->sicr);
+}
+
+struct snd_ac97_bus_ops soc_ac97_ops = {
+ .read = psc_ac97_read,
+ .write = psc_ac97_write,
+ .reset = psc_ac97_cold_reset,
+ .warm_reset = psc_ac97_warm_reset,
+};
+EXPORT_SYMBOL_GPL(soc_ac97_ops);
+
+#ifdef CONFIG_PM
+static int psc_ac97_suspend(struct snd_soc_dai *dai)
+{
+ return 0;
+}
+
+static int psc_ac97_resume(struct snd_soc_dai *dai)
+{
+ return 0;
+}
+
+#else
+#define psc_ac97_suspend NULL
+#define psc_ac97_resume NULL
+#endif
+
+static int psc_ac97_hw_analog_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+
+ dev_dbg(psc_dma->dev, "%s(substream=%p) p_size=%i p_bytes=%i"
+ " periods=%i buffer_size=%i buffer_bytes=%i channels=%i"
+ " rate=%i format=%i\n",
+ __func__, substream, params_period_size(params),
+ params_period_bytes(params), params_periods(params),
+ params_buffer_size(params), params_buffer_bytes(params),
+ params_channels(params), params_rate(params), params_format(params));
+
+
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (params_channels(params) == 1)
+ psc_dma->slots |= 0x00000100;
+ else
+ psc_dma->slots |= 0x00000300;
+ } else {
+ if (params_channels(params) == 1)
+ psc_dma->slots |= 0x01000000;
+ else
+ psc_dma->slots |= 0x03000000;
+ }
+
+ spin_lock(&psc_dma->lock);
+ out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+ spin_unlock(&psc_dma->lock);
+
+ return 0;
+}
+
+static int psc_ac97_hw_digital_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+
+ spin_lock(&psc_dma->lock);
+ if (params_channels(params) == 1)
+ out_be32(&psc_dma->psc_regs->ac97_slots, 0x01000000);
+ else
+ out_be32(&psc_dma->psc_regs->ac97_slots, 0x03000000);
+ spin_unlock(&psc_dma->lock);
+
+ return 0;
+}
+
+static int psc_ac97_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct psc_dma *psc_dma = rtd->dai->cpu_dai->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_STOP:
+ if (substream->pstr->stream == SNDRV_PCM_STREAM_CAPTURE)
+ psc_dma->slots &= 0xFFFF0000;
+ else
+ psc_dma->slots &= 0x0000FFFF;
+
+ spin_lock(&psc_dma->lock);
+ out_be32(&psc_dma->psc_regs->ac97_slots, psc_dma->slots);
+ spin_unlock(&psc_dma->lock);
+ break;
+ }
+ return 0;
+}
+
+/* ---------------------------------------------------------------------
+ * ALSA SoC Bindings
+ *
+ * - Digital Audio Interface (DAI) template
+ * - create/destroy dai hooks
+ */
+
+/**
+ * psc_ac97_dai_template: template CPU Digital Audio Interface
+ */
+static struct snd_soc_dai_ops psc_ac97_analog_ops = {
+ .hw_params = psc_ac97_hw_analog_params,
+ .trigger = psc_ac97_trigger,
+};
+
+static struct snd_soc_dai_ops psc_ac97_digital_ops = {
+ .hw_params = psc_ac97_hw_digital_params,
+};
+
+struct snd_soc_dai psc_ac97_dai[] = {
+{
+ .name = "AC97",
+ .suspend = psc_ac97_suspend,
+ .resume = psc_ac97_resume,
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 6,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+ },
+ .capture = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S32_BE,
+ },
+ .ops = &psc_ac97_analog_ops,
+},
+{
+ .name = "SPDIF",
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_32000 | \
+ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
+ .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_BE,
+ },
+ .ops = &psc_ac97_digital_ops,
+}};
+EXPORT_SYMBOL_GPL(psc_ac97_dai);
+
+
+
+/* ---------------------------------------------------------------------
+ * OF platform bus binding code:
+ * - Probe/remove operations
+ * - OF device match table
+ */
+static int __devinit psc_ac97_of_probe(struct of_device *op,
+ const struct of_device_id *match)
+{
+ int rc, i, id1, id2, timeout, max_reset;
+ struct snd_ac97 ac97;
+ struct mpc52xx_psc __iomem *regs;
+
+ rc = mpc5200_audio_dma_create(op);
+ if (rc != 0)
+ return rc;
+
+ for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+ psc_ac97_dai[i].dev = &op->dev;
+
+ rc = snd_soc_register_dais(psc_ac97_dai, ARRAY_SIZE(psc_ac97_dai));
+ if (rc != 0) {
+ pr_err("Failed to register DAI\n");
+ return 0;
+ }
+
+ psc_dma = dev_get_drvdata(&op->dev);
+ regs = psc_dma->psc_regs;
+ ac97.private_data = psc_dma;
+
+ for (i = 0; i < ARRAY_SIZE(psc_ac97_dai); i++)
+ psc_ac97_dai[i].private_data = psc_dma;
+
+ psc_dma->imr = 0;
+ out_be16(&psc_dma->psc_regs->isr_imr.imr, psc_dma->imr);
+
+ /* Configure the serial interface mode to AC97 */
+ psc_dma->sicr = MPC52xx_PSC_SICR_SIM_AC97 | MPC52xx_PSC_SICR_ENAC97;
+ out_be32(®s->sicr, psc_dma->sicr);
+
+ /* No slots active */
+ out_be32(®s->ac97_slots, 0x00000000);
+
+ /* AC97 clock is generated by the codec.
+ * Ensure that it starts ticking after codec reset.
+ */
+ max_reset = 0;
+reset:
+ if (max_reset++ > 5) {
+ dev_err(&op->dev, "AC97 codec failed to reset\n");
+ mpc5200_audio_dma_destroy(op);
+ return -ENODEV;
+ }
+
+ psc_ac97_cold_reset(&ac97);
+ psc_ac97_warm_reset(&ac97);
+
+ /* first make sure it is low */
+ timeout = 0;
+ while ((in_8(®s->ipcr_acr.ipcr) & 0x80) != 0) {
+ udelay(1);
+ if (timeout++ > 1000)
+ goto reset;
+ }
+ /* then wait for the transition to high */
+ timeout = 0;
+ while ((in_8(®s->ipcr_acr.ipcr) & 0x80) == 0) {
+ udelay(1);
+ if (timeout++ > 1000)
+ psc_ac97_warm_reset(&ac97);
+ }
+
+ /* Go */
+ out_8(®s->command, MPC52xx_PSC_TX_ENABLE | MPC52xx_PSC_RX_ENABLE);
+
+ id1 = psc_ac97_read(&ac97, AC97_VENDOR_ID1);
+ id2 = psc_ac97_read(&ac97, AC97_VENDOR_ID2);
+
+ dev_info(&op->dev, "Codec ID is %04x %04x\n", id1, id2);
+
+ return 0;
+}
+
+static int __devexit psc_ac97_of_remove(struct of_device *op)
+{
+ return mpc5200_audio_dma_destroy(op);
+}
+
+/* Match table for of_platform binding */
+static struct of_device_id psc_ac97_match[] __devinitdata = {
+ { .compatible = "fsl,mpc5200-psc-ac97", },
+ { .compatible = "fsl,mpc5200b-psc-ac97", },
+ {}
+};
+MODULE_DEVICE_TABLE(of, psc_ac97_match);
+
+static struct of_platform_driver psc_ac97_driver = {
+ .match_table = psc_ac97_match,
+ .probe = psc_ac97_of_probe,
+ .remove = __devexit_p(psc_ac97_of_remove),
+ .driver = {
+ .name = "mpc5200-psc-ac97",
+ .owner = THIS_MODULE,
+ },
+};
+
+/* ---------------------------------------------------------------------
+ * Module setup and teardown; simply register the of_platform driver
+ * for the PSC in AC97 mode.
+ */
+static int __init psc_ac97_init(void)
+{
+ return of_register_platform_driver(&psc_ac97_driver);
+}
+module_init(psc_ac97_init);
+
+static void __exit psc_ac97_exit(void)
+{
+ of_unregister_platform_driver(&psc_ac97_driver);
+}
+module_exit(psc_ac97_exit);
+
+MODULE_AUTHOR("Jon Smirl <jonsmirl at gmail.com>");
+MODULE_DESCRIPTION("mpc5200 AC97 module");
+MODULE_LICENSE("GPL");
+
diff --git a/sound/soc/fsl/mpc5200_psc_ac97.h b/sound/soc/fsl/mpc5200_psc_ac97.h
new file mode 100644
index 0000000..4bc18c3
--- /dev/null
+++ b/sound/soc/fsl/mpc5200_psc_ac97.h
@@ -0,0 +1,15 @@
+/*
+ * Freescale MPC5200 PSC in AC97 mode
+ * ALSA SoC Digital Audio Interface (DAI) driver
+ *
+ */
+
+#ifndef __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+#define __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__
+
+extern struct snd_soc_dai psc_ac97_dai[];
+
+#define MPC5200_AC97_NORMAL 0
+#define MPC5200_AC97_SPDIF 1
+
+#endif /* __SOUND_SOC_FSL_MPC52xx_PSC_AC97_H__ */
------------------------------
_______________________________________________
Alsa-devel mailing list
Alsa-devel at alsa-project.org
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
End of Alsa-devel Digest, Vol 27, Issue 124
*******************************************
More information about the Alsa-devel
mailing list