* [PATCH 0/2] Cygnus Audio Driver
@ 2015-03-31 3:16 Scott Branden
2015-03-31 3:16 ` [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings Scott Branden
` (2 more replies)
0 siblings, 3 replies; 20+ messages in thread
From: Scott Branden @ 2015-03-31 3:16 UTC (permalink / raw)
To: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Mark Brown, Jaroslav Kysela, Takashi Iwai,
alsa-devel
Cc: linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, linux-rpi-kernel, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin, Scott Branden
Please provide comments on the initial version of this driver.
This patchset contains the devicetree bindings and core audio driver
for the Cygnus SoC.
There is an open question on how to fit this driver into the clock
framework (if at all).
The audio PLL is embedded in the audio block and only used
by the audio block. The audio PLL registers are also in the middle of
the audio register map.
In addition, the audio PLL is adjustable to less than 1 Hz.
The existing clock driver framework does not provide a mechanism to take
advantage of the resolution of the hardware.
Can the audio PLL remain within the audio driver and/or any modifications
required?
Lori Hikichi (2):
ASoC: cygnus-audio: adding device tree bindings
ASoC: add core audio driver for Broadcom Cygnus SOC.
.../bindings/sound/brcm,cygnus-audio.txt | 68 +
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 +
7 files changed, 2743 insertions(+), 1 deletion(-)
create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
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
--
2.3.3
^ permalink raw reply [flat|nested] 20+ messages in thread
* [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings
2015-03-31 3:16 [PATCH 0/2] Cygnus Audio Driver Scott Branden
@ 2015-03-31 3:16 ` Scott Branden
2015-03-31 5:58 ` Mark Brown
` (2 more replies)
2015-03-31 3:16 ` [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC Scott Branden
2015-03-31 6:43 ` [PATCH 0/2] Cygnus Audio Driver Mark Brown
2 siblings, 3 replies; 20+ messages in thread
From: Scott Branden @ 2015-03-31 3:16 UTC (permalink / raw)
To: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Mark Brown, Jaroslav Kysela, Takashi Iwai,
alsa-devel
Cc: linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, linux-rpi-kernel, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin, Lori Hikichi,
Scott Branden
From: Lori Hikichi <lhikichi@broadcom.com>
Add device tree binding documentation for the Cygnus SOC audio block
Reviewed-by: Jonathan Richardson <jonathar@broadcom.com>
Reviewed-by: Ray Jui <rjui@broadcom.com>
Signed-off-by: Scott Branden <sbranden@broadcom.com>
Signed-off-by: Lori Hikichi <lhikichi@broadcom.com>
---
.../bindings/sound/brcm,cygnus-audio.txt | 68 ++++++++++++++++++++++
1 file changed, 68 insertions(+)
create mode 100644 Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
diff --git a/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
new file mode 100644
index 0000000..5358cc3
--- /dev/null
+++ b/Documentation/devicetree/bindings/sound/brcm,cygnus-audio.txt
@@ -0,0 +1,68 @@
+BROADCOM Cygnus Audio I2S/TDM/SPDIF controller
+
+Required properties:
+ - compatible : "brcm,cygnus-audio"
+ - #address-cells: 32bit valued, 1 cell. <1>
+ - #size-cells: 32bit valued, 1 cell. <1>
+ - reg : Should contain audio registers location and length
+ - interrupts: audio DMA interrupt number
+
+SSP Subnode properties:
+- dai-name: The name of the DAI registered with ASOC
+- ssp-port-id: The ssp port interface to use
+ Valid value are 0, 1, 2, or 3 (for spdif)
+- mode: Controls if this port should be configured as I2S or TDM mode.
+ Valid values are: "tdm" or "i2s"
+- tdm-bits-per-frame: only if mode is "tdm" then this property must set.
+ Valid values are (128/256/512)
+- port-status: Controls if port is enabled or not
+ Valid values "enabled" or "disabled"
+- channel-group: Control grouping of serial port
+ Valid values are "2_0", "3_1", or "5_1"
+
+Example:
+ cygnus_audio: audio@0x180ae000 {
+ compatible = "brcm,cygnus-audio";
+ #address-cells = <1>;
+ #size-cells = <1>;
+ reg = <0x180ae000 0x1000>;
+
+ interrupts = <GIC_SPI 143 IRQ_TYPE_LEVEL_HIGH>;
+
+ ssp0: ssp_port@0 {
+ dai-name = "cygnus-ssp0";
+ ssp-port-id = <0>;
+
+ mode = "i2s"; /* "i2s" or "tdm" */
+ channel-group = "2_0"; /* Use 2_0, 3_1, 5_1 */
+ port-status = "enabled";
+ };
+
+ ssp1: ssp_port@1 {
+ dai-name = "cygnus-ssp1";
+ ssp-port-id = <1>;
+
+ mode = "i2s"; /* "i2s" or "tdm" */
+ channel-group = "2_0"; /* Use 2_0, 3_1, 5_1 */
+ port-status = "disabled";
+ };
+
+ ssp2: ssp_port@2 {
+ dai-name = "cygnus-ssp2";
+ ssp-port-id = <2>;
+
+ mode = "tdm"; /* "i2s" or "tdm" */
+ tdm-bits-per-frame = <256>;
+ channel-group = "2_0"; /* Use 2_0, 3_1, 5_1 */
+ port-status = "disabled";
+ };
+
+ spdif: spdif_port@3 {
+ dai-name = "cygnus-spdif";
+ ssp-port-id = <3>;
+
+ mode = "i2s"; /* "i2s" or "tdm" */
+ channel-group = "2_0"; /* Use 2_0, 3_1, 5_1 */
+ port-status = "disabled";
+ };
+ };
--
2.3.3
^ permalink raw reply related [flat|nested] 20+ messages in thread
* [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
2015-03-31 3:16 [PATCH 0/2] Cygnus Audio Driver Scott Branden
2015-03-31 3:16 ` [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings Scott Branden
@ 2015-03-31 3:16 ` Scott Branden
2015-03-31 6:42 ` Mark Brown
2015-03-31 6:43 ` [PATCH 0/2] Cygnus Audio Driver Mark Brown
2 siblings, 1 reply; 20+ messages in thread
From: Scott Branden @ 2015-03-31 3:16 UTC (permalink / raw)
To: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Mark Brown, Jaroslav Kysela, Takashi Iwai,
alsa-devel
Cc: linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, linux-rpi-kernel, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin, Lori Hikichi,
Scott Branden
From: Lori Hikichi <lhikichi@broadcom.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@broadcom.com>
Signed-off-by: Lori Hikichi <lhikichi@broadcom.com>
Signed-off-by: Scott Branden <sbranden@broadcom.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
^ permalink raw reply related [flat|nested] 20+ messages in thread
* Re: [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings
2015-03-31 3:16 ` [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings Scott Branden
@ 2015-03-31 5:58 ` Mark Brown
2015-04-02 18:47 ` Lori Hikichi
2015-03-31 7:00 ` Mark Brown
2015-03-31 7:26 ` [alsa-devel] " Lars-Peter Clausen
2 siblings, 1 reply; 20+ messages in thread
From: Mark Brown @ 2015-03-31 5:58 UTC (permalink / raw)
To: Scott Branden
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, linux-rpi-kernel, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin, Lori Hikichi
[-- Attachment #1: Type: text/plain, Size: 1124 bytes --]
On Mon, Mar 30, 2015 at 08:16:23PM -0700, Scott Branden wrote:
> +SSP Subnode properties:
> +- dai-name: The name of the DAI registered with ASOC
ASoC.
Why is this in the DT - it sounds like this is just an internal
implementation detail for Linux, not a property of the hardware.
> +- mode: Controls if this port should be configured as I2S or TDM mode.
> + Valid values are: "tdm" or "i2s"
> +- tdm-bits-per-frame: only if mode is "tdm" then this property must set.
> + Valid values are (128/256/512)
We'd normally leave these up to the machine driver to set as they're
link wide things for system integration. The bits per frame in
particular looks like something that's not going to be fixed by the
hardware and could be varied at runtime.
> +- port-status: Controls if port is enabled or not
> + Valid values "enabled" or "disabled"
This sounds like it's replicating the DT standard status property?
> +- channel-group: Control grouping of serial port
> + Valid values are "2_0", "3_1", or "5_1"
What does this mean? It looks like it's setting the number of channels
which again seems like a runtime thing.
[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 473 bytes --]
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
2015-03-31 3:16 ` [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC Scott Branden
@ 2015-03-31 6:42 ` Mark Brown
2015-04-02 18:47 ` Lori Hikichi
0 siblings, 1 reply; 20+ messages in thread
From: Mark Brown @ 2015-03-31 6:42 UTC (permalink / raw)
To: Scott Branden
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, linux-rpi-kernel, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin, Lori Hikichi
[-- Attachment #1: Type: text/plain, Size: 9315 bytes --]
On Mon, Mar 30, 2015 at 08:16:24PM -0700, Scott Branden wrote:
The CC list for this patch is pretty wide - please look at who you're
sending this to and try to send to only relevant people (for example I'm
not sure the Raspberry Pi people need to review this). People working
upstream get a lot of mail so it's helpful to avoid filling their
inboxes with random irrelevant stuff.
> 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(-)
Send the DMA and DAI drivers as separate patches please, it makes review
easier.
> +config SND_SOC_CYGNUS
> + tristate "SoC platform audio for Broadcom Cygnus chips"
> + depends on ARCH_BCM_CYGNUS || COMPILE_TEST
> + default ARCH_BCM_CYGNUS
Remove the default setting here - we don't do this for other drivers, we
shouldn't do it here.
> +/*
> + * 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
This sounds like it's a setting rather than actually the minimum?
> +static const struct snd_pcm_hardware cygnus_pcm_hw = {
> + .info = SNDRV_PCM_INFO_MMAP |
> + SNDRV_PCM_INFO_MMAP_VALID |
> + SNDRV_PCM_INFO_INTERLEAVED,
The DMA controller is integrated into the IP?
> +static int enable_count;
This looks bogus - why is this a global variable not part of the device
struct and if it does need to be global why does it need no locking?
> + 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;
This looks like a switch statement, there are many places in the code
where you're writing switch statements as chains of ifs.
> +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);
> +}
So it looks like we're getting an interrupt per period and we have to
manually advance to the next one?
> +static irqreturn_t cygnus_dma_irq(int irq, void *data)
> +{
> + u32 r5_status;
> + struct cygnus_audio *cygaud;
> +
> + cygaud = (struct cygnus_audio *)data;
If you need to cast away from void something is very wrong.
> + /*
> + * 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);
This feels racy especially given that we seem to need every interrupt
delivering. What if another period happens during the servicing? I
don't understand what "overwriting esr_status" means here.
> + return IRQ_HANDLED;
If neither playback nor capture interrupts were flagged we should return
IRQ_NONE.
> +/*
> + * 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)
This is obviously not suitable for mainline - "let's just cut'n'paste
the shared code in case we want to add diagnostics in future" does not
scale, you can always add local diagnostics in the core code.
> +};
> +/*
Blank line between these two please. Missing blanks between bits of
code seem to be a common issue in this driver.
> +static int audio_ssp_in_enable(struct cygnus_aio_port *aio, bool enable)
> +{
> + u32 value;
> +
> + if (enable) {
> + } else {
Why not just write two functions?
> +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;
dev_err() (this applies throughout the driver).
It's not clear why this is a function and not just written as a single
statement in the one place that it's used (there are multiple calls but
they're all together and could just be inlined).
> + if (!aio->bitrate) {
> + pr_err("%s Use .set_clkdiv() to set bitrate\n", __func__);
> + return -EINVAL;
> + }
This seems bogus - why are we enforcing the use of set_clkdiv()? Can't
we figure out something esneible?
> + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
> + /* Set the SSP up as slave */
> + case SND_SOC_DAIFMT_CBM_CFM:
The comments here look odd due to the indentation and really aren't
adding much anyway.
> + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
> + case SND_SOC_DAIFMT_RIGHT_J:
> + case SND_SOC_DAIFMT_LEFT_J:
> + return -EINVAL;
These are just eaten by the default case.
> + 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);
That if statement isn't doing what was intended...
> + } 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;
> + }
We can just ignore other triggers? It's not clear why this is different
to the playback case.
> +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);
What is this for, it's setting off alarm bells? Note also that ASoC is
_GPL() only.
> +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),
> +};
Whatever this control is doing it should be a separate patch (as I think
we discussed in person a ELC?), it's clearly something that's unusual
and is likely to block the rest of the code as a result. At a high
level this needs at least some documentation.
> +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);
Again, why is this being exported and why is it not _GPL? If the driver
is adding controls I'd expect it to just add its controls itself.
> +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;
> + }
This error message is misleading - you mean to say that the driver got
loaded for a device it doesn't understand.
> + 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);
devm_ioremap_resource() will complain for you, and in general you should
be printing error codes.
> + node_count = 0;
This doesn't seem to be needed?
> + 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;
Spelling mistake there and it would be helpful to the user to tell them
what we parsed.
[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 473 bytes --]
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 0/2] Cygnus Audio Driver
2015-03-31 3:16 [PATCH 0/2] Cygnus Audio Driver Scott Branden
2015-03-31 3:16 ` [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings Scott Branden
2015-03-31 3:16 ` [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC Scott Branden
@ 2015-03-31 6:43 ` Mark Brown
2015-04-03 19:33 ` Scott Branden
2 siblings, 1 reply; 20+ messages in thread
From: Mark Brown @ 2015-03-31 6:43 UTC (permalink / raw)
To: Scott Branden
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, linux-rpi-kernel, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
[-- Attachment #1: Type: text/plain, Size: 383 bytes --]
On Mon, Mar 30, 2015 at 08:16:22PM -0700, Scott Branden wrote:
> The audio PLL is embedded in the audio block and only used
> by the audio block. The audio PLL registers are also in the middle of
> the audio register map.
When you say it's only used by the audio block do you mean to say that
the audio block exposes no clock signals other than the bit and frame
clocks?
[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 473 bytes --]
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings
2015-03-31 3:16 ` [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings Scott Branden
2015-03-31 5:58 ` Mark Brown
@ 2015-03-31 7:00 ` Mark Brown
2015-03-31 7:26 ` [alsa-devel] " Lars-Peter Clausen
2 siblings, 0 replies; 20+ messages in thread
From: Mark Brown @ 2015-03-31 7:00 UTC (permalink / raw)
To: Scott Branden
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, linux-rpi-kernel, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin, Lori Hikichi
[-- Attachment #1: Type: text/plain, Size: 502 bytes --]
On Mon, Mar 30, 2015 at 08:16:23PM -0700, Scott Branden wrote:
> +Required properties:
> + - compatible : "brcm,cygnus-audio"
> + - #address-cells: 32bit valued, 1 cell. <1>
> + - #size-cells: 32bit valued, 1 cell. <1>
> + - reg : Should contain audio registers location and length
> + - interrupts: audio DMA interrupt number
> +
> +SSP Subnode properties:
This document does rather gloss over the fact that the binding requires
subnodes, it should mention this requirement and say what they mean.
[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 473 bytes --]
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [alsa-devel] [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings
2015-03-31 3:16 ` [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings Scott Branden
2015-03-31 5:58 ` Mark Brown
2015-03-31 7:00 ` Mark Brown
@ 2015-03-31 7:26 ` Lars-Peter Clausen
2015-04-02 22:44 ` Lori Hikichi
2 siblings, 1 reply; 20+ messages in thread
From: Lars-Peter Clausen @ 2015-03-31 7:26 UTC (permalink / raw)
To: Scott Branden, Rob Herring, Pawel Moll, Mark Rutland,
Ian Campbell, Kumar Gala, Liam Girdwood, Mark Brown,
Jaroslav Kysela, Takashi Iwai, alsa-devel
Cc: devicetree, Lori Hikichi, linux-kernel, bryeung, abrestic,
bcm-kernel-feedback-list, linux-arm-kernel, pwestin,
Anatol Pomazao, olofj, Dmitry Torokhov, linux-rpi-kernel
On 03/31/2015 05:16 AM, Scott Branden wrote:
[...]
> +- ssp-port-id: The ssp port interface to use
> + Valid value are 0, 1, 2, or 3 (for spdif)
How about using 'reg' as the property name here. It is the standard property
name for identifying or assigning an address to a sub-node.
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings
2015-03-31 5:58 ` Mark Brown
@ 2015-04-02 18:47 ` Lori Hikichi
0 siblings, 0 replies; 20+ messages in thread
From: Lori Hikichi @ 2015-04-02 18:47 UTC (permalink / raw)
To: Mark Brown, Scott Branden
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, Dmitry Torokhov, Anatol Pomazao,
abrestic, bryeung, olofj, pwestin
On 15-03-30 10:58 PM, Mark Brown wrote:
> On Mon, Mar 30, 2015 at 08:16:23PM -0700, Scott Branden wrote:
>
>> +SSP Subnode properties:
>> +- dai-name: The name of the DAI registered with ASOC
>
> ASoC.
>
Okay.
> Why is this in the DT - it sounds like this is just an internal
> implementation detail for Linux, not a property of the hardware.
>
Will move into the driver code.
>> +- mode: Controls if this port should be configured as I2S or TDM mode.
>> + Valid values are: "tdm" or "i2s"
>> +- tdm-bits-per-frame: only if mode is "tdm" then this property must set.
>> + Valid values are (128/256/512)
>
> We'd normally leave these up to the machine driver to set as they're
> link wide things for system integration. The bits per frame in
> particular looks like something that's not going to be fixed by the
> hardware and could be varied at runtime.
>
I was using the device tree to set some board specific properties of our
audio serial port driver. The idea was that these properties would only
need to be one value for a specific board. Therefore, we could set them
in the device tree and then not have to set them in the
machine file. I thought it would help keep the machine file
a little more basic. Moving these to the machine file is reasonable and
will allow for a more dynamic usage of our driver. I will move them to
machine file.
>> +- port-status: Controls if port is enabled or not
>> + Valid values "enabled" or "disabled"
>
> This sounds like it's replicating the DT standard status property?
>
Okay. Will replace with "status"
>> +- channel-group: Control grouping of serial port
>> + Valid values are "2_0", "3_1", or "5_1"
>
> What does this mean? It looks like it's setting the number of channels
> which again seems like a runtime thing.
>
It is a configuration where we can link together two or three of the
ports to allow for synchronized starts and stop. I will look into
another method of configuring this operational mode.
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
2015-03-31 6:42 ` Mark Brown
@ 2015-04-02 18:47 ` Lori Hikichi
2015-04-06 16:19 ` Mark Brown
0 siblings, 1 reply; 20+ messages in thread
From: Lori Hikichi @ 2015-04-02 18:47 UTC (permalink / raw)
To: Mark Brown, Scott Branden
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, Dmitry Torokhov, Anatol Pomazao,
abrestic, bryeung, olofj, pwestin
On 15-03-30 11:42 PM, Mark Brown wrote:
> On Mon, Mar 30, 2015 at 08:16:24PM -0700, Scott Branden wrote:
>
> The CC list for this patch is pretty wide - please look at who you're
> sending this to and try to send to only relevant people (for example I'm
> not sure the Raspberry Pi people need to review this). People working
> upstream get a lot of mail so it's helpful to avoid filling their
> inboxes with random irrelevant stuff.
>
>> 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(-)
>
> Send the DMA and DAI drivers as separate patches please, it makes review
> easier.
>
>> +config SND_SOC_CYGNUS
>> + tristate "SoC platform audio for Broadcom Cygnus chips"
>> + depends on ARCH_BCM_CYGNUS || COMPILE_TEST
>> + default ARCH_BCM_CYGNUS
>
Okay.
> Remove the default setting here - we don't do this for other drivers, we
> shouldn't do it here.
>
>> +/*
>> + * 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
>
> This sounds like it's a setting rather than actually the minimum?
>
It is a bad comment. I will update. This is the minimum period (in
bytes) at which the interrupt can tick. Other possible value for the
period must be multiple of this value.
>> +static const struct snd_pcm_hardware cygnus_pcm_hw = {
>> + .info = SNDRV_PCM_INFO_MMAP |
>> + SNDRV_PCM_INFO_MMAP_VALID |
>> + SNDRV_PCM_INFO_INTERLEAVED,
>
> The DMA controller is integrated into the IP?
>
Yes, it is dedicated for the audio driver.
>> +static int enable_count;
>
> This looks bogus - why is this a global variable not part of the device
> struct and if it does need to be global why does it need no locking?
>
I will fix.
>> + 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;
>
> This looks like a switch statement, there are many places in the code
> where you're writing switch statements as chains of ifs.
>
No problem. Will use switch statements.
>> +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);
>> +}
>
> So it looks like we're getting an interrupt per period and we have to
> manually advance to the next one?
>
Yes.
>> +static irqreturn_t cygnus_dma_irq(int irq, void *data)
>> +{
>> + u32 r5_status;
>> + struct cygnus_audio *cygaud;
>> +
>> + cygaud = (struct cygnus_audio *)data;
>
> If you need to cast away from void something is very wrong.
>
Okay, will fix.
>> + /*
>> + * 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);
>
> This feels racy especially given that we seem to need every interrupt
> delivering. What if another period happens during the servicing? I
> don't understand what "overwriting esr_status" means here.
>
Let me review this handler and I will enhance as needed.
>> + return IRQ_HANDLED;
>
> If neither playback nor capture interrupts were flagged we should return
> IRQ_NONE.
>
Okay, will fix.
>> +/*
>> + * 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)
>
> This is obviously not suitable for mainline - "let's just cut'n'paste
> the shared code in case we want to add diagnostics in future" does not
> scale, you can always add local diagnostics in the core code.
>
Okay, will remove.
>> +};
>> +/*
>
> Blank line between these two please. Missing blanks between bits of
> code seem to be a common issue in this driver.
>
Okay, will fix.
>> +static int audio_ssp_in_enable(struct cygnus_aio_port *aio, bool enable)
>> +{
>> + u32 value;
>> +
>> + if (enable) {
>
>> + } else {
>
> Why not just write two functions?
>
Okay, will change.
>> +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;
>
> dev_err() (this applies throughout the driver).
>
Okay, will convert pr_err to dev_err.
> It's not clear why this is a function and not just written as a single
> statement in the one place that it's used (there are multiple calls but
> they're all together and could just be inlined).
>
I can integrate it with the calling code.
>> + if (!aio->bitrate) {
>> + pr_err("%s Use .set_clkdiv() to set bitrate\n", __func__);
>> + return -EINVAL;
>> + }
>
> This seems bogus - why are we enforcing the use of set_clkdiv()? Can't
> we figure out something esneible?
>
Yes, I believe I can set this via "set_tdm_slot".
>> + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
>> + /* Set the SSP up as slave */
>> + case SND_SOC_DAIFMT_CBM_CFM:
>
> The comments here look odd due to the indentation and really aren't
> adding much anyway.
>
Okay, will remove.
>> + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
>> + case SND_SOC_DAIFMT_RIGHT_J:
>> + case SND_SOC_DAIFMT_LEFT_J:
>> + return -EINVAL;
>
> These are just eaten by the default case.
>
Okay, will remove.
>> + 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);
>
> That if statement isn't doing what was intended...
>
Yikes. I will fix that.
>> + } 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;
>> + }
>
> We can just ignore other triggers? It's not clear why this is different
> to the playback case.
>
I will review this behaviour and fix it up.
>> +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);
>
> What is this for, it's setting off alarm bells? Note also that ASoC is
> _GPL() only.
>
Okay, will remove. It is not needed if I set the port mode from the
machine file (current done via device tree).
>> +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),
>> +};
>
> Whatever this control is doing it should be a separate patch (as I think
> we discussed in person a ELC?), it's clearly something that's unusual
> and is likely to block the rest of the code as a result. At a high
> level this needs at least some documentation.
>
Okay, will remove.
>> +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);
>
> Again, why is this being exported and why is it not _GPL? If the driver
> is adding controls I'd expect it to just add its controls itself.
>
Okay, will remove.
>> +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;
>> + }
>
> This error message is misleading - you mean to say that the driver got
> loaded for a device it doesn't understand.
>
Okay, will remove.
>> + 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);
>
> devm_ioremap_resource() will complain for you, and in general you should
> be printing error codes.
>
Okay. Will remove and rely on devm_ipremap message.
>> + node_count = 0;
>
> This doesn't seem to be needed?
>
Okay, will clean up.
>> + 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;
>
> Spelling mistake there and it would be helpful to the user to tell them
> what we parsed.
>
Okay, will fix.
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [alsa-devel] [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings
2015-03-31 7:26 ` [alsa-devel] " Lars-Peter Clausen
@ 2015-04-02 22:44 ` Lori Hikichi
0 siblings, 0 replies; 20+ messages in thread
From: Lori Hikichi @ 2015-04-02 22:44 UTC (permalink / raw)
To: Lars-Peter Clausen, Scott Branden, Rob Herring, Pawel Moll,
Mark Rutland, Ian Campbell, Kumar Gala, Liam Girdwood,
Mark Brown, Jaroslav Kysela, Takashi Iwai, alsa-devel
Cc: devicetree, linux-kernel, bryeung, abrestic,
bcm-kernel-feedback-list, linux-arm-kernel, pwestin,
Anatol Pomazao, olofj, Dmitry Torokhov
On 15-03-31 12:26 AM, Lars-Peter Clausen wrote:
> On 03/31/2015 05:16 AM, Scott Branden wrote:
> [...]
>> +- ssp-port-id: The ssp port interface to use
>> + Valid value are 0, 1, 2, or 3 (for spdif)
>
> How about using 'reg' as the property name here. It is the standard
> property name for identifying or assigning an address to a sub-node.
>
Okay. Thanks. I will change.
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 0/2] Cygnus Audio Driver
2015-03-31 6:43 ` [PATCH 0/2] Cygnus Audio Driver Mark Brown
@ 2015-04-03 19:33 ` Scott Branden
2015-04-06 9:58 ` Mark Brown
0 siblings, 1 reply; 20+ messages in thread
From: Scott Branden @ 2015-04-03 19:33 UTC (permalink / raw)
To: Mark Brown
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, Lori Hikichi, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
Hi Mark,
On 15-03-30 11:43 PM, Mark Brown wrote:
> On Mon, Mar 30, 2015 at 08:16:22PM -0700, Scott Branden wrote:
>
>> The audio PLL is embedded in the audio block and only used
>> by the audio block. The audio PLL registers are also in the middle of
>> the audio register map.
>
> When you say it's only used by the audio block do you mean to say that
> the audio block exposes no clock signals other than the bit and frame
> clocks?
>
The audio block exposes the MCLK in addition to the bit and frame clock.
Regards,
Scott
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 0/2] Cygnus Audio Driver
2015-04-03 19:33 ` Scott Branden
@ 2015-04-06 9:58 ` Mark Brown
2015-04-08 2:28 ` Lori Hikichi
0 siblings, 1 reply; 20+ messages in thread
From: Mark Brown @ 2015-04-06 9:58 UTC (permalink / raw)
To: Scott Branden
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, Lori Hikichi, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
[-- Attachment #1: Type: text/plain, Size: 780 bytes --]
On Fri, Apr 03, 2015 at 12:33:12PM -0700, Scott Branden wrote:
> On 15-03-30 11:43 PM, Mark Brown wrote:
> >On Mon, Mar 30, 2015 at 08:16:22PM -0700, Scott Branden wrote:
> >>The audio PLL is embedded in the audio block and only used
> >>by the audio block. The audio PLL registers are also in the middle of
> >>the audio register map.
> >When you say it's only used by the audio block do you mean to say that
> >the audio block exposes no clock signals other than the bit and frame
> >clocks?
> The audio block exposes the MCLK in addition to the bit and frame clock.
OK, then it's going to need to be a clock provider at some point - the
clock will be going into external devices which are going to need to be
able to interact with the clock (for example, to get the rate).
[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 473 bytes --]
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
2015-04-02 18:47 ` Lori Hikichi
@ 2015-04-06 16:19 ` Mark Brown
2015-04-08 2:30 ` Lori Hikichi
0 siblings, 1 reply; 20+ messages in thread
From: Mark Brown @ 2015-04-06 16:19 UTC (permalink / raw)
To: Lori Hikichi
Cc: Scott Branden, Rob Herring, Pawel Moll, Mark Rutland,
Ian Campbell, Kumar Gala, Liam Girdwood, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-kernel, devicetree,
linux-arm-kernel, bcm-kernel-feedback-list, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
[-- Attachment #1: Type: text/plain, Size: 1173 bytes --]
On Thu, Apr 02, 2015 at 11:47:18AM -0700, Lori Hikichi wrote:
> On 15-03-30 11:42 PM, Mark Brown wrote:
> >>+config SND_SOC_CYGNUS
> >>+ tristate "SoC platform audio for Broadcom Cygnus chips"
> >>+ depends on ARCH_BCM_CYGNUS || COMPILE_TEST
> >>+ default ARCH_BCM_CYGNUS
> Okay.
You don't need to reply to every single comment, the general assumption
will be that if there's no other followup all review comments will be
addressed. It's better to just reply to things where there's something
more detailed to say, if you explicitly reply to everything then that
makes it easier for actual replies to be missed since there's a lot of
there's a lot of the mail that's just going to be skipped through.
> >>+static void ringbuf_inc(void __iomem *audio_io, bool is_playback,
> >>+ const struct ringbuf_regs *p_rbuf)
> >So it looks like we're getting an interrupt per period and we have to
> >manually advance to the next one?
> Yes.
OK, so that seems fragile - what happens if we're slightly late
processing an interrupt or miss one entirely? Most hardware has some
way to read back the current position which tends to be more reliable,
is that not an option here?
[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 473 bytes --]
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 0/2] Cygnus Audio Driver
2015-04-06 9:58 ` Mark Brown
@ 2015-04-08 2:28 ` Lori Hikichi
2015-04-08 18:54 ` Mark Brown
0 siblings, 1 reply; 20+ messages in thread
From: Lori Hikichi @ 2015-04-08 2:28 UTC (permalink / raw)
To: Mark Brown, Scott Branden
Cc: Rob Herring, Pawel Moll, Mark Rutland, Ian Campbell, Kumar Gala,
Liam Girdwood, Jaroslav Kysela, Takashi Iwai, alsa-devel,
linux-kernel, devicetree, linux-arm-kernel,
bcm-kernel-feedback-list, Dmitry Torokhov, Anatol Pomazao,
abrestic, bryeung, olofj, pwestin
On 15-04-06 02:58 AM, Mark Brown wrote:
> On Fri, Apr 03, 2015 at 12:33:12PM -0700, Scott Branden wrote:
>> On 15-03-30 11:43 PM, Mark Brown wrote:
>>> On Mon, Mar 30, 2015 at 08:16:22PM -0700, Scott Branden wrote:
>
>>>> The audio PLL is embedded in the audio block and only used
>>>> by the audio block. The audio PLL registers are also in the middle of
>>>> the audio register map.
>
>>> When you say it's only used by the audio block do you mean to say that
>>> the audio block exposes no clock signals other than the bit and frame
>>> clocks?
>
>> The audio block exposes the MCLK in addition to the bit and frame clock.
>
> OK, then it's going to need to be a clock provider at some point - the
> clock will be going into external devices which are going to need to be
> able to interact with the clock (for example, to get the rate).
>
Currently, the ASoC machine driver is responsible for requesting a
certain frequency of MCLK be generated from our driver and then also
sending the frequency information along to the external device (codec).
This is done via the snd_soc_dai_set_sysclk. That is the only clock
interaction we have needed for the core part of the driver. For
enhanced features, we also have the need to make minor adjustments
(tweaks) to the PLL. The tweaks are used to make the PLLs output
frequency match as closely as possible to a true reference frequency.
As such, we would like to provide the finest adjustment resolution as
possible. The clocking framework only seems to allow for a 1 Hz
adjustment. This limitation and the fact that no other device seems to
need to interact directly will the PLL are why we have not put it in the
clocking framework.
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
2015-04-06 16:19 ` Mark Brown
@ 2015-04-08 2:30 ` Lori Hikichi
2015-04-08 19:23 ` Mark Brown
0 siblings, 1 reply; 20+ messages in thread
From: Lori Hikichi @ 2015-04-08 2:30 UTC (permalink / raw)
To: Mark Brown
Cc: Scott Branden, Rob Herring, Pawel Moll, Mark Rutland,
Ian Campbell, Kumar Gala, Liam Girdwood, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-kernel, devicetree,
linux-arm-kernel, bcm-kernel-feedback-list, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
On 15-04-06 09:19 AM, Mark Brown wrote:
> On Thu, Apr 02, 2015 at 11:47:18AM -0700, Lori Hikichi wrote:
>> On 15-03-30 11:42 PM, Mark Brown wrote:
>
>>>> +config SND_SOC_CYGNUS
>>>> + tristate "SoC platform audio for Broadcom Cygnus chips"
>>>> + depends on ARCH_BCM_CYGNUS || COMPILE_TEST
>>>> + default ARCH_BCM_CYGNUS
>
>> Okay.
>
> You don't need to reply to every single comment, the general assumption
> will be that if there's no other followup all review comments will be
> addressed. It's better to just reply to things where there's something
> more detailed to say, if you explicitly reply to everything then that
> makes it easier for actual replies to be missed since there's a lot of
> there's a lot of the mail that's just going to be skipped through.
>
>>>> +static void ringbuf_inc(void __iomem *audio_io, bool is_playback,
>>>> + const struct ringbuf_regs *p_rbuf)
>
>>> So it looks like we're getting an interrupt per period and we have to
>>> manually advance to the next one?
>
>> Yes.
>
> OK, so that seems fragile - what happens if we're slightly late
> processing an interrupt or miss one entirely? Most hardware has some
> way to read back the current position which tends to be more reliable,
> is that not an option here?
>
The hardware updates a read pointer (rdaddr) which we feed to ALSA via
the ".pointer" hook. So yes, the hardware does have a register that
tells us its progress. This routine (ringbuf_inc) actually updates a
write pointer (wraddr) which is a misnomer. The write pointer doesn’t
actually tell us where we are writing to – ALSA keeps track of that.
The wraddr only prevents the hardware from reading past it. We just use
it, along with a low water mark configuration register, to keep the
periodic interrupts firing. The hardware is tracking the distance
between rdaddr and wraddr and comparing that to the low water mark.
Being late processing the interrupt is okay since there are more samples
to read. That is, it was only a low water mark interrupt and we have
one period of valid data still (we configure low water to be one
period). Missing an interrupt is okay since the hardware will just stop
reading from the ring buffer when rdaddr has hit wraddr.
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 0/2] Cygnus Audio Driver
2015-04-08 2:28 ` Lori Hikichi
@ 2015-04-08 18:54 ` Mark Brown
2015-04-11 3:06 ` Lori Hikichi
0 siblings, 1 reply; 20+ messages in thread
From: Mark Brown @ 2015-04-08 18:54 UTC (permalink / raw)
To: Lori Hikichi
Cc: Scott Branden, Rob Herring, Pawel Moll, Mark Rutland,
Ian Campbell, Kumar Gala, Liam Girdwood, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-kernel, devicetree,
linux-arm-kernel, bcm-kernel-feedback-list, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
[-- Attachment #1: Type: text/plain, Size: 1973 bytes --]
On Tue, Apr 07, 2015 at 07:28:40PM -0700, Lori Hikichi wrote:
> On 15-04-06 02:58 AM, Mark Brown wrote:
> >OK, then it's going to need to be a clock provider at some point - the
> >clock will be going into external devices which are going to need to be
> >able to interact with the clock (for example, to get the rate).
> Currently, the ASoC machine driver is responsible for requesting a certain
> frequency of MCLK be generated from our driver and then also sending the
> frequency information along to the external device (codec).
> This is done via the snd_soc_dai_set_sysclk. That is the only clock
> interaction we have needed for the core part of the driver. For enhanced
I have some passing familiarity with ASoC... if you look at newer
drivers, especially those for DT systems, you'll see that we're
transitioning CODEC drivers to use the clock API for their clocks since
this makes integrating with both generic ASoC things like simple card
and non-ASoC clocks.
> features, we also have the need to make minor adjustments (tweaks) to the
> PLL. The tweaks are used to make the PLLs output frequency match as closely
> as possible to a true reference frequency. As such, we would like to provide
> the finest adjustment resolution as possible. The clocking framework only
> seems to allow for a 1 Hz adjustment. This limitation and the fact that no
> other device seems to need to interact directly will the PLL are why we have
> not put it in the clocking framework.
That's going to be an issue no matter where you put the control - the
ASoC specific clocking APIs don't have any control here either. I don't
know if we want to add the functionality for doing very fine grained
adjustments into the clock API or not (the use cases seem limited though
I'm sure they exist), though I do think we should have that discussion
if only to confirm, but that's a separate thing to how we expose any
userspace control - the clock API is a kernel internal thing.
[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 473 bytes --]
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
2015-04-08 2:30 ` Lori Hikichi
@ 2015-04-08 19:23 ` Mark Brown
2015-04-10 2:06 ` Lori Hikichi
0 siblings, 1 reply; 20+ messages in thread
From: Mark Brown @ 2015-04-08 19:23 UTC (permalink / raw)
To: Lori Hikichi
Cc: Scott Branden, Rob Herring, Pawel Moll, Mark Rutland,
Ian Campbell, Kumar Gala, Liam Girdwood, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-kernel, devicetree,
linux-arm-kernel, bcm-kernel-feedback-list, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
[-- Attachment #1: Type: text/plain, Size: 2352 bytes --]
On Tue, Apr 07, 2015 at 07:30:57PM -0700, Lori Hikichi wrote:
> On 15-04-06 09:19 AM, Mark Brown wrote:
> >OK, so that seems fragile - what happens if we're slightly late
> >processing an interrupt or miss one entirely? Most hardware has some
> >way to read back the current position which tends to be more reliable,
> >is that not an option here?
> The hardware updates a read pointer (rdaddr) which we feed to ALSA via the
> ".pointer" hook. So yes, the hardware does have a register that tells us its
> progress. This routine (ringbuf_inc) actually updates a write pointer
> (wraddr) which is a misnomer. The write pointer doesn’t actually tell us
> where we are writing to – ALSA keeps track of that. The wraddr only prevents
> the hardware from reading past it. We just use it, along with a low water
> mark configuration register, to keep the periodic interrupts firing. The
> hardware is tracking the distance between rdaddr and wraddr and comparing
> that to the low water mark.
The code has handling for both read and write so it's not just updating
a write pointer. Is there no flexibility in the hardware regarding
interrupt generation?
> Being late processing the interrupt is okay since there are more samples to
> read. That is, it was only a low water mark interrupt and we have one
> period of valid data still (we configure low water to be one period).
> Missing an interrupt is okay since the hardware will just stop reading from
> buffer when rdaddr has hit wraddr.
Stopping if we miss an interrupt is precisely the sort of situation we
want to avoid if we can - if the application is sufficiently far ahead
of the hardware everything should continue to work fine. The minimal
period size appears to be very small so this is a potential issue, if an
application tries to use many very small periods it's both more
vulnerable to some other interrupt taking longer than might be desirable
and likely that things would be fine as the application is hopefully
more than one period ahead of where the hardware is at.
If the hardware is always going to halt at the end of the period there's
not a huge amount we can do about this except possibly raise the minimum
period if systems are running into trouble but if there's a way to avoid
doing that then that would be even better.
[-- Attachment #2: Digital signature --]
[-- Type: application/pgp-signature, Size: 473 bytes --]
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC.
2015-04-08 19:23 ` Mark Brown
@ 2015-04-10 2:06 ` Lori Hikichi
0 siblings, 0 replies; 20+ messages in thread
From: Lori Hikichi @ 2015-04-10 2:06 UTC (permalink / raw)
To: Mark Brown
Cc: Scott Branden, Rob Herring, Pawel Moll, Mark Rutland,
Ian Campbell, Kumar Gala, Liam Girdwood, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-kernel, devicetree,
linux-arm-kernel, bcm-kernel-feedback-list, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
On 15-04-08 12:23 PM, Mark Brown wrote:
> On Tue, Apr 07, 2015 at 07:30:57PM -0700, Lori Hikichi wrote:
>> On 15-04-06 09:19 AM, Mark Brown wrote:
>
>>> OK, so that seems fragile - what happens if we're slightly late
>>> processing an interrupt or miss one entirely? Most hardware has some
>>> way to read back the current position which tends to be more reliable,
>>> is that not an option here?
>
>> The hardware updates a read pointer (rdaddr) which we feed to ALSA via the
>> ".pointer" hook. So yes, the hardware does have a register that tells us its
>> progress. This routine (ringbuf_inc) actually updates a write pointer
>> (wraddr) which is a misnomer. The write pointer doesn’t actually tell us
>> where we are writing to – ALSA keeps track of that. The wraddr only prevents
>> the hardware from reading past it. We just use it, along with a low water
>> mark configuration register, to keep the periodic interrupts firing. The
>> hardware is tracking the distance between rdaddr and wraddr and comparing
>> that to the low water mark.
>
> The code has handling for both read and write so it's not just updating
> a write pointer. Is there no flexibility in the hardware regarding
> interrupt generation?
>
>> Being late processing the interrupt is okay since there are more samples to
>> read. That is, it was only a low water mark interrupt and we have one
>> period of valid data still (we configure low water to be one period).
>> Missing an interrupt is okay since the hardware will just stop reading from
>> buffer when rdaddr has hit wraddr.
>
> Stopping if we miss an interrupt is precisely the sort of situation we
> want to avoid if we can - if the application is sufficiently far ahead
> of the hardware everything should continue to work fine. The minimal
> period size appears to be very small so this is a potential issue, if an
> application tries to use many very small periods it's both more
> vulnerable to some other interrupt taking longer than might be desirable
> and likely that things would be fine as the application is hopefully
> more than one period ahead of where the hardware is at.
>
> If the hardware is always going to halt at the end of the period there's
> not a huge amount we can do about this except possibly raise the minimum
> period if systems are running into trouble but if there's a way to avoid
> doing that then that would be even better.
>
Makes sense, thanks for clarifying the desired behaviour, I will make the
necessary adjustments to keep the hardware supplied with valid data even if
interrupts are held off past a whole period.
^ permalink raw reply [flat|nested] 20+ messages in thread
* Re: [PATCH 0/2] Cygnus Audio Driver
2015-04-08 18:54 ` Mark Brown
@ 2015-04-11 3:06 ` Lori Hikichi
0 siblings, 0 replies; 20+ messages in thread
From: Lori Hikichi @ 2015-04-11 3:06 UTC (permalink / raw)
To: Mark Brown
Cc: Scott Branden, Rob Herring, Pawel Moll, Mark Rutland,
Ian Campbell, Kumar Gala, Liam Girdwood, Jaroslav Kysela,
Takashi Iwai, alsa-devel, linux-kernel, devicetree,
linux-arm-kernel, bcm-kernel-feedback-list, Dmitry Torokhov,
Anatol Pomazao, abrestic, bryeung, olofj, pwestin
On 15-04-08 11:54 AM, Mark Brown wrote:
> On Tue, Apr 07, 2015 at 07:28:40PM -0700, Lori Hikichi wrote:
>> On 15-04-06 02:58 AM, Mark Brown wrote:
>
>>> OK, then it's going to need to be a clock provider at some point - the
>>> clock will be going into external devices which are going to need to be
>>> able to interact with the clock (for example, to get the rate).
>
>> Currently, the ASoC machine driver is responsible for requesting a certain
>> frequency of MCLK be generated from our driver and then also sending the
>> frequency information along to the external device (codec).
>> This is done via the snd_soc_dai_set_sysclk. That is the only clock
>> interaction we have needed for the core part of the driver. For enhanced
>
> I have some passing familiarity with ASoC... if you look at newer
> drivers, especially those for DT systems, you'll see that we're
> transitioning CODEC drivers to use the clock API for their clocks since
> this makes integrating with both generic ASoC things like simple card
> and non-ASoC clocks.
>
>> features, we also have the need to make minor adjustments (tweaks) to the
>> PLL. The tweaks are used to make the PLLs output frequency match as closely
>> as possible to a true reference frequency. As such, we would like to provide
>> the finest adjustment resolution as possible. The clocking framework only
>> seems to allow for a 1 Hz adjustment. This limitation and the fact that no
>> other device seems to need to interact directly will the PLL are why we have
>> not put it in the clocking framework.
>
> That's going to be an issue no matter where you put the control - the
> ASoC specific clocking APIs don't have any control here either. I don't
> know if we want to add the functionality for doing very fine grained
> adjustments into the clock API or not (the use cases seem limited though
> I'm sure they exist), though I do think we should have that discussion
> if only to confirm, but that's a separate thing to how we expose any
> userspace control - the clock API is a kernel internal thing.
>
Seems like there are some benefits to integrating with the clocking framework.
I will have to consider what kernel APIs we want to expose as well. I believe
we may have another kernel driver wanting to control this tweaking. Do you feel
it is ok to have to PLL code reside in this driver for now, and then we can patch
later after we get this clocking control sorted out.
^ permalink raw reply [flat|nested] 20+ messages in thread
end of thread, other threads:[~2015-04-11 3:06 UTC | newest]
Thread overview: 20+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2015-03-31 3:16 [PATCH 0/2] Cygnus Audio Driver Scott Branden
2015-03-31 3:16 ` [PATCH 1/2] ASoC: cygnus-audio: adding device tree bindings Scott Branden
2015-03-31 5:58 ` Mark Brown
2015-04-02 18:47 ` Lori Hikichi
2015-03-31 7:00 ` Mark Brown
2015-03-31 7:26 ` [alsa-devel] " Lars-Peter Clausen
2015-04-02 22:44 ` Lori Hikichi
2015-03-31 3:16 ` [PATCH 2/2] ASoC: add core audio driver for Broadcom Cygnus SOC Scott Branden
2015-03-31 6:42 ` Mark Brown
2015-04-02 18:47 ` Lori Hikichi
2015-04-06 16:19 ` Mark Brown
2015-04-08 2:30 ` Lori Hikichi
2015-04-08 19:23 ` Mark Brown
2015-04-10 2:06 ` Lori Hikichi
2015-03-31 6:43 ` [PATCH 0/2] Cygnus Audio Driver Mark Brown
2015-04-03 19:33 ` Scott Branden
2015-04-06 9:58 ` Mark Brown
2015-04-08 2:28 ` Lori Hikichi
2015-04-08 18:54 ` Mark Brown
2015-04-11 3:06 ` Lori Hikichi
This is a public inbox, see mirroring instructions
for how to clone and mirror all data and code used for this inbox;
as well as URLs for NNTP newsgroup(s).