| lists.openwall.net | lists / announce owl-users owl-dev john-users john-dev passwdqc-users yescrypt popa3d-users / oss-security kernel-hardening musl sabotage tlsify passwords / crypt-dev xvendor / Bugtraq Full-Disclosure linux-kernel linux-netdev linux-ext4 linux-hardening linux-cve-announce PHC | |
|
Open Source and information security mailing list archives
| ||
|
Date: Mon, 30 Mar 2015 20:16:24 -0700
From: Scott Branden <sbranden@...adcom.com>
To: Rob Herring <robh+dt@...nel.org>, Pawel Moll <pawel.moll@....com>,
"Mark Rutland" <mark.rutland@....com>,
Ian Campbell <ijc+devicetree@...lion.org.uk>,
Kumar Gala <galak@...eaurora.org>,
"Liam Girdwood" <lgirdwood@...il.com>,
Mark Brown <broonie@...nel.org>,
"Jaroslav Kysela" <perex@...ex.cz>, Takashi Iwai <tiwai@...e.de>,
<alsa-devel@...a-project.org>
CC: <linux-kernel@...r.kernel.org>, <devicetree@...r.kernel.org>,
<linux-arm-kernel@...ts.infradead.org>,
<bcm-kernel-feedback-list@...adcom.com>,
<linux-rpi-kernel@...ts.infradead.org>,
Dmitry Torokhov <dtor@...gle.com>,
Anatol Pomazao <anatol@...gle.com>, <abrestic@...gle.com>,
<bryeung@...gle.com>, <olofj@...gle.com>, <pwestin@...gle.com>,
Lori Hikichi <lhikichi@...adcom.com>,
Scott Branden <sbranden@...adcom.com>
Subject: [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
From: Lori Hikichi <lhikichi@...adcom.com>
The audio block has 4 serial ports. 3 ports are configurable as
either I2S or TDM. The 4th port is for SPDIF transmit.
This audio block is found on the bcm958305, bcm958300, and bcm911360.
Reviewed-by: Jonathan Richardson <jonathar@...adcom.com>
Signed-off-by: Lori Hikichi <lhikichi@...adcom.com>
Signed-off-by: Scott Branden <sbranden@...adcom.com>
---
sound/soc/bcm/Kconfig | 11 +
sound/soc/bcm/Makefile | 5 +-
sound/soc/bcm/cygnus-pcm.c | 918 +++++++++++++++++++++++++
sound/soc/bcm/cygnus-pcm.h | 45 ++
sound/soc/bcm/cygnus-ssp.c | 1613 ++++++++++++++++++++++++++++++++++++++++++++
sound/soc/bcm/cygnus-ssp.h | 84 +++
6 files changed, 2675 insertions(+), 1 deletion(-)
create mode 100644 sound/soc/bcm/cygnus-pcm.c
create mode 100644 sound/soc/bcm/cygnus-pcm.h
create mode 100644 sound/soc/bcm/cygnus-ssp.c
create mode 100644 sound/soc/bcm/cygnus-ssp.h
diff --git a/sound/soc/bcm/Kconfig b/sound/soc/bcm/Kconfig
index 6a834e1..2c5ff37 100644
--- a/sound/soc/bcm/Kconfig
+++ b/sound/soc/bcm/Kconfig
@@ -7,3 +7,14 @@ config SND_BCM2835_SOC_I2S
Say Y or M if you want to add support for codecs attached to
the BCM2835 I2S interface. You will also need
to select the audio interfaces to support below.
+
+config SND_SOC_CYGNUS
+ tristate "SoC platform audio for Broadcom Cygnus chips"
+ depends on ARCH_BCM_CYGNUS || COMPILE_TEST
+ default ARCH_BCM_CYGNUS
+ help
+ Say Y if you want to add support for ASoC audio on Broadcom
+ Cygnus chips (bcm958300, bcm958305, bcm911360)
+
+ If you don't know what to do here, say N.
+
diff --git a/sound/soc/bcm/Makefile b/sound/soc/bcm/Makefile
index bc816b7..5c39790 100644
--- a/sound/soc/bcm/Makefile
+++ b/sound/soc/bcm/Makefile
@@ -1,5 +1,8 @@
# BCM2835 Platform Support
snd-soc-bcm2835-i2s-objs := bcm2835-i2s.o
-obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o
+# CYGNUS Platform Support
+snd-soc-cygnus-objs := cygnus-pcm.o cygnus-ssp.o
+obj-$(CONFIG_SND_BCM2835_SOC_I2S) += snd-soc-bcm2835-i2s.o
+obj-$(CONFIG_SND_SOC_CYGNUS) += snd-soc-cygnus.o
diff --git a/sound/soc/bcm/cygnus-pcm.c b/sound/soc/bcm/cygnus-pcm.c
new file mode 100644
index 0000000..3a4106b
--- /dev/null
+++ b/sound/soc/bcm/cygnus-pcm.c
@@ -0,0 +1,918 @@
+/*
+ * Copyright (C) 2014-2015 Broadcom Corporation
+ *
+ * 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.
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/dma-mapping.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+#include <linux/debugfs.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <linux/io.h>
+#include <linux/timer.h>
+
+#include "cygnus-ssp.h"
+#include "cygnus-pcm.h"
+
+/* Register offset needed for ASoC PCM module */
+
+#define INTH_R5F_STATUS_OFFSET 0x040
+#define INTH_R5F_CLEAR_OFFSET 0x048
+#define INTH_R5F_MASK_SET_OFFSET 0x050
+#define INTH_R5F_MASK_CLEAR_OFFSET 0x054
+
+#define BF_REARM_FREE_MARK_OFFSET 0x344
+#define BF_REARM_FULL_MARK_OFFSET 0x348
+
+/* Ring Buffer Ctrl Regs --- Start */
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_RDADDR_REG_BASE */
+#define SRC_RBUF_0_RDADDR_OFFSET 0x500
+#define SRC_RBUF_1_RDADDR_OFFSET 0x518
+#define SRC_RBUF_2_RDADDR_OFFSET 0x530
+#define SRC_RBUF_3_RDADDR_OFFSET 0x548
+#define SRC_RBUF_4_RDADDR_OFFSET 0x560
+#define SRC_RBUF_5_RDADDR_OFFSET 0x578
+#define SRC_RBUF_6_RDADDR_OFFSET 0x590
+
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_WRADDR_REG_BASE */
+#define SRC_RBUF_0_WRADDR_OFFSET 0x504
+#define SRC_RBUF_1_WRADDR_OFFSET 0x51c
+#define SRC_RBUF_2_WRADDR_OFFSET 0x534
+#define SRC_RBUF_3_WRADDR_OFFSET 0x54c
+#define SRC_RBUF_4_WRADDR_OFFSET 0x564
+#define SRC_RBUF_5_WRADDR_OFFSET 0x57c
+#define SRC_RBUF_6_WRADDR_OFFSET 0x594
+
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_BASEADDR_REG_BASE */
+#define SRC_RBUF_0_BASEADDR_OFFSET 0x508
+#define SRC_RBUF_1_BASEADDR_OFFSET 0x520
+#define SRC_RBUF_2_BASEADDR_OFFSET 0x538
+#define SRC_RBUF_3_BASEADDR_OFFSET 0x550
+#define SRC_RBUF_4_BASEADDR_OFFSET 0x568
+#define SRC_RBUF_5_BASEADDR_OFFSET 0x580
+#define SRC_RBUF_6_BASEADDR_OFFSET 0x598
+
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_ENDADDR_REG_BASE */
+#define SRC_RBUF_0_ENDADDR_OFFSET 0x50c
+#define SRC_RBUF_1_ENDADDR_OFFSET 0x524
+#define SRC_RBUF_2_ENDADDR_OFFSET 0x53c
+#define SRC_RBUF_3_ENDADDR_OFFSET 0x554
+#define SRC_RBUF_4_ENDADDR_OFFSET 0x56c
+#define SRC_RBUF_5_ENDADDR_OFFSET 0x584
+#define SRC_RBUF_6_ENDADDR_OFFSET 0x59c
+
+/* AUD_FMM_BF_CTRL_SOURCECH_RINGBUF_X_FREE_MARK_REG_BASE */
+#define SRC_RBUF_0_FREE_MARK_OFFSET 0x510
+#define SRC_RBUF_1_FREE_MARK_OFFSET 0x528
+#define SRC_RBUF_2_FREE_MARK_OFFSET 0x540
+#define SRC_RBUF_3_FREE_MARK_OFFSET 0x558
+#define SRC_RBUF_4_FREE_MARK_OFFSET 0x570
+#define SRC_RBUF_5_FREE_MARK_OFFSET 0x588
+#define SRC_RBUF_6_FREE_MARK_OFFSET 0x5a0
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_RDADDR_REG_BASE */
+#define DST_RBUF_0_RDADDR_OFFSET 0x5c0
+#define DST_RBUF_1_RDADDR_OFFSET 0x5d8
+#define DST_RBUF_2_RDADDR_OFFSET 0x5f0
+#define DST_RBUF_3_RDADDR_OFFSET 0x608
+#define DST_RBUF_4_RDADDR_OFFSET 0x620
+#define DST_RBUF_5_RDADDR_OFFSET 0x638
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_WRADDR_REG_BASE */
+#define DST_RBUF_0_WRADDR_OFFSET 0x5c4
+#define DST_RBUF_1_WRADDR_OFFSET 0x5dc
+#define DST_RBUF_2_WRADDR_OFFSET 0x5f4
+#define DST_RBUF_3_WRADDR_OFFSET 0x60c
+#define DST_RBUF_4_WRADDR_OFFSET 0x624
+#define DST_RBUF_5_WRADDR_OFFSET 0x63c
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_BASEADDR_REG_BASE */
+#define DST_RBUF_0_BASEADDR_OFFSET 0x5c8
+#define DST_RBUF_1_BASEADDR_OFFSET 0x5e0
+#define DST_RBUF_2_BASEADDR_OFFSET 0x5f8
+#define DST_RBUF_3_BASEADDR_OFFSET 0x610
+#define DST_RBUF_4_BASEADDR_OFFSET 0x628
+#define DST_RBUF_5_BASEADDR_OFFSET 0x640
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_ENDADDR_REG_BASE */
+#define DST_RBUF_0_ENDADDR_OFFSET 0x5cc
+#define DST_RBUF_1_ENDADDR_OFFSET 0x5e4
+#define DST_RBUF_2_ENDADDR_OFFSET 0x5fc
+#define DST_RBUF_3_ENDADDR_OFFSET 0x614
+#define DST_RBUF_4_ENDADDR_OFFSET 0x62c
+#define DST_RBUF_5_ENDADDR_OFFSET 0x644
+
+/* AUD_FMM_BF_CTRL_DESTCH_RINGBUF_X_FULL_MARK_REG_BASE */
+#define DST_RBUF_0_FULL_MARK_OFFSET 0x5d0
+#define DST_RBUF_1_FULL_MARK_OFFSET 0x5e8
+#define DST_RBUF_2_FULL_MARK_OFFSET 0x600
+#define DST_RBUF_3_FULL_MARK_OFFSET 0x618
+#define DST_RBUF_4_FULL_MARK_OFFSET 0x630
+#define DST_RBUF_5_FULL_MARK_OFFSET 0x648
+/* Ring Buffer Ctrl Regs --- End */
+
+/* Error Status Regs --- Start */
+/* AUD_FMM_BF_ESR_ESRX_STATUS_REG_BASE */
+#define ESR0_STATUS_OFFSET 0x900
+#define ESR1_STATUS_OFFSET 0x918
+#define ESR2_STATUS_OFFSET 0x930
+#define ESR3_STATUS_OFFSET 0x948
+#define ESR4_STATUS_OFFSET 0x960
+
+/* AUD_FMM_BF_ESR_ESRX_STATUS_CLEAR_REG_BASE */
+#define ESR0_STATUS_CLR_OFFSET 0x908
+#define ESR1_STATUS_CLR_OFFSET 0x920
+#define ESR2_STATUS_CLR_OFFSET 0x938
+#define ESR3_STATUS_CLR_OFFSET 0x950
+#define ESR4_STATUS_CLR_OFFSET 0x968
+
+/* AUD_FMM_BF_ESR_ESRX_MASK_REG_BASE */
+#define ESR0_MASK_STATUS_OFFSET 0x90c
+#define ESR1_MASK_STATUS_OFFSET 0x924
+#define ESR2_MASK_STATUS_OFFSET 0x93c
+#define ESR3_MASK_STATUS_OFFSET 0x954
+#define ESR4_MASK_STATUS_OFFSET 0x96c
+
+/* AUD_FMM_BF_ESR_ESRX_MASK_SET_REG_BASE */
+#define ESR0_MASK_SET_OFFSET 0x910
+#define ESR1_MASK_SET_OFFSET 0x928
+#define ESR2_MASK_SET_OFFSET 0x940
+#define ESR3_MASK_SET_OFFSET 0x958
+#define ESR4_MASK_SET_OFFSET 0x970
+
+/* AUD_FMM_BF_ESR_ESRX_MASK_CLEAR_REG_BASE */
+#define ESR0_MASK_CLR_OFFSET 0x914
+#define ESR1_MASK_CLR_OFFSET 0x92c
+#define ESR2_MASK_CLR_OFFSET 0x944
+#define ESR3_MASK_CLR_OFFSET 0x95c
+#define ESR4_MASK_CLR_OFFSET 0x974
+/* Error Status Regs --- End */
+
+#define R5F_ESR0_SHIFT 0 /* esr0 = fifo underflow */
+#define R5F_ESR1_SHIFT 1 /* esr1 = ringbuf underflow */
+#define R5F_ESR2_SHIFT 2 /* esr2 = ringbuf overflow */
+#define R5F_ESR3_SHIFT 3 /* esr3 = freemark */
+#define R5F_ESR4_SHIFT 4 /* esr4 = fullmark */
+
+
+/* Mask for R5F register. Set all relevant interrupt for playback handler */
+#define ANY_PLAYBACK_IRQ (BIT(R5F_ESR0_SHIFT) | \
+ BIT(R5F_ESR1_SHIFT) | \
+ BIT(R5F_ESR3_SHIFT))
+
+/* Mask for R5F register. Set all relevant interrupt for capture handler */
+#define ANY_CAPTURE_IRQ (BIT(R5F_ESR2_SHIFT) | BIT(R5F_ESR4_SHIFT))
+
+/*
+ * PERIOD_BYTES_MIN is the number of bytes to at which the interrupt will tick.
+ * This number should be a multiple of 256
+ */
+#define PERIOD_BYTES_MIN 0x100
+
+static const struct snd_pcm_hardware cygnus_pcm_hw = {
+ .info = SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+
+ /* A period is basically an interrupt */
+ .period_bytes_min = PERIOD_BYTES_MIN,
+ .period_bytes_max = 0x10000,
+
+ /* period_min/max gives range of approx interrupts per buffer */
+ .periods_min = 2,
+ .periods_max = 8,
+
+ /*
+ * maximum buffer size in bytes = period_bytes_max * periods_max
+ * We allocate this amount of data for each enabled channel
+ */
+ .buffer_bytes_max = 4 * 0x8000,
+};
+
+static u64 cygnus_dma_dmamask = DMA_BIT_MASK(32);
+
+static int enable_count;
+
+static struct cygnus_aio_port *cygnus_dai_get_dma_data(
+ struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *soc_runtime = substream->private_data;
+
+ return snd_soc_dai_get_dma_data(soc_runtime->cpu_dai, substream);
+}
+
+static void ringbuf_set_initial(void __iomem *audio_io,
+ struct ringbuf_regs *p_rbuf,
+ bool is_playback,
+ u32 start,
+ u32 periodsize,
+ u32 bufsize)
+{
+ u32 initial_rd;
+ u32 initial_wr;
+ u32 end;
+ u32 fmark_val; /* free or full mark */
+
+ p_rbuf->period_bytes = periodsize;
+ p_rbuf->buf_size = bufsize;
+
+ if (is_playback) {
+ /*
+ * Set the read pointer one period behind the write.
+ * This should cause an immediate freemark interrupt
+ */
+ initial_rd = start;
+ initial_wr = start + periodsize;
+ } else {
+ /*
+ * Set the write pointer one period behind the read.
+ * This should cause an immediate fullmark interrupt
+ */
+ initial_rd = start + periodsize;
+ initial_wr = start;
+ }
+
+ end = start + bufsize - 1;
+ fmark_val = bufsize - periodsize;
+
+ writel(start, audio_io + p_rbuf->baseaddr);
+ writel(end, audio_io + p_rbuf->endaddr);
+ writel(fmark_val, audio_io + p_rbuf->fmark);
+ writel(initial_rd, audio_io + p_rbuf->rdaddr);
+ writel(initial_wr, audio_io + p_rbuf->wraddr);
+}
+
+static int configure_ringbuf_regs(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ struct ringbuf_regs *p_rbuf;
+ int status = 0;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ /* Map the ssp portnum to a set of ring buffers. */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ p_rbuf = &aio->play_rb_regs;
+
+ if (aio->portnum == 0)
+ *p_rbuf = RINGBUF_REG_PLAYBACK(0);
+ else if (aio->portnum == 1)
+ *p_rbuf = RINGBUF_REG_PLAYBACK(2);
+ else if (aio->portnum == 2)
+ *p_rbuf = RINGBUF_REG_PLAYBACK(4);
+ else if (aio->portnum == 3)
+ *p_rbuf = RINGBUF_REG_PLAYBACK(6); /*SPDIF */
+ else
+ status = -EINVAL;
+ } else {
+ p_rbuf = &aio->capture_rb_regs;
+
+ if (aio->portnum == 0)
+ *p_rbuf = RINGBUF_REG_CAPTURE(0);
+ else if (aio->portnum == 1)
+ *p_rbuf = RINGBUF_REG_CAPTURE(2);
+ else if (aio->portnum == 2)
+ *p_rbuf = RINGBUF_REG_CAPTURE(4);
+ else
+ status = -EINVAL;
+ }
+
+ return status;
+}
+
+static void ringbuf_inc(void __iomem *audio_io, bool is_playback,
+ const struct ringbuf_regs *p_rbuf)
+{
+ u32 regval, endval, active_ptr;
+
+ if (is_playback)
+ active_ptr = p_rbuf->wraddr;
+ else
+ active_ptr = p_rbuf->rdaddr;
+
+ endval = readl(audio_io + p_rbuf->endaddr);
+ regval = readl(audio_io + active_ptr);
+ regval = regval + p_rbuf->period_bytes;
+ if (regval > endval)
+ regval -= p_rbuf->buf_size;
+
+ writel(regval, audio_io + active_ptr);
+}
+
+static struct ringbuf_regs *get_ringbuf(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ struct ringbuf_regs *p_rbuf = NULL;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ p_rbuf = &aio->play_rb_regs;
+ else
+ p_rbuf = &aio->capture_rb_regs;
+
+ return p_rbuf;
+}
+
+static void enable_intr(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct cygnus_aio_port *aio;
+ u32 clear_mask;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ /* The port number maps to the bit position to be cleared */
+ clear_mask = BIT(aio->portnum);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Clear interrupt status before enabling them */
+ writel(clear_mask, aio->audio + ESR0_STATUS_CLR_OFFSET);
+ writel(clear_mask, aio->audio + ESR1_STATUS_CLR_OFFSET);
+ writel(clear_mask, aio->audio + ESR3_STATUS_CLR_OFFSET);
+ /* Unmask the interrupts of the given port*/
+ writel(clear_mask, aio->audio + ESR0_MASK_CLR_OFFSET);
+ writel(clear_mask, aio->audio + ESR1_MASK_CLR_OFFSET);
+ writel(clear_mask, aio->audio + ESR3_MASK_CLR_OFFSET);
+ } else {
+ writel(clear_mask, aio->audio + ESR2_STATUS_CLR_OFFSET);
+ writel(clear_mask, aio->audio + ESR4_STATUS_CLR_OFFSET);
+ writel(clear_mask, aio->audio + ESR2_MASK_CLR_OFFSET);
+ writel(clear_mask, aio->audio + ESR4_MASK_CLR_OFFSET);
+ }
+
+ if (!enable_count) {
+ /* One time clear all the ESR registers */
+ writel(0x1f, aio->audio + INTH_R5F_CLEAR_OFFSET);
+ writel(0x1f, aio->audio + INTH_R5F_MASK_CLEAR_OFFSET);
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d once\n",
+ __func__, aio->portnum);
+ }
+ enable_count++;
+}
+
+static void disable_intr(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct cygnus_aio_port *aio;
+ u32 set_mask;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ dev_dbg(rtd->cpu_dai->dev, "%s on port %d\n", __func__, aio->portnum);
+
+ /* The port number maps to the bit position to be set */
+ set_mask = BIT(aio->portnum);
+
+ enable_count--;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Mask the interrupts of the given port*/
+ writel(set_mask, aio->audio + ESR0_MASK_SET_OFFSET);
+ writel(set_mask, aio->audio + ESR1_MASK_SET_OFFSET);
+ writel(set_mask, aio->audio + ESR3_MASK_SET_OFFSET);
+ } else {
+ writel(set_mask, aio->audio + ESR2_MASK_SET_OFFSET);
+ writel(set_mask, aio->audio + ESR4_MASK_SET_OFFSET);
+ }
+
+ if (!enable_count) {
+ /* Disable all the ESR registers after all streams are closed*/
+ writel(0x1F, aio->audio + INTH_R5F_MASK_SET_OFFSET);
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d once\n",
+ __func__, aio->portnum);
+ }
+}
+
+static int cygnus_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ enable_intr(substream);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ disable_intr(substream);
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static void cygnus_pcm_period_elapsed(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ struct ringbuf_regs *p_rbuf = NULL;
+ bool is_play;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ p_rbuf = get_ringbuf(substream);
+
+ is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+
+ /*
+ * If free/full mark interrupt occurs, provide timestamp
+ * to ALSA and update appropriate idx by period_bytes
+ */
+ snd_pcm_period_elapsed(substream);
+
+ ringbuf_inc(aio->audio, is_play, p_rbuf);
+}
+
+/*
+ * ESR0/1/3 status Description
+ * 0x1 I2S0_out port caused interrupt
+ * 0x2 I2S1_out port caused interrupt
+ * 0x4 I2S2_out port caused interrupt
+ * 0x8 SPDIF_out port caused interrupt
+ */
+static void handle_playback_irq(struct cygnus_audio *cygaud)
+{
+ void __iomem *audio_io;
+ u32 port;
+ u32 esr_status0, esr_status1, esr_status3;
+
+ audio_io = cygaud->audio;
+
+ /*
+ * ESR status gets updates with/without interrupts enabled.
+ * So, check the ESR mask, which provides interrupt enable/
+ * disable status and use it to determine which ESR status
+ * should be serviced.
+ */
+ esr_status0 = readl(audio_io + ESR0_STATUS_OFFSET);
+ esr_status0 &= ~readl(audio_io + ESR0_MASK_STATUS_OFFSET);
+ esr_status1 = readl(audio_io + ESR1_STATUS_OFFSET);
+ esr_status1 &= ~readl(audio_io + ESR1_MASK_STATUS_OFFSET);
+ esr_status3 = readl(audio_io + ESR3_STATUS_OFFSET);
+ esr_status3 &= ~readl(audio_io + ESR3_MASK_STATUS_OFFSET);
+
+ for (port = 0; port < CYGNUS_MAX_PLAYBACK_PORTS; port++) {
+ u32 esrmask = BIT(port);
+
+ /*
+ * Ringbuffer or FIFO underflow
+ * If we get this interrupt then, it is also true that we have
+ * not yet responded to the freemark interrupt.
+ * Log a debug message. The freemark handler below will
+ * handle getting everything going again.
+ */
+ if ((esrmask & esr_status1) || (esrmask & esr_status0)) {
+ pr_debug("Underrun: esr0=0x%x, esr1=0x%x esr3=0x%x\n",
+ esr_status0, esr_status1, esr_status3);
+ }
+
+ /*
+ * Freemark is hit. This is the normal interrupt.
+ * In typical operation the read and write regs will be equal
+ */
+ if (esrmask & esr_status3) {
+ struct snd_pcm_substream *playstr;
+
+ playstr = cygaud->portinfo[port].play_stream;
+ cygnus_pcm_period_elapsed(playstr);
+ }
+ }
+
+ /* Clear ESR interrupt */
+ writel(esr_status0, audio_io + ESR0_STATUS_CLR_OFFSET);
+ writel(esr_status1, audio_io + ESR1_STATUS_CLR_OFFSET);
+ writel(esr_status3, audio_io + ESR3_STATUS_CLR_OFFSET);
+ /* Rearm freemark logic by writing 1 to the correct bit */
+ writel(esr_status3, audio_io + BF_REARM_FREE_MARK_OFFSET);
+}
+
+/*
+ * ESR2/4 status Description
+ * 0x1 I2S0_in port caused interrupt
+ * 0x2 I2S1_in port caused interrupt
+ * 0x4 I2S2_in port caused interrupt
+ */
+static void handle_capture_irq(struct cygnus_audio *cygaud)
+{
+ void __iomem *audio_io;
+ u32 port;
+ u32 esr_status2, esr_status4;
+
+ audio_io = cygaud->audio;
+
+ /*
+ * ESR status gets updates with/without interrupts enabled.
+ * So, check the ESR mask, which provides interrupt enable/
+ * disable status and use it to determine which ESR status
+ * should be serviced.
+ */
+ esr_status2 = readl(audio_io + ESR2_STATUS_OFFSET);
+ esr_status2 &= ~readl(audio_io + ESR2_MASK_STATUS_OFFSET);
+ esr_status4 = readl(audio_io + ESR4_STATUS_OFFSET);
+ esr_status4 &= ~readl(audio_io + ESR4_MASK_STATUS_OFFSET);
+
+ for (port = 0; port < CYGNUS_MAX_CAPTURE_PORTS; port++) {
+ u32 esrmask = BIT(port);
+
+ /*
+ * Ringbuffer or FIFO overflow
+ * If we get this interrupt then, it is also true that we have
+ * not yet responded to the fullmark interrupt.
+ * Log a debug message. The fullmark handler below will
+ * handle getting everything going again.
+ */
+ if (esrmask & esr_status2)
+ pr_debug("Overflow: esr2=0x%x\n", esr_status2);
+
+ if (esrmask & esr_status4) {
+ struct snd_pcm_substream *capstr;
+
+ capstr = cygaud->portinfo[port].capture_stream;
+ cygnus_pcm_period_elapsed(capstr);
+ }
+ }
+
+ writel(esr_status2, audio_io + ESR2_STATUS_CLR_OFFSET);
+ writel(esr_status4, audio_io + ESR4_STATUS_CLR_OFFSET);
+ /* Rearm fullmark logic by writing 1 to the correct bit */
+ writel(esr_status4, audio_io + BF_REARM_FULL_MARK_OFFSET);
+}
+
+static irqreturn_t cygnus_dma_irq(int irq, void *data)
+{
+ u32 r5_status;
+ struct cygnus_audio *cygaud;
+
+ cygaud = (struct cygnus_audio *)data;
+
+ if (!cygaud) {
+ pr_err("ERROR: cyg_aud is NULL\n");
+ return IRQ_NONE;
+ }
+
+ /*
+ * R5 status bits Description
+ * 0 ESR0 (playback FIFO interrupt)
+ * 1 ESR1 (playback rbuf interrupt)
+ * 2 ESR2 (capture rbuf interrupt)
+ * 3 ESR3 (Freemark play. interrupt)
+ * 4 ESR4 (Fullmark capt. interrupt)
+ */
+ r5_status = readl(cygaud->audio + INTH_R5F_STATUS_OFFSET);
+
+ /* If playback interrupt happened */
+ if (ANY_PLAYBACK_IRQ & r5_status)
+ handle_playback_irq(cygaud);
+
+ /* If capture interrupt happened */
+ if (ANY_CAPTURE_IRQ & r5_status)
+ handle_capture_irq(cygaud);
+
+ /*
+ * clear r5 interrupts after servicing them to avoid overwriting
+ * esr_status
+ */
+ writel(r5_status, cygaud->audio + INTH_R5F_CLEAR_OFFSET);
+ return IRQ_HANDLED;
+}
+
+static int cygnus_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cygnus_aio_port *aio;
+ int ret;
+
+ aio = cygnus_dai_get_dma_data(substream);
+ if (!aio)
+ return -ENODEV;
+
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ snd_soc_set_runtime_hwparams(substream, &cygnus_pcm_hw);
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, PERIOD_BYTES_MIN);
+ if (ret < 0)
+ return ret;
+
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, PERIOD_BYTES_MIN);
+ if (ret < 0)
+ return ret;
+ /*
+ * Keep track of which substream belongs to which port.
+ * This info is needed by snd_pcm_period_elapsed() in irq_handler
+ */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aio->play_stream = substream;
+ else
+ aio->capture_stream = substream;
+
+ return 0;
+}
+
+static int cygnus_pcm_close(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct cygnus_aio_port *aio;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ aio->play_stream = NULL;
+ else
+ aio->capture_stream = NULL;
+
+ if (!aio->play_stream && !aio->capture_stream)
+ dev_dbg(rtd->cpu_dai->dev, "freed port %d\n", aio->portnum);
+
+ return 0;
+}
+
+static int cygnus_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cygnus_aio_port *aio;
+ int ret = 0;
+
+ aio = cygnus_dai_get_dma_data(substream);
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ runtime->dma_bytes = params_buffer_bytes(params);
+
+ return ret;
+}
+
+static int cygnus_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct cygnus_aio_port *aio;
+
+ aio = cygnus_dai_get_dma_data(substream);
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ snd_pcm_set_runtime_buffer(substream, NULL);
+ return 0;
+}
+
+/*
+ * Ringbuffer startup logic
+ * Capture Playback
+ * +---------+ <= rdaddr/baseaddr +---------+ <= wraddr/baseaddr
+ * | | | |
+ * | | | |
+ * | | | |
+ * | | | |
+ * | | | |
+ * | | | |
+ * | | | |
+ * | | | |
+ * | | <= wraddr/fullmark | | <= rdaddr/freemark
+ * | | (size-PERIOD_BYTES) | | (size-PERIOD_BYTES)
+ * | | | |
+ * +---------+ <= endaddr +---------+ <= endaddr
+ *
+ * size = endaddr - baseaddr
+ */
+static int cygnus_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct cygnus_aio_port *aio;
+ unsigned long bufsize, periodsize;
+ int ret = 0;
+ bool is_play;
+ u32 start;
+ struct ringbuf_regs *p_rbuf = NULL;
+
+ aio = cygnus_dai_get_dma_data(substream);
+ dev_dbg(rtd->cpu_dai->dev, "%s port %d\n", __func__, aio->portnum);
+
+ bufsize = snd_pcm_lib_buffer_bytes(substream);
+ periodsize = snd_pcm_lib_period_bytes(substream);
+
+ dev_dbg(rtd->cpu_dai->dev, "%s (buf_size %lu) (period_size %lu)\n",
+ __func__, bufsize, periodsize);
+
+ configure_ringbuf_regs(substream);
+
+ p_rbuf = get_ringbuf(substream);
+
+ start = runtime->dma_addr;
+
+ is_play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 1 : 0;
+
+ ringbuf_set_initial(aio->audio, p_rbuf, is_play, start,
+ periodsize, bufsize);
+
+ return ret;
+}
+
+static snd_pcm_uframes_t cygnus_pcm_pointer(struct snd_pcm_substream *substream)
+{
+ struct cygnus_aio_port *aio;
+ int res = 0;
+ unsigned int cur = 0, base = 0;
+ struct ringbuf_regs *p_rbuf = NULL;
+
+ aio = cygnus_dai_get_dma_data(substream);
+
+ /*
+ * Get the offset of the current read (for playack) or write
+ * index (for capture). Report this value back to the asoc framework.
+ */
+ p_rbuf = get_ringbuf(substream);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ cur = readl(aio->audio + p_rbuf->rdaddr);
+ else
+ cur = readl(aio->audio + p_rbuf->wraddr);
+
+ base = readl(aio->audio + p_rbuf->baseaddr);
+
+ /*
+ * Mask off the MSB of the rdaddr,wraddr and baseaddr
+ * since MSB is not part of the address
+ */
+ res = (cur & 0x7fffffff) - (base & 0x7fffffff);
+
+ return bytes_to_frames(substream->runtime, res);
+}
+
+static int cygnus_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream)
+{
+ struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_dma_buffer *buf = &substream->dma_buffer;
+ size_t size;
+
+ size = cygnus_pcm_hw.buffer_bytes_max;
+
+ buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ buf->dev.dev = pcm->card->dev;
+ buf->private_data = NULL;
+ buf->area = dma_alloc_coherent(pcm->card->dev, size,
+ &buf->addr, GFP_KERNEL);
+
+ dev_dbg(rtd->cpu_dai->dev, "%s: size 0x%x @ 0x%p\n",
+ __func__, size, buf->area);
+
+ if (!buf->area) {
+ dev_err(rtd->cpu_dai->dev, "%s: dma_alloc failed\n", __func__);
+ return -ENOMEM;
+ }
+ buf->bytes = size;
+
+ return 0;
+}
+
+/*
+ * This code is identical to what is done by the framework, when we do not
+ * supply a 'copy' function. Having our own copy hook in place allows for
+ * us to easily add some diagnotics when needed.
+ */
+int cygnus_pcm_copy(struct snd_pcm_substream *substream, int channel,
+ snd_pcm_uframes_t pos,
+ void __user *buf, snd_pcm_uframes_t count)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ char *hwbuf = runtime->dma_area + frames_to_bytes(runtime, pos);
+ int size = frames_to_bytes(runtime, count);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (copy_from_user(hwbuf, buf, size))
+ return -EFAULT;
+ } else {
+ if (copy_to_user(buf, hwbuf, size))
+ return -EFAULT;
+ }
+ return 0;
+}
+
+static struct snd_pcm_ops cygnus_pcm_ops = {
+ .open = cygnus_pcm_open,
+ .close = cygnus_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = cygnus_pcm_hw_params,
+ .hw_free = cygnus_pcm_hw_free,
+ .prepare = cygnus_pcm_prepare,
+ .trigger = cygnus_pcm_trigger,
+ .pointer = cygnus_pcm_pointer,
+ .copy = cygnus_pcm_copy,
+};
+
+static void cygnus_dma_free_dma_buffers(struct snd_pcm *pcm)
+{
+ struct snd_pcm_substream *substream;
+ struct snd_dma_buffer *buf;
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (substream) {
+ buf = &substream->dma_buffer;
+ if (buf->area) {
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+ }
+
+ substream = pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream;
+ if (substream) {
+ buf = &substream->dma_buffer;
+ if (buf->area) {
+ dma_free_coherent(pcm->card->dev, buf->bytes,
+ buf->area, buf->addr);
+ buf->area = NULL;
+ }
+ }
+}
+
+static int cygnus_dma_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ struct snd_pcm *pcm = rtd->pcm;
+ int ret;
+
+ if (!card->dev->dma_mask)
+ card->dev->dma_mask = &cygnus_dma_dmamask;
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ret = cygnus_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret)
+ return ret;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ret = cygnus_pcm_preallocate_dma_buffer(pcm,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret) {
+ cygnus_dma_free_dma_buffers(pcm);
+ return ret;
+ }
+ }
+
+ return 0;
+}
+
+static struct snd_soc_platform_driver cygnus_soc_platform = {
+ .ops = &cygnus_pcm_ops,
+ .pcm_new = cygnus_dma_new,
+ .pcm_free = cygnus_dma_free_dma_buffers,
+};
+
+int cygnus_soc_platform_register(struct device *dev,
+ struct cygnus_audio *cygaud)
+{
+ int rc = 0;
+
+ dev_dbg(dev, "%s Enter\n", __func__);
+
+ rc = devm_request_irq(dev, cygaud->irq_num, cygnus_dma_irq,
+ IRQF_SHARED, "cygnus-audio", cygaud);
+ if (rc) {
+ dev_err(dev, "%s request_irq error %d\n", __func__, rc);
+ return rc;
+ }
+
+ rc = devm_snd_soc_register_platform(dev, &cygnus_soc_platform);
+ if (rc) {
+ dev_err(dev, "%s failed\n", __func__);
+ return rc;
+ }
+
+ return 0;
+}
+
+int cygnus_soc_platform_unregister(struct device *dev)
+{
+ return 0;
+}
+
+MODULE_LICENSE("GPL v2");
+MODULE_AUTHOR("Broadcom");
+MODULE_DESCRIPTION("Cygnus ASoC PCM module");
diff --git a/sound/soc/bcm/cygnus-pcm.h b/sound/soc/bcm/cygnus-pcm.h
new file mode 100644
index 0000000..fd9f6ff
--- /dev/null
+++ b/sound/soc/bcm/cygnus-pcm.h
@@ -0,0 +1,45 @@
+/*
+ * Copyright (C) 2014-2015 Broadcom Corporation
+ *
+ * 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 __CYGNUS_PCM_H__
+#define __CYGNUS_PCM_H__
+
+struct ringbuf_regs {
+ unsigned rdaddr;
+ unsigned wraddr;
+ unsigned baseaddr;
+ unsigned endaddr;
+ unsigned fmark; /* freemark for play, fullmark for caputure */
+ unsigned period_bytes;
+ unsigned buf_size;
+};
+
+#define RINGBUF_REG_PLAYBACK(num) ((struct ringbuf_regs) { \
+ .rdaddr = SRC_RBUF_ ##num## _RDADDR_OFFSET, \
+ .wraddr = SRC_RBUF_ ##num## _WRADDR_OFFSET, \
+ .baseaddr = SRC_RBUF_ ##num## _BASEADDR_OFFSET, \
+ .endaddr = SRC_RBUF_ ##num## _ENDADDR_OFFSET, \
+ .fmark = SRC_RBUF_ ##num## _FREE_MARK_OFFSET, \
+ .period_bytes = 0, \
+ .buf_size = 0, \
+})
+
+#define RINGBUF_REG_CAPTURE(num) ((struct ringbuf_regs) { \
+ .rdaddr = DST_RBUF_ ##num## _RDADDR_OFFSET, \
+ .wraddr = DST_RBUF_ ##num## _WRADDR_OFFSET, \
+ .baseaddr = DST_RBUF_ ##num## _BASEADDR_OFFSET, \
+ .endaddr = DST_RBUF_ ##num## _ENDADDR_OFFSET, \
+ .fmark = DST_RBUF_ ##num## _FULL_MARK_OFFSET, \
+ .period_bytes = 0, \
+ .buf_size = 0, \
+})
+#endif
diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c
new file mode 100644
index 0000000..f37198d
--- /dev/null
+++ b/sound/soc/bcm/cygnus-ssp.c
@@ -0,0 +1,1613 @@
+/*
+ * Copyright (C) 2014-2015 Broadcom Corporation
+ *
+ * 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.
+ */
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <linux/of_device.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dai.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include "cygnus-ssp.h"
+
+#define I2S0 0
+#define I2S1 1
+#define I2S2 2
+#define SPDIF 3
+
+#define CYGNUS_TDM_RATE \
+ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000 | \
+ SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \
+ SNDRV_PCM_RATE_48000)
+
+#define PLL_NDIV_FRACT_MAX (BIT(20)-1) /* 20 bits max */
+
+#define CAPTURE_FCI_ID_BASE 0x180
+
+/* Begin register offset defines */
+#define AUD_MISC_SEROUT_OE_REG_BASE 0x01c
+#define AUD_MISC_SEROUT_MCLK_OE 0x8
+#define AUD_MISC_SEROUT_LRCK_OE 0x4
+#define AUD_MISC_SEROUT_SCLK_OE 0x2
+#define AUD_MISC_SEROUT_SDAT_OE 0x1
+
+/* AUD_FMM_BF_CTRL_xxx regs */
+#define BF_DST_CFG0_OFFSET 0x100
+#define BF_DST_CFG1_OFFSET 0x104
+#define BF_DST_CFG2_OFFSET 0x108
+
+#define BF_DST_CTRL0_OFFSET 0x130
+#define BF_DST_CTRL1_OFFSET 0x134
+#define BF_DST_CTRL2_OFFSET 0x138
+
+#define BF_SRC_CFG0_OFFSET 0x148
+#define BF_SRC_CFG1_OFFSET 0x14c
+#define BF_SRC_CFG2_OFFSET 0x150
+#define BF_SRC_CFG3_OFFSET 0x154
+
+#define BF_SRC_CTRL0_OFFSET 0x1c0
+#define BF_SRC_CTRL1_OFFSET 0x1c4
+#define BF_SRC_CTRL2_OFFSET 0x1c8
+#define BF_SRC_CTRL3_OFFSET 0x1cc
+
+#define BF_SRC_GRP0_OFFSET 0x1fc
+#define BF_SRC_GRP1_OFFSET 0x200
+#define BF_SRC_GRP2_OFFSET 0x204
+#define BF_SRC_GRP3_OFFSET 0x208
+
+/* AUD_FMM_IOP_OUT_I2S_xxx regs */
+#define OUT_I2S_0_STREAM_CFG_OFFSET 0xa00
+#define OUT_I2S_0_CFG_OFFSET 0xa04
+#define OUT_I2S_0_MCLK_CFG_OFFSET 0xa0c
+
+#define OUT_I2S_1_STREAM_CFG_OFFSET 0xa40
+#define OUT_I2S_1_CFG_OFFSET 0xa44
+#define OUT_I2S_1_MCLK_CFG_OFFSET 0xa4c
+
+#define OUT_I2S_2_STREAM_CFG_OFFSET 0xa80
+#define OUT_I2S_2_CFG_OFFSET 0xa84
+#define OUT_I2S_2_MCLK_CFG_OFFSET 0xa8c
+
+/* AUD_FMM_IOP_OUT_SPDIF_xxx regs */
+#define SPDIF_STREAM_CFG_OFFSET 0xac0
+#define SPDIF_CTRL_OFFSET 0xac4
+#define SPDIF_FORMAT_CFG_OFFSET 0xad8
+#define SPDIF_MCLK_CFG_OFFSET 0xadc
+
+/* AUD_FMM_IOP_PLL_0_xxx regs */
+#define IOP_PLL_0_MACRO_OFFSET 0xb00
+#define IOP_PLL_0_MDIV_Ch0_OFFSET 0xb14
+#define IOP_PLL_0_MDIV_Ch1_OFFSET 0xb18
+#define IOP_PLL_0_MDIV_Ch2_OFFSET 0xb1c
+
+#define IOP_PLL_0_ACTIVE_MDIV_Ch0_OFFSET 0xb30
+#define IOP_PLL_0_ACTIVE_MDIV_Ch1_OFFSET 0xb34
+#define IOP_PLL_0_ACTIVE_MDIV_Ch2_OFFSET 0xb38
+
+/* AUD_FMM_IOP_xxx regs */
+#define IOP_PLL_0_CONTROL_OFFSET 0xb04
+#define IOP_PLL_0_USER_NDIV_OFFSET 0xb08
+#define IOP_PLL_0_ACTIVE_NDIV_OFFSET 0xb20
+#define IOP_PLL_0_RESET_OFFSET 0xb5c
+#define IOP_NCO_0_CONTROL_OFFSET 0xb80
+#define IOP_NCO_1_CONTROL_OFFSET 0xbc0
+
+/* AUD_FMM_IOP_IN_I2S_xxx regs */
+#define IN_I2S_0_STREAM_CFG_OFFSET 0xc00
+#define IN_I2S_0_CFG_OFFSET 0xc04
+#define IN_I2S_1_STREAM_CFG_OFFSET 0xc40
+#define IN_I2S_1_CFG_OFFSET 0xc44
+#define IN_I2S_2_STREAM_CFG_OFFSET 0xc80
+#define IN_I2S_2_CFG_OFFSET 0xc84
+
+/* End register offset defines */
+
+
+/* AUD_FMM_IOP_OUT_I2S_x_MCLK_CFG_0_REG */
+#define I2S_OUT_MCLKRATE_SHIFT 16
+
+/* AUD_FMM_IOP_OUT_I2S_x_MCLK_CFG_REG */
+#define I2S_OUT_PLLCLKSEL_SHIFT 0
+
+/* AUD_FMM_IOP_OUT_I2S_x_STREAM_CFG */
+#define I2S_OUT_STREAM_ENA 31
+#define I2S_OUT_STREAM_CFG_GROUP_ID 20
+#define I2S_OUT_STREAM_CFG_CHANNEL_GROUPING 24
+
+/* AUD_FMM_IOP_IN_I2S_x_CAP */
+#define I2S_IN_STREAM_CFG_CAP_ENA 31
+#define I2S_IN_STREAM_CFG_0_GROUP_ID 4
+
+/* AUD_FMM_IOP_OUT_I2S_x_I2S_CFG_REG */
+#define I2S_OUT_CFGX_CLK_ENA 0
+#define I2S_OUT_CFGX_DATA_ENABLE 1
+#define I2S_OUT_CFGX_DATA_ALIGNMENT 6
+#define I2S_OUT_CFGX_BITS_PER_SLOT 13
+#define I2S_OUT_CFGX_VALID_SLOT 14
+#define I2S_OUT_CFGX_FSYNC_WIDTH 18
+#define I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32 26
+#define I2S_OUT_CFGX_SLAVE_MODE 30
+#define I2S_OUT_CFGX_TDM_MODE 31
+
+/* AUD_FMM_BF_CTRL_SOURCECH_CFGx_REG */
+#define BF_SRC_CFGX_SFIFO_ENA 0
+#define BF_SRC_CFGX_BUFFER_PAIR_ENABLE 1
+#define BF_SRC_CFGX_SAMPLE_CH_MODE 2
+#define BF_SRC_CFGX_SFIFO_SZ_DOUBLE 5
+#define BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY 10
+#define BF_SRC_CFGX_BIT_RES 20
+#define BF_SRC_CFGX_PROCESS_SEQ_ID_VALID 31
+
+/* AUD_FMM_BF_CTRL_DESTCH_CFGx_REG */
+#define BF_DST_CFGX_CAP_ENA 0
+#define BF_DST_CFGX_BUFFER_PAIR_ENABLE 1
+#define BF_DST_CFGX_DFIFO_SZ_DOUBLE 2
+#define BF_DST_CFGX_NOT_PAUSE_WHEN_FULL 11
+#define BF_DST_CFGX_FCI_ID 12
+#define BF_DST_CFGX_CAP_MODE 24
+#define BF_DST_CFGX_PROC_SEQ_ID_VALID 31
+
+/* AUD_FMM_IOP_OUT_SPDIF_xxx */
+#define SPDIF_0_OUT_DITHER_ENA 3
+#define SPDIF_0_OUT_STREAM_ENA 31
+
+/* AUD_FMM_IOP_NCO_xxx */
+#define IOP_NCO_NUMERATOR 16
+#define IOP_NCO_CONTROL_RESET 12
+#define IOP_NCO_CONTROL_FREE_RUN 4
+
+/* AUD_FMM_IOP_PLL_0_USER */
+#define IOP_PLL_0_USER_NDIV_FRAC 10
+
+/* AUD_FMM_IOP_PLL_0_ACTIVE */
+#define IOP_PLL_0_ACTIVE_NDIV_FRAC 10
+
+
+#define INIT_SSP_REGS(num) { \
+ .i2s_stream_cfg = OUT_I2S_ ##num## _STREAM_CFG_OFFSET, \
+ .i2s_cap_stream_cfg = IN_I2S_ ##num## _STREAM_CFG_OFFSET, \
+ .i2s_cfg = OUT_I2S_ ##num## _CFG_OFFSET, \
+ .i2s_cap_cfg = IN_I2S_ ##num## _CFG_OFFSET, \
+ .i2s_mclk_cfg = OUT_I2S_ ##num## _MCLK_CFG_OFFSET, \
+ .bf_destch_ctrl = BF_DST_CTRL ##num## _OFFSET, \
+ .bf_destch_cfg = BF_DST_CFG ##num## _OFFSET, \
+ .bf_sourcech_ctrl = BF_SRC_CTRL ##num## _OFFSET, \
+ .bf_sourcech_cfg = BF_SRC_CFG ##num## _OFFSET \
+}
+
+
+static int group_id[CYGNUS_MAX_PLAYBACK_PORTS] = {0, 1, 2, 3};
+
+/*
+ * Choose one of the following valid mclk rates:
+ * -----------------------------------------
+ * macro pll_ch0 pll_ch1 pll_ch2
+ * ----- ------- ---------- ----------
+ * 0 4,096,000 8,192,000 16,384,000
+ * 1 5,644,800 11,289,600 22,579,200
+ * 2 6,144,000 12,288,000 24,576,000
+ * 3 12,288,000 24,576,000 49,152,000
+ * 4 22,579,200 45,158,400 90,316,800
+ * 5 24,576,000 49,152,000 98,304,000
+ * -----------------------------------------
+ *
+ * Use this table to look up the "macro" setting for the audio pll block.
+ * There are 6 macro settings that produce some common mclk frequencies.
+ * The pll has 3 output channels (1x, 2x, and 4x).
+ */
+struct pll_macro_entry {
+ u32 mclk;
+ u32 macro;
+ u32 pll_ch_num;
+};
+
+static const struct pll_macro_entry pll_predef_mclk[] = {
+ { 4096000, 0, 0},
+ { 8192000, 0, 1},
+ {16384000, 0, 2},
+
+ { 5644800, 1, 0},
+ {11289600, 1, 1},
+ {22579200, 1, 2},
+
+ { 6144000, 2, 0},
+ {12288000, 2, 1},
+ {24576000, 2, 2},
+
+ {12288000, 3, 0},
+ {24576000, 3, 1},
+ {49152000, 3, 2},
+
+ {22579200, 4, 0},
+ {45158400, 4, 1},
+ {90316800, 4, 2},
+
+ {24576000, 5, 0},
+ {49152000, 5, 1},
+ {98304000, 5, 2},
+};
+
+
+struct _nco_clk_coeff {
+ u32 mclk;
+ u32 sample_inc;
+ u32 numer;
+ u32 denom;
+ u32 phase_inc;
+};
+
+static const struct _nco_clk_coeff nco_clk_coeff[] = {
+ {2048000, 0xD, 0x02F, 0x100, 0x029F16},
+ {4096000, 0x6, 0x12F, 0x200, 0x053E2D},
+ {6144000, 0x4, 0x065, 0x100, 0x07DD44},
+ {8192000, 0x3, 0x12F, 0x400, 0x0A7C5A},
+ {12288000, 0x2, 0x065, 0x200, 0x0FBA88},
+ {24576000, 0x1, 0x065, 0x400, 0x1F7510},
+ {49152000, 0x0, 0x465, 0x800, 0x3EEA20},
+
+ {2822400, 0x9, 0x06F, 0x0C4, 0x039CD8},
+ {5644800, 0x4, 0x133, 0x188, 0x0739B0},
+ {11289600, 0x2, 0x133, 0x310, 0x0E7360},
+ {22579200, 0x1, 0x133, 0x620, 0x1CE6C0},
+ {45158400, 0x0, 0x753, 0xC40, 0x39CD81},
+};
+/*
+ * Use this relationship to derive the sampling rate (lrclk)
+ * lrclk = (mclk) / ((2*mclk_to_sclk_ratio) * (32 * SCLK))).
+ *
+ * Use mclk, macro and pll_ch from the table above
+ *
+ * Valid SCLK = 0/1/2/4/8/12
+ *
+ * mclk_to_sclk_ratio = number of MCLK per SCLK. Division is twice the
+ * value programmed in this field.
+ * Valid mclk_to_sclk_ratio = 1 through to 15
+ *
+ * eg: To set lrclk = 48khz, set mclk = 12288000, mclk_to_sclk_ratio = 2,
+ * SCLK = 64
+ */
+struct _ssp_clk_coeff {
+ u32 mclk;
+ u32 rate;
+ u32 sclk_rate;
+ u32 mclk_rate;
+};
+
+static const struct _ssp_clk_coeff ssp_clk_coeff[] = {
+ /* 8k */
+ { 4096000, 8000, 64, 4},
+ { 6144000, 8000, 64, 6},
+ { 8192000, 8000, 64, 8},
+ {12288000, 8000, 64, 12},
+
+ { 4096000, 8000, 128, 2},
+ { 6144000, 8000, 128, 3},
+ { 8192000, 8000, 128, 4},
+ {12288000, 8000, 128, 6},
+ {16384000, 8000, 128, 8},
+ {24576000, 8000, 128, 12},
+
+ { 4096000, 8000, 256, 1},
+ { 8192000, 8000, 256, 2},
+ {12288000, 8000, 256, 3},
+ {16384000, 8000, 256, 4},
+ {24576000, 8000, 256, 6},
+ {49152000, 8000, 256, 12},
+
+ { 8192000, 8000, 512, 1},
+ {16384000, 8000, 512, 2},
+ {24576000, 8000, 512, 3},
+ {49152000, 8000, 512, 6},
+ {98304000, 8000, 512, 12},
+
+ /* 16k */
+ { 4096000, 16000, 64, 2},
+ { 6144000, 16000, 64, 3},
+ { 8192000, 16000, 64, 4},
+ {12288000, 16000, 64, 6},
+ {16384000, 16000, 64, 8},
+ {24576000, 16000, 64, 12},
+
+ { 4096000, 16000, 128, 1},
+ { 8192000, 16000, 128, 2},
+ {12288000, 16000, 128, 3},
+ {16384000, 16000, 128, 4},
+ {24576000, 16000, 128, 6},
+ {49152000, 16000, 128, 12},
+
+ { 8192000, 16000, 256, 1},
+ {16384000, 16000, 256, 2},
+ {24576000, 16000, 256, 3},
+ {49152000, 16000, 256, 6},
+ {98304000, 16000, 256, 12},
+
+ {16384000, 16000, 512, 1},
+ {49152000, 16000, 512, 3},
+ {98304000, 16000, 512, 6},
+
+ /* 32k */
+ { 4096000, 32000, 64, 1},
+ { 8192000, 32000, 64, 2},
+ {12288000, 32000, 64, 3},
+ {16384000, 32000, 64, 4},
+ {24576000, 32000, 64, 6},
+ {49152000, 32000, 64, 12},
+
+ { 8192000, 32000, 128, 1},
+ {16384000, 32000, 128, 2},
+ {24576000, 32000, 128, 3},
+ {49152000, 32000, 128, 6},
+ {98304000, 32000, 128, 12},
+
+ {16384000, 32000, 256, 1},
+ {49152000, 32000, 256, 3},
+ {98304000, 32000, 256, 6},
+
+ {98304000, 32000, 512, 3},
+
+ /* 44.1k */
+ { 5644800, 44100, 64, 1},
+ {11289600, 44100, 64, 2},
+ {22579200, 44100, 64, 4},
+ {45158400, 44100, 64, 8},
+
+ {11289600, 44100, 128, 1},
+ {22579200, 44100, 128, 2},
+ {45158400, 44100, 128, 4},
+ {90316800, 44100, 128, 8},
+
+ {22579200, 44100, 256, 1},
+ {45158400, 44100, 256, 2},
+ {90316800, 44100, 256, 4},
+
+ {45158400, 44100, 512, 1},
+ {90316800, 44100, 512, 2},
+
+ /* 48k */
+ { 6144000, 48000, 64, 1},
+ {12288000, 48000, 64, 2},
+ {24576000, 48000, 64, 4},
+ {49152000, 48000, 64, 8},
+
+ {12288000, 48000, 128, 1},
+ {24576000, 48000, 128, 2},
+ {49152000, 48000, 128, 4},
+ {98304000, 48000, 128, 8},
+
+ {24576000, 48000, 256, 1},
+ {49152000, 48000, 256, 2},
+ {98304000, 48000, 256, 4},
+
+ {49152000, 48000, 512, 1},
+ {98304000, 48000, 512, 2},
+
+ /* 88.2k */
+ {11289600, 88200, 64, 1},
+ {22579200, 88200, 64, 2},
+ {45158400, 88200, 64, 4},
+ {90316800, 88200, 64, 8},
+
+ /* 96k */
+ {12288000, 96000, 64, 1},
+ {24576000, 96000, 64, 2},
+ {49152000, 96000, 64, 4},
+ {98304000, 96000, 64, 8},
+
+ /* 176.4k */
+ {22579200, 176400, 64, 1},
+ {45158400, 176400, 64, 2},
+ {90316800, 176400, 64, 4},
+
+ /* 192k */
+ {24576000, 192000, 64, 1},
+ {49152000, 192000, 64, 2},
+ {98304000, 192000, 64, 4},
+
+ {49152000, 192000, 128, 1},
+};
+
+static struct cygnus_aio_port *cygnus_dai_get_portinfo(struct snd_soc_dai *dai)
+{
+ struct cygnus_audio *cygaud = snd_soc_dai_get_drvdata(dai);
+
+ return &cygaud->portinfo[dai->id];
+}
+
+static void audio_pll0_init(void __iomem *audio_io)
+{
+ /* Set clock channel post divider ratio to 0x24 */
+ writel(0x24, audio_io + IOP_PLL_0_MDIV_Ch0_OFFSET);
+ writel(0x24, audio_io + IOP_PLL_0_MDIV_Ch1_OFFSET);
+ writel(0x24, audio_io + IOP_PLL_0_MDIV_Ch2_OFFSET);
+ /* Disable and enable digital and analog PLL */
+ writel(0x3, audio_io + IOP_PLL_0_RESET_OFFSET);
+ writel(0x0, audio_io + IOP_PLL_0_RESET_OFFSET);
+}
+
+static int audio_ssp_init_portregs(struct cygnus_aio_port *aio)
+{
+ u32 value, fci_id;
+
+ if ((aio->portnum == I2S0) || (aio->portnum == I2S1)
+ || (aio->portnum == I2S2)) {
+ value = readl(aio->audio + aio->regs.i2s_stream_cfg);
+ value &= ~0xff003ff;
+
+ /* Set Group ID */
+ writel(group_id[0], aio->audio + BF_SRC_GRP0_OFFSET);
+ writel(group_id[1], aio->audio + BF_SRC_GRP1_OFFSET);
+ writel(group_id[2], aio->audio + BF_SRC_GRP2_OFFSET);
+ writel(group_id[3], aio->audio + BF_SRC_GRP3_OFFSET);
+
+ /* Configure the AUD_FMM_IOP_OUT_I2S_x_I2S_CFG reg */
+ value |= group_id[aio->portnum] << I2S_OUT_STREAM_CFG_GROUP_ID;
+ value |= aio->portnum; /* FCI ID is the port num */
+ value |= aio->channel_grouping <<
+ I2S_OUT_STREAM_CFG_CHANNEL_GROUPING;
+ writel(value, aio->audio + aio->regs.i2s_stream_cfg);
+
+ /* Configure the AUD_FMM_BF_CTRL_SOURCECH_CFGX reg */
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY);
+ value |= BIT(BF_SRC_CFGX_SFIFO_SZ_DOUBLE);
+ value |= BIT(BF_SRC_CFGX_PROCESS_SEQ_ID_VALID);
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+
+ /* Configure the AUD_FMM_IOP_IN_I2S_x_CAP_STREAM_CFG_0 reg */
+ value = readl(aio->audio + aio->regs.i2s_cap_stream_cfg);
+ value &= ~0xf0;
+ value |= group_id[aio->portnum] << I2S_IN_STREAM_CFG_0_GROUP_ID;
+ writel(value, aio->audio + aio->regs.i2s_cap_stream_cfg);
+
+ /* Configure the AUD_FMM_BF_CTRL_DESTCH_CFGX_REG_BASE reg */
+ fci_id = CAPTURE_FCI_ID_BASE + aio->portnum;
+
+ value = readl(aio->audio + aio->regs.bf_destch_cfg);
+ value |= BIT(BF_DST_CFGX_DFIFO_SZ_DOUBLE);
+ value &= ~BIT(BF_DST_CFGX_NOT_PAUSE_WHEN_FULL);
+ value |= (fci_id << BF_DST_CFGX_FCI_ID);
+ value |= BIT(BF_DST_CFGX_PROC_SEQ_ID_VALID);
+ writel(value, aio->audio + aio->regs.bf_destch_cfg);
+
+ } else if (aio->portnum == SPDIF) {
+ writel(group_id[3], aio->audio + BF_SRC_GRP3_OFFSET);
+
+ value = readl(aio->audio + SPDIF_CTRL_OFFSET);
+ value |= BIT(SPDIF_0_OUT_DITHER_ENA);
+ writel(value, aio->audio + SPDIF_CTRL_OFFSET);
+
+ /* Enable and set the FCI ID for the SPDIF channel */
+ value = readl(aio->audio + SPDIF_STREAM_CFG_OFFSET);
+ value &= ~0x3ff;
+ value |= aio->portnum; /* FCI ID is the port num */
+ value |= BIT(SPDIF_0_OUT_STREAM_ENA);
+ writel(value, aio->audio + SPDIF_STREAM_CFG_OFFSET);
+
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_NOT_PAUSE_WHEN_EMPTY);
+ value |= BIT(BF_SRC_CFGX_SFIFO_SZ_DOUBLE);
+ value |= BIT(BF_SRC_CFGX_PROCESS_SEQ_ID_VALID);
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+ } else {
+ pr_err("Port not supported\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int audio_ssp_in_enable(struct cygnus_aio_port *aio, bool enable)
+{
+ u32 value;
+
+ if (enable) {
+ value = readl(aio->audio + aio->regs.bf_destch_cfg);
+ value |= BIT(BF_DST_CFGX_CAP_ENA);
+ writel(value, aio->audio + aio->regs.bf_destch_cfg);
+
+ writel(0x1, aio->audio + aio->regs.bf_destch_ctrl);
+
+ value = readl(aio->audio + aio->regs.i2s_cfg);
+ value |= BIT(I2S_OUT_CFGX_CLK_ENA);
+ writel(value, aio->audio + aio->regs.i2s_cfg);
+
+ value = readl(aio->audio + aio->regs.i2s_cap_stream_cfg);
+ value |= BIT(I2S_IN_STREAM_CFG_CAP_ENA);
+ writel(value, aio->audio + aio->regs.i2s_cap_stream_cfg);
+ aio->streams_on++;
+ } else {
+ value = readl(aio->audio + aio->regs.i2s_cap_stream_cfg);
+ value &= ~BIT(I2S_IN_STREAM_CFG_CAP_ENA);
+ writel(value, aio->audio + aio->regs.i2s_cap_stream_cfg);
+ if (aio->streams_on == 1) {
+ value = readl(aio->audio + aio->regs.i2s_cfg);
+ value &= ~BIT(I2S_OUT_CFGX_CLK_ENA);
+ writel(value, aio->audio + aio->regs.i2s_cfg);
+ }
+
+ writel(0x0, aio->audio + aio->regs.bf_destch_ctrl);
+
+ value = readl(aio->audio + aio->regs.bf_destch_cfg);
+ value &= ~BIT(BF_DST_CFGX_CAP_ENA);
+ writel(value, aio->audio + aio->regs.bf_destch_cfg);
+ aio->streams_on--;
+ }
+
+ return 0;
+}
+
+static int audio_ssp_out_enable(struct cygnus_aio_port *aio, bool enable)
+{
+ u32 value;
+
+ if (aio->portnum < SPDIF) {
+ if (enable) {
+ value = readl(aio->audio + aio->regs.i2s_stream_cfg);
+ value |= BIT(I2S_OUT_STREAM_ENA);
+ writel(value, aio->audio + aio->regs.i2s_stream_cfg);
+
+ writel(1, aio->audio + aio->regs.bf_sourcech_ctrl);
+
+ value = readl(aio->audio + aio->regs.i2s_cfg);
+ value |= BIT(I2S_OUT_CFGX_CLK_ENA);
+ value |= BIT(I2S_OUT_CFGX_DATA_ENABLE);
+ writel(value, aio->audio + aio->regs.i2s_cfg);
+
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value |= BIT(BF_SRC_CFGX_SFIFO_ENA);
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+ aio->streams_on++;
+ } else {
+ value = readl(aio->audio + aio->regs.i2s_stream_cfg);
+ value &= ~BIT(I2S_OUT_STREAM_ENA);
+ writel(value, aio->audio + aio->regs.i2s_stream_cfg);
+ writel(0, aio->audio + aio->regs.bf_sourcech_ctrl);
+
+ value = readl(aio->audio + aio->regs.i2s_cfg);
+ if (aio->streams_on == 1) {
+ value &= ~BIT(I2S_OUT_CFGX_CLK_ENA);
+ value &= ~BIT(I2S_OUT_CFGX_DATA_ENABLE);
+ } else {
+ value &= ~BIT(I2S_OUT_CFGX_DATA_ENABLE);
+ }
+ writel(value, aio->audio + aio->regs.i2s_cfg);
+
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_SFIFO_ENA);
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+
+ aio->streams_on--;
+ }
+ } else if (aio->portnum == SPDIF) {
+ if (enable) {
+ value = readl(aio->audio + SPDIF_FORMAT_CFG_OFFSET);
+ value |= 0x3;
+ writel(value, aio->audio + SPDIF_FORMAT_CFG_OFFSET);
+
+ writel(1, aio->audio + aio->regs.bf_sourcech_ctrl);
+
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value |= BIT(BF_SRC_CFGX_SFIFO_ENA);
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+ } else {
+ value = readl(aio->audio + SPDIF_FORMAT_CFG_OFFSET);
+ value &= ~0x3;
+ writel(value, aio->audio + SPDIF_FORMAT_CFG_OFFSET);
+ writel(0, aio->audio + aio->regs.bf_sourcech_ctrl);
+
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_SFIFO_ENA);
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+ }
+ } else {
+ pr_err("Port not supported\n");
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int audio_ssp_source_bitres(struct cygnus_aio_port *aio, unsigned bits)
+{
+ u32 mask = 0x1f;
+ u32 value = 0;
+
+ if ((bits == 8) || (bits == 16) || (bits == 32)) {
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~(mask << BF_SRC_CFGX_BIT_RES);
+
+ /* 32 bit mode is coded as 0 */
+ if ((bits == 8) || (bits == 16))
+ value |= (bits << BF_SRC_CFGX_BIT_RES);
+
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+ return 0;
+ }
+
+ pr_err("Bit resolution not supported %d\n", bits);
+ return -EINVAL;
+}
+
+static int audio_ssp_dst_bitres(struct cygnus_aio_port *aio, unsigned bits)
+{
+ u32 value;
+
+ value = readl(aio->audio + aio->regs.bf_destch_cfg);
+ if (bits == 16) {
+ value |= BIT(BF_DST_CFGX_CAP_MODE);
+ writel(value, aio->audio + aio->regs.bf_destch_cfg);
+ } else if (bits == 32) {
+ value &= ~BIT(BF_DST_CFGX_CAP_MODE);
+ writel(value, aio->audio + aio->regs.bf_destch_cfg);
+ } else {
+ pr_err("Bit resolution not supported %d\n", bits);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int nco_configure_mclk(void __iomem *audio_base, int nco_id, int mclk)
+{
+ u32 value;
+ int i;
+ bool found = 0;
+ int offset;
+ int select = -1;
+ const struct _nco_clk_coeff *p_entry = NULL;
+
+ if (mclk % 256) {
+ pr_err("%s mclk must be divisible by 256\n", __func__);
+ return -EINVAL;
+ }
+
+ if (nco_id == 0) {
+ offset = IOP_NCO_0_CONTROL_OFFSET;
+ select = 3;
+ } else {
+ offset = IOP_NCO_1_CONTROL_OFFSET;
+ select = 4;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(nco_clk_coeff); i++) {
+ p_entry = &nco_clk_coeff[i];
+
+ if (p_entry->mclk == mclk) {
+ found = 1;
+ break;
+ }
+ }
+ if (!found) {
+ pr_err("No valid match found in nco_clk_coeff array\n");
+ return -EINVAL;
+ }
+
+ pr_debug("mclk = %d, i = %d, denom = %d\n", mclk, i, p_entry->denom);
+
+ /* Assert Reset rate manager loops */
+ value = readl(audio_base + offset);
+ value |= BIT(IOP_NCO_CONTROL_FREE_RUN);
+ writel(value, audio_base + offset);
+
+ /* set denominator */
+ writel(p_entry->denom, audio_base + offset + 4);
+
+ /* set numerator and sample_inc */
+ value = p_entry->numer << IOP_NCO_NUMERATOR;
+ value |= p_entry->sample_inc;
+ writel(value, audio_base + offset + 8);
+
+ /* set phase_inc */
+ writel(p_entry->phase_inc, audio_base + offset + 12);
+
+ pr_debug("NCO Control = 0x%x\n", readl(audio_base + offset + 0));
+ pr_debug("NCO Denom = 0x%x\n", readl(audio_base + offset + 4));
+ pr_debug("NCO sample_inc = 0x%x\n", readl(audio_base + offset + 8));
+ pr_debug("NCO phase_inc = 0x%x\n", readl(audio_base + offset + 12));
+
+ return select;
+}
+
+static int pll_configure_mclk(void __iomem *audio_base, u32 mclk)
+{
+ int i = 0;
+ bool found = false;
+ const struct pll_macro_entry *p_entry;
+
+ for (i = 0; i < ARRAY_SIZE(pll_predef_mclk); i++) {
+ p_entry = &pll_predef_mclk[i];
+
+ if (p_entry->mclk == mclk) {
+ found = true;
+ break;
+ }
+ }
+ if (!found) {
+ pr_err("%s No valid mclk freq (%u) found!\n", __func__, mclk);
+ return -EINVAL;
+ }
+
+ writel(p_entry->macro, audio_base + IOP_PLL_0_MACRO_OFFSET);
+ pr_debug("PLL MACRO = %d\n", p_entry->macro);
+
+ return p_entry->pll_ch_num;
+}
+
+static int cygnus_ssp_set_clocks(struct cygnus_aio_port *aio)
+{
+ u32 value, i = 0;
+ u32 mask = 0xF;
+ u32 sclk;
+ bool found = false;
+ const struct _ssp_clk_coeff *p_entry = NULL;
+
+ if (!aio->lrclk) {
+ pr_err("First set lrclk through hw_params()\n");
+ return -EINVAL;
+ }
+
+ if (!aio->bitrate) {
+ pr_err("%s Use .set_clkdiv() to set bitrate\n", __func__);
+ return -EINVAL;
+ }
+
+ for (i = 0; i < ARRAY_SIZE(ssp_clk_coeff); i++) {
+ p_entry = &ssp_clk_coeff[i];
+ if ((p_entry->rate == aio->lrclk) &&
+ (p_entry->sclk_rate == aio->bitrate) &&
+ (p_entry->mclk == aio->mclk)) {
+ found = true;
+ break;
+ }
+ }
+ if (!found) {
+ pr_err("No valid match found in ssp_clk_coeff array\n");
+ return -EINVAL;
+ }
+
+ sclk = aio->bitrate;
+ if (sclk == 512)
+ sclk = 0;
+ /* sclks_per_1fs_div = sclk cycles/32 */
+ sclk /= 32;
+ /* Set sclk rate */
+ if (aio->portnum != SPDIF) {
+ /* Set number of bitclks per frame */
+ value = readl(aio->audio + aio->regs.i2s_cfg);
+ value &= ~(mask << I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32);
+ value |= sclk << I2S_OUT_CFGX_SCLKS_PER_1FS_DIV32;
+ writel(value, aio->audio + aio->regs.i2s_cfg);
+ pr_debug("SCLS_PER_1FS_DIV32 = 0x%x\n", value);
+ }
+
+ /* Set MCLK_RATE ssp port (spdif and ssp are the same) */
+ value = readl(aio->audio + aio->regs.i2s_mclk_cfg);
+ value &= ~(0xF << I2S_OUT_MCLKRATE_SHIFT);
+ value |= (p_entry->mclk_rate << I2S_OUT_MCLKRATE_SHIFT);
+ writel(value, aio->audio + aio->regs.i2s_mclk_cfg);
+
+ pr_debug("mclk cfg reg = 0x%x\n", value);
+ pr_debug("bits per frame = %d, mclk = %d Hz, lrclk = %d Hz\n",
+ aio->bitrate, aio->mclk, aio->lrclk);
+ return 0;
+}
+
+static int cygnus_ssp_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+ int rate, value;
+ int ret = 0;
+
+ pr_debug("%s port = %d\n", __func__, aio->portnum);
+ pr_debug("params_channels %d\n", params_channels(params));
+ pr_debug("rate %d\n", params_rate(params));
+ pr_debug("format %d\n", params_format(params));
+
+ rate = params_rate(params);
+ if (aio->mode == CYGNUS_SSPMODE_TDM) {
+ if ((rate == 192000) && (params_channels(params) > 4)) {
+ pr_err("Cannot run %d channels at %dHz\n",
+ params_channels(params), rate);
+ return -EINVAL;
+ }
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ /* Configure channels as mono or stereo */
+ if (params_channels(params) == 1) {
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value |= BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
+ value &= ~BIT(BF_SRC_CFGX_BUFFER_PAIR_ENABLE);
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+ } else {
+ value = readl(aio->audio + aio->regs.bf_sourcech_cfg);
+ value &= ~BIT(BF_SRC_CFGX_SAMPLE_CH_MODE);
+ writel(value, aio->audio + aio->regs.bf_sourcech_cfg);
+ }
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S8:
+ audio_ssp_source_bitres(aio, 8);
+ break;
+
+ case SNDRV_PCM_FORMAT_S16_LE:
+ audio_ssp_source_bitres(aio, 16);
+ break;
+
+ case SNDRV_PCM_FORMAT_S32_LE:
+ case SNDRV_PCM_FORMAT_S24_LE:
+ audio_ssp_source_bitres(aio, 32);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ } else {
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S16_LE:
+ audio_ssp_dst_bitres(aio, 16);
+ break;
+
+ case SNDRV_PCM_FORMAT_S32_LE:
+ case SNDRV_PCM_FORMAT_S24_LE:
+ audio_ssp_dst_bitres(aio, 32);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ }
+
+ aio->lrclk = rate;
+
+ pr_debug("%s clksrc = %d\n", __func__, aio->clksrc);
+
+ switch (aio->clksrc) {
+ case CYGNUS_SSP_CLKSRC_NCO_0:
+ case CYGNUS_SSP_CLKSRC_NCO_1:
+ case CYGNUS_SSP_CLKSRC_PLL:
+ ret = cygnus_ssp_set_clocks(aio);
+ break;
+
+ default:
+ pr_err("clksrc is invalid. Use .set_sysclk to set.\n");
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+/*
+ * This function sets if the ssp should use uses the pll or NCO and will
+ * set the mclk frequency for that clock
+ */
+static int cygnus_ssp_set_sysclk(struct snd_soc_dai *dai,
+ int clk_id, unsigned int freq, int dir)
+{
+ int sel;
+ u32 value;
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+
+ pr_debug("%s Enter port = %d\n", __func__, aio->portnum);
+
+ switch (clk_id) {
+ case CYGNUS_SSP_CLKSRC_NCO_0:
+ sel = nco_configure_mclk(aio->audio, 0, freq);
+ break;
+
+ case CYGNUS_SSP_CLKSRC_NCO_1:
+ sel = nco_configure_mclk(aio->audio, 1, freq);
+ break;
+
+ case CYGNUS_SSP_CLKSRC_PLL:
+ sel = pll_configure_mclk(aio->audio, freq);
+ break;
+
+ default:
+ pr_err("clksrc is not valid\n");
+ return -EINVAL;
+ }
+
+ if (sel < 0) {
+ pr_err("%s Setting mclk failed.\n", __func__);
+ return -EINVAL;
+ }
+
+ aio->mclk = freq;
+ aio->clksrc = clk_id;
+
+ pr_debug("%s Setting MCLKSEL to %d\n", __func__, sel);
+ value = readl(aio->audio + aio->regs.i2s_mclk_cfg);
+ value &= ~(0xF << I2S_OUT_PLLCLKSEL_SHIFT);
+ value |= (sel << I2S_OUT_PLLCLKSEL_SHIFT);
+ writel(value, aio->audio + aio->regs.i2s_mclk_cfg);
+
+ return 0;
+}
+
+static int cygnus_ssp_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+
+ pr_debug("%s port = %d\n", __func__, aio->portnum);
+ snd_soc_dai_set_dma_data(dai, substream, aio);
+
+ return 0;
+}
+
+static int cygnus_ssp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
+ int div_id, int div)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+
+ pr_debug("%s Enter\n", __func__);
+
+ if (div_id != CYGNUS_SSP_FRAMEBITS_DIV)
+ return -EINVAL;
+
+ /* Can only run 64 bits per frame in i2s mode */
+ if (aio->mode == CYGNUS_SSPMODE_I2S)
+ return -EINVAL;
+
+ if ((div != 128) && (div != 256) && (div != 512)) {
+ pr_err("In TDM mode, bits per frame should be 128/256/512\n");
+ return -EINVAL;
+ }
+
+ aio->bitrate = div;
+
+ return 0;
+}
+
+/*
+ * Bit Update Notes
+ * 31 Yes TDM Mode (1 = TDM, 0 = i2s)
+ * 30 Yes Slave Mode (1 = Slave, 0 = Master)
+ * 29:26 No Sclks per frame
+ * 25:18 Yes FS Width
+ * 17:14 No Valid Slots
+ * 13 Yes Bits (1 = 16 bits, 0 = 32 bits)
+ * 12:08 Yes Bits per samp
+ * 07 Yes Justifcation (1 = LSB, 0 = MSB)
+ * 06 Yes Alignment (1 = Delay 1 clk, 0 = no delay
+ * 05 Yes SCLK polarity (1 = Rising, 0 = Falling)
+ * 04 Yes LRCLK Polarity (1 = High for left, 0 = Low for left)
+ * 03:02 Yes Reserved - write as zero
+ * 01 No Data Enable
+ * 00 No CLK Enable
+ */
+#define I2S_IN_CFG_REG_UPDATE_MASK 0x3C03C003
+
+/* Input cfg is same as output, but the FS width is not a valid field */
+#define I2S_OUT_CFG_REG_UPDATE_MASK (I2S_IN_CFG_REG_UPDATE_MASK | 0x03FC0000)
+
+static int cygnus_ssp_set_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+ u32 ssp_curcfg;
+ u32 ssp_newcfg;
+ u32 val;
+ u32 mask;
+
+ pr_debug("%s Enter\n", __func__);
+
+ if (aio->portnum == SPDIF)
+ return -EINVAL;
+
+ ssp_newcfg = 0;
+
+ switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+ /* Set the SSP up as slave */
+ case SND_SOC_DAIFMT_CBM_CFM:
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_SLAVE_MODE);
+ aio->slave = 1;
+ break;
+ /* Set the SSP up as master */
+ case SND_SOC_DAIFMT_CBS_CFS:
+ ssp_newcfg &= ~BIT(I2S_OUT_CFGX_SLAVE_MODE);
+ aio->slave = 0;
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+ case SND_SOC_DAIFMT_RIGHT_J:
+ case SND_SOC_DAIFMT_LEFT_J:
+ return -EINVAL;
+ case SND_SOC_DAIFMT_I2S:
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT);
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_FSYNC_WIDTH);
+ break;
+
+ case SND_SOC_DAIFMT_DSP_A:
+ case SND_SOC_DAIFMT_DSP_B:
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_TDM_MODE);
+
+ /* DSP_A = data after FS, DSP_B = data during FS */
+ if (SND_SOC_DAIFMT_DSP_A)
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_DATA_ALIGNMENT);
+
+ ssp_newcfg |= BIT(I2S_OUT_CFGX_FSYNC_WIDTH);
+ break;
+
+ default:
+ return -EINVAL;
+ }
+
+ /*
+ * SSP out cfg.
+ * Retain bits we do not want to update, then OR in new bits
+ */
+ ssp_curcfg = readl(aio->audio + aio->regs.i2s_cfg);
+ ssp_newcfg = (ssp_curcfg & I2S_IN_CFG_REG_UPDATE_MASK) | ssp_newcfg;
+ writel(ssp_newcfg, aio->audio + aio->regs.i2s_cfg);
+
+ /*
+ * SSP in cfg.
+ * Retain bits we do not want to update, then OR in new bits
+ */
+ ssp_curcfg = readl(aio->audio + aio->regs.i2s_cap_cfg);
+ ssp_newcfg = (ssp_curcfg & I2S_OUT_CFG_REG_UPDATE_MASK) | ssp_newcfg;
+ writel(ssp_newcfg, aio->audio + aio->regs.i2s_cap_cfg);
+
+ if ((0 <= aio->portnum) && (aio->portnum <= 2)) {
+ val = readl(aio->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+
+ /*
+ * Configure the word clk and bit clk as output or tristate
+ * Each port has 4 bits for controlling its pins.
+ * Shift the mask based upon port number.
+ */
+ mask = AUD_MISC_SEROUT_LRCK_OE | AUD_MISC_SEROUT_SCLK_OE;
+ mask = mask << (aio->portnum * 4);
+ if (aio->slave)
+ val |= mask;
+ else
+ val &= ~mask;
+
+ pr_debug("%s Set OE bits 0x%x\n", __func__, val);
+ writel(val, aio->audio + AUD_MISC_SEROUT_OE_REG_BASE);
+ }
+
+ return 0;
+}
+
+static int cygnus_ssp_trigger(struct snd_pcm_substream *substream, int cmd,
+ struct snd_soc_dai *dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+
+ pr_debug("%s cmd %d at port = %d\n", __func__, cmd, aio->portnum);
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ audio_ssp_out_enable(aio, 1);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ audio_ssp_out_enable(aio, 0);
+ break;
+
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ } else {
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ audio_ssp_in_enable(aio, 1);
+ break;
+
+ case SNDRV_PCM_TRIGGER_STOP:
+ audio_ssp_in_enable(aio, 0);
+ break;
+ }
+ }
+
+ return 0;
+}
+
+static int cygnus_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai,
+ unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+ u32 value;
+
+ pr_debug("==> %s\n", __func__);
+
+ if ((slots < 0) || (slots > 16))
+ return -EINVAL;
+
+ /* Slot value must be even */
+ if (slots % 2)
+ return -EINVAL;
+
+ /* We encode 16 slots as 0 in the reg */
+ if (slots == 16)
+ slots = 0;
+
+ value = readl(aio->audio + aio->regs.i2s_cap_cfg);
+ value &= ~(0xF << I2S_OUT_CFGX_VALID_SLOT);
+ value |= (slots << I2S_OUT_CFGX_VALID_SLOT);
+ writel(value, aio->audio + aio->regs.i2s_cap_cfg);
+
+ value = readl(aio->audio + aio->regs.i2s_cfg);
+ value &= ~(0xF << I2S_OUT_CFGX_VALID_SLOT);
+ value |= (slots << I2S_OUT_CFGX_VALID_SLOT);
+ writel(value, aio->audio + aio->regs.i2s_cfg);
+
+ return 0;
+}
+
+int cygnus_ssp_get_mode(struct snd_soc_dai *cpu_dai)
+{
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(cpu_dai);
+
+ return aio->mode;
+}
+EXPORT_SYMBOL(cygnus_ssp_get_mode);
+
+
+static void pll_fract_tweak_set(void __iomem *audio_base, u32 value)
+{
+ /*
+ * Read ACTIVE PLL registers for current values
+ * Write new values to the USER PLL registers
+ * Transition PLL Control to update the active PLL registers with user
+ * PLL registers
+ */
+ u32 ndiv, mdiv0, mdiv1, mdiv2;
+
+ ndiv = readl(audio_base + IOP_PLL_0_ACTIVE_NDIV_OFFSET);
+ mdiv0 = readl(audio_base + IOP_PLL_0_ACTIVE_MDIV_Ch0_OFFSET);
+ mdiv1 = readl(audio_base + IOP_PLL_0_ACTIVE_MDIV_Ch1_OFFSET);
+ mdiv2 = readl(audio_base + IOP_PLL_0_ACTIVE_MDIV_Ch2_OFFSET);
+
+ ndiv &= 0x3ff;
+ ndiv |= value << IOP_PLL_0_USER_NDIV_FRAC;
+
+ writel(7, audio_base + IOP_PLL_0_MACRO_OFFSET);
+ writel(0, audio_base + IOP_PLL_0_CONTROL_OFFSET);
+ writel(mdiv0, audio_base + IOP_PLL_0_MDIV_Ch0_OFFSET);
+ writel(mdiv1, audio_base + IOP_PLL_0_MDIV_Ch1_OFFSET);
+ writel(mdiv2, audio_base + IOP_PLL_0_MDIV_Ch2_OFFSET);
+ writel(ndiv, audio_base + IOP_PLL_0_USER_NDIV_OFFSET);
+ writel(1, audio_base + IOP_PLL_0_CONTROL_OFFSET);
+}
+
+static u32 pll_fract_tweak_get(void __iomem *audio_base)
+{
+ u32 ndiv_fract, ndiv_int, value;
+
+ value = readl(audio_base + IOP_PLL_0_USER_NDIV_OFFSET);
+ ndiv_fract = value >> IOP_PLL_0_USER_NDIV_FRAC;
+ ndiv_int = value & 0x3FF;
+ pr_debug("\nuser fract = %d, user int = %d\n", ndiv_fract, ndiv_int);
+
+ value = readl(audio_base + IOP_PLL_0_ACTIVE_NDIV_OFFSET);
+ ndiv_fract = value >> IOP_PLL_0_ACTIVE_NDIV_FRAC;
+ ndiv_int = value & 0x3FF;
+ pr_debug("\nactive fract = %d, active int = %d\n",
+ ndiv_fract, ndiv_int);
+ pr_debug("\nuser mdiv0 = %d, mdiv1 = %d, mdiv2 = %d\n",
+ readl(audio_base + IOP_PLL_0_MDIV_Ch0_OFFSET),
+ readl(audio_base + IOP_PLL_0_MDIV_Ch1_OFFSET),
+ readl(audio_base + IOP_PLL_0_MDIV_Ch2_OFFSET));
+
+ pr_debug("\nactive mdiv0 = %d, mdiv1 = %d, mdiv2 = %d\n",
+ readl(audio_base + IOP_PLL_0_ACTIVE_MDIV_Ch0_OFFSET),
+ readl(audio_base + IOP_PLL_0_ACTIVE_MDIV_Ch1_OFFSET),
+ readl(audio_base + IOP_PLL_0_ACTIVE_MDIV_Ch2_OFFSET));
+
+ return ndiv_fract;
+}
+
+/*
+ * pll_tweak_get - read the pll fractional setting.
+ * kcontrol: The control for the speaker gain.
+ * ucontrol: The value that needs to be updated.
+ */
+static int pll_tweak_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol);
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+
+ pr_debug("Enter %s\n", __func__);
+
+ ucontrol->value.integer.value[0] = pll_fract_tweak_get(aio->audio);
+
+ return 0;
+}
+
+/*
+ * pll_tweak_put - set the pll fractional setting.
+ * kcontrol: The control for the pll tweak.
+ * ucontrol: The value that needs to be set.
+ */
+static int pll_tweak_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_dai *dai = snd_kcontrol_chip(kcontrol);
+ struct cygnus_aio_port *aio = cygnus_dai_get_portinfo(dai);
+ int value;
+
+ value = ucontrol->value.integer.value[0];
+ if (value > PLL_NDIV_FRACT_MAX) {
+ pr_err("Invalid range. (0 - %lu)\n", PLL_NDIV_FRACT_MAX);
+ return -EINVAL;
+ }
+
+ pr_debug("Enter %s with value %d\n", __func__, value);
+ pll_fract_tweak_set(aio->audio, value);
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new pll_tweak_controls[] = {
+ SOC_SINGLE_EXT("PLL Tweak", 0, 0, PLL_NDIV_FRACT_MAX, 0,
+ pll_tweak_get, pll_tweak_put),
+};
+
+int cygnus_ssp_add_pll_tweak_controls(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+
+ return snd_soc_add_dai_controls(cpu_dai,
+ pll_tweak_controls,
+ ARRAY_SIZE(pll_tweak_controls));
+}
+EXPORT_SYMBOL(cygnus_ssp_add_pll_tweak_controls);
+
+
+static const struct snd_soc_dai_ops cygnus_ssp_dai_ops = {
+ .startup = cygnus_ssp_startup,
+ .trigger = cygnus_ssp_trigger,
+ .hw_params = cygnus_ssp_hw_params,
+ .set_fmt = cygnus_ssp_set_fmt,
+ .set_sysclk = cygnus_ssp_set_sysclk,
+ .set_clkdiv = cygnus_ssp_dai_set_clkdiv,
+ .set_tdm_slot = cygnus_set_dai_tdm_slot,
+};
+
+static struct snd_soc_dai_driver cygnus_tdm_dai_template = {
+ .playback = {
+ .channels_min = 2,
+ .channels_max = 16,
+ .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 16,
+ .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &cygnus_ssp_dai_ops,
+};
+
+static struct snd_soc_dai_driver cygnus_i2s_dai_template = {
+ .playback = {
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S8 |
+ SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .capture = {
+ .channels_min = 2,
+ .channels_max = 2,
+ .rates = CYGNUS_TDM_RATE | SNDRV_PCM_RATE_88200 |
+ SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 |
+ SNDRV_PCM_RATE_192000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S32_LE,
+ },
+ .ops = &cygnus_ssp_dai_ops,
+};
+
+static struct snd_soc_dai_driver cygnus_ssp_dai[CYGNUS_MAX_PORTS];
+
+static const struct snd_soc_component_driver cygnus_ssp_component = {
+ .name = "cygnus-audio",
+};
+
+static const struct of_device_id cygnus_ssp_of_match[] = {
+ { .compatible = "brcm,cygnus-audio" },
+ {},
+};
+/*
+ * Return < 0 if error
+ * Return 0 if disabled
+ * Return 1 if enabled and node is parsed successfully
+ */
+static int parse_ssp_child_node(struct platform_device *pdev,
+ struct device_node *dn,
+ struct cygnus_audio *cygaud,
+ struct snd_soc_dai_driver *p_dai)
+{
+ struct cygnus_aio_port *aio;
+ const char *mode;
+ const char *channel_grp;
+ const char *port_status;
+ struct cygnus_ssp_regs ssp_regs[3];
+ int rawval;
+ int portnum = -1;
+ int frame_bits;
+
+ if ((of_property_read_string(dn, "port-status", &port_status)) != 0) {
+ dev_err(&pdev->dev, "Missing port-status property\n");
+ return -EINVAL;
+ }
+
+ /* If not enabled return 1 */
+ if (strcmp(port_status, "enabled") != 0)
+ return 1;
+
+ if (of_property_read_u32(dn, "ssp-port-id", &rawval) != 0) {
+ dev_err(&pdev->dev, "%s: invalid ssp-port-id\n", __func__);
+ return -EINVAL;
+ }
+
+ if (rawval == 0)
+ portnum = I2S0;
+ else if (rawval == 1)
+ portnum = I2S1;
+ else if (rawval == 2)
+ portnum = I2S2;
+ else if (rawval == 3)
+ portnum = SPDIF;
+ else
+ return -EINVAL;
+
+ if ((of_property_read_string(dn, "mode", &mode)) != 0) {
+ dev_err(&pdev->dev, "Missing mode property\n");
+ return -EINVAL;
+ }
+
+ if (of_property_read_string(dn, "channel-group", &channel_grp) != 0) {
+ dev_err(&pdev->dev, "Missing channel_group property\n");
+ return -EINVAL;
+ }
+
+ aio = &cygaud->portinfo[portnum];
+
+ aio->audio = cygaud->audio;
+ aio->clksrc = -1;
+ aio->portnum = portnum;
+
+ if ((portnum == I2S0) || (portnum == I2S1) || (portnum == I2S2)) {
+ ssp_regs[I2S0] = (struct cygnus_ssp_regs) INIT_SSP_REGS(0);
+ ssp_regs[I2S1] = (struct cygnus_ssp_regs) INIT_SSP_REGS(1);
+ ssp_regs[I2S2] = (struct cygnus_ssp_regs) INIT_SSP_REGS(2);
+
+ aio->regs = ssp_regs[portnum];
+
+ if (strstr(mode, "i2s")) {
+ *p_dai = cygnus_i2s_dai_template;
+ aio->mode = CYGNUS_SSPMODE_I2S;
+ } else if (strstr(mode, "tdm")) {
+ *p_dai = cygnus_tdm_dai_template;
+ aio->mode = CYGNUS_SSPMODE_TDM;
+ } else {
+ return -EINVAL;
+ }
+
+ } else { /* SPDIF case */
+ aio->regs.bf_sourcech_cfg = BF_SRC_CFG3_OFFSET;
+ aio->regs.bf_sourcech_ctrl = BF_SRC_CTRL3_OFFSET;
+ aio->regs.i2s_mclk_cfg = SPDIF_MCLK_CFG_OFFSET;
+ aio->regs.i2s_stream_cfg = SPDIF_STREAM_CFG_OFFSET;
+
+ *p_dai = cygnus_i2s_dai_template;
+
+ /* For the purposes of this code SPDIF can be I2S mode */
+ aio->mode = CYGNUS_SSPMODE_I2S;
+ }
+
+ if (of_property_read_string(dn, "dai-name", &p_dai->name) != 0) {
+ dev_err(&pdev->dev, "Missing dai-name property\n");
+ return -EINVAL;
+ }
+
+ if (aio->mode == CYGNUS_SSPMODE_TDM) {
+ const char *propname = "tdm-bits-per-frame";
+
+ if (of_property_read_u32(dn, propname, &frame_bits)) {
+ dev_err(&pdev->dev, "%s: %s not found\n",
+ __func__, propname);
+ return -EINVAL;
+ }
+
+ if ((frame_bits != 128) && (frame_bits != 256)
+ && (frame_bits != 512)) {
+ dev_err(&pdev->dev, "In TDM mode, frame bits should be 128/256/512\n");
+ return -EINVAL;
+ }
+
+ aio->bitrate = frame_bits;
+ } else {
+ aio->bitrate = 64; /* I2S must be 64 */
+ }
+
+ /* Handle the channel grouping */
+ if (portnum == I2S0) {
+ if (strstr(channel_grp, "2_0")) {
+ group_id[portnum] = 0;
+ aio->channel_grouping = 0x1;
+ } else if (strstr(channel_grp, "3_1")) {
+ group_id[portnum] = 0;
+ aio->channel_grouping = 0x3;
+ } else if (strstr(channel_grp, "5_1")) {
+ group_id[portnum] = 0;
+ aio->channel_grouping = 0x7;
+ } else {
+ dev_err(&pdev->dev, "Invalid channel grouping\n");
+ return -EINVAL;
+ }
+ }
+ if (portnum == I2S1) {
+ if (strstr(channel_grp, "2_0")) {
+ group_id[portnum] = 1;
+ aio->channel_grouping = 0x1;
+ } else if (strstr(channel_grp, "3_1")) {
+ group_id[portnum] = 0;
+ aio->channel_grouping = 0x3;
+ } else if (strstr(channel_grp, "5_1")) {
+ group_id[portnum] = 0;
+ aio->channel_grouping = 0x7;
+ } else {
+ dev_err(&pdev->dev, "Invalid channel grouping\n");
+ return -EINVAL;
+ }
+ }
+ if (portnum == I2S2) {
+ if (strstr(channel_grp, "2_0")) {
+ group_id[I2S2] = 2;
+ aio->channel_grouping = 0x1;
+ } else if (strstr(channel_grp, "5_1")) {
+ group_id[I2S2] = 0;
+ aio->channel_grouping = 0x7;
+ } else {
+ dev_err(&pdev->dev, "Invalid channel grouping\n");
+ return -EINVAL;
+ }
+ }
+ if (portnum == SPDIF) {
+ group_id[SPDIF] = 3;
+ aio->channel_grouping = 0x1;
+ }
+
+ dev_dbg(&pdev->dev, "%s portnum = %d\n", __func__, aio->portnum);
+ aio->streams_on = 0;
+ audio_ssp_init_portregs(aio);
+ return 0;
+}
+
+static int cygnus_ssp_probe(struct platform_device *pdev)
+{
+ struct device *dev = &pdev->dev;
+ struct device_node *child_node;
+ struct resource *res = pdev->resource;
+ struct cygnus_audio *cygaud;
+ int err = -EINVAL;
+ int node_count;
+ int active_port_count;
+
+ if (!of_match_device(cygnus_ssp_of_match, dev)) {
+ dev_err(dev, "Failed to find ssp controller\n");
+ return -ENODEV;
+ }
+
+ cygaud = devm_kzalloc(dev, sizeof(struct cygnus_audio), GFP_KERNEL);
+ if (!cygaud)
+ return -ENOMEM;
+
+ dev_set_drvdata(dev, cygaud);
+
+ res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ cygaud->audio = devm_ioremap_resource(dev, res);
+ if (IS_ERR(cygaud->audio)) {
+ dev_err(dev, "audio_io ioremap failed\n");
+ return PTR_ERR(cygaud->audio);
+ }
+
+ audio_pll0_init(cygaud->audio);
+
+ node_count = 0;
+ node_count = of_get_child_count(pdev->dev.of_node);
+ if ((node_count < 1) || (node_count > CYGNUS_MAX_PORTS)) {
+ dev_err(dev, "Incorrct number of child nodes\n");
+ return -EINVAL;
+ }
+
+ active_port_count = 0;
+ for_each_available_child_of_node(pdev->dev.of_node, child_node) {
+ err = parse_ssp_child_node(pdev, child_node, cygaud,
+ &cygnus_ssp_dai[active_port_count]);
+
+ /* negative is err, 0 is active and good, 1 is disabled */
+ if (err < 0)
+ return err;
+ else if (err == 0)
+ active_port_count++;
+ }
+
+ dev_dbg(dev, "Registering %d DAIs\n", active_port_count);
+ err = devm_snd_soc_register_component(dev, &cygnus_ssp_component,
+ cygnus_ssp_dai, active_port_count);
+ if (err) {
+ dev_err(dev, "snd_soc_register_dai failed\n");
+ return err;
+ }
+
+ cygaud->irq_num = platform_get_irq(pdev, 0);
+ if (cygaud->irq_num <= 0) {
+ dev_err(dev, "platform_get_irq failed\n");
+ return cygaud->irq_num;
+ }
+
+ err = cygnus_soc_platform_register(dev, cygaud);
+ if (err) {
+ dev_err(dev, "platform reg error %d\n", err);
+ return err;
+ }
+
+ return 0;
+}
+
+static int cygnus_ssp_remove(struct platform_device *pdev)
+{
+ cygnus_soc_platform_unregister(&pdev->dev);
+
+ return 0;
+}
+
+static struct platform_driver cygnus_ssp_driver = {
+ .probe = cygnus_ssp_probe,
+ .remove = cygnus_ssp_remove,
+ .driver = {
+ .name = "cygnus-ssp",
+ .of_match_table = cygnus_ssp_of_match,
+ },
+};
+
+module_platform_driver(cygnus_ssp_driver)
+
+MODULE_LICENSE("GPL v2");
+MODULE_AUTHOR("Broadcom");
+MODULE_DESCRIPTION("Cygnus ASoC SSP Interface");
diff --git a/sound/soc/bcm/cygnus-ssp.h b/sound/soc/bcm/cygnus-ssp.h
new file mode 100644
index 0000000..d11951c
--- /dev/null
+++ b/sound/soc/bcm/cygnus-ssp.h
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2014-2015 Broadcom Corporation
+ *
+ * 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 __CYGNUS_SSP_H__
+#define __CYGNUS_SSP_H__
+
+#include "cygnus-pcm.h"
+
+#define CYGNUS_TDM_DAI_MAX_SLOTS 16
+
+#define CYGNUS_MAX_PLAYBACK_PORTS 4
+#define CYGNUS_MAX_CAPTURE_PORTS 3
+#define CYGNUS_MAX_PORTS CYGNUS_MAX_PLAYBACK_PORTS
+
+#define CYGNUS_SSP_FRAMEBITS_DIV 1
+
+#define CYGNUS_SSPMODE_I2S 0
+#define CYGNUS_SSPMODE_TDM 1
+
+#define CYGNUS_SSP_CLKSRC_PLL 0
+#define CYGNUS_SSP_CLKSRC_NCO_0 1
+#define CYGNUS_SSP_CLKSRC_NCO_1 2
+
+struct cygnus_ssp_regs {
+ u32 i2s_stream_cfg;
+ u32 i2s_cfg;
+ u32 i2s_cap_stream_cfg;
+ u32 i2s_cap_cfg;
+ u32 i2s_mclk_cfg;
+
+ u32 bf_destch_ctrl;
+ u32 bf_destch_cfg;
+ u32 bf_sourcech_ctrl;
+ u32 bf_sourcech_cfg;
+};
+
+struct cygnus_aio_port {
+ int portnum;
+ int mode;
+ bool slave; /* 0 = master mode, 1 = slave mode */
+ int streams_on; /* will be 0 if both capture and play are off */
+ int channel_grouping;
+ int clksrc;
+
+ u32 mclk;
+ u32 lrclk;
+ u32 bitrate;
+
+ void __iomem *audio;
+
+ struct cygnus_ssp_regs regs;
+
+ struct ringbuf_regs play_rb_regs;
+ struct ringbuf_regs capture_rb_regs;
+
+ struct snd_pcm_substream *play_stream;
+ struct snd_pcm_substream *capture_stream;
+};
+
+
+struct cygnus_audio {
+ struct cygnus_aio_port portinfo[CYGNUS_MAX_PORTS];
+
+ int irq_num;
+ void __iomem *audio;
+};
+
+extern int cygnus_ssp_get_mode(struct snd_soc_dai *cpu_dai);
+extern int cygnus_ssp_add_pll_tweak_controls(struct snd_soc_pcm_runtime *rtd);
+extern int cygnus_soc_platform_register(struct device *dev,
+ struct cygnus_audio *cygaud);
+extern int cygnus_soc_platform_unregister(struct device *dev);
+
+
+#endif
--
2.3.3
--
To unsubscribe from this list: send the line "unsubscribe linux-kernel" in
the body of a message to majordomo@...r.kernel.org
More majordomo info at http://vger.kernel.org/majordomo-info.html
Please read the FAQ at http://www.tux.org/lkml/
Powered by blists - more mailing lists