Merge branch 'for-2.6.37' of git://git.kernel.org/pub/scm/linux/kernel/git/lrg/asoc-2.6 into for-2.6.37
diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c
new file mode 100644
index 0000000..0cd5909
--- /dev/null
+++ b/sound/soc/codecs/wl1273.c
@@ -0,0 +1,525 @@
+/*
+ * ALSA SoC WL1273 codec driver
+ *
+ * Author: Matti Aaltonen, <matti.j.aaltonen@nokia.com>
+ *
+ * Copyright: (C) 2010 Nokia Corporation
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/mfd/wl1273-core.h>
+#include <linux/slab.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc-dai.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+
+#include "wl1273.h"
+
+enum wl1273_mode { WL1273_MODE_BT, WL1273_MODE_FM_RX, WL1273_MODE_FM_TX };
+
+/* codec private data */
+struct wl1273_priv {
+ enum wl1273_mode mode;
+ struct wl1273_core *core;
+ unsigned int channels;
+};
+
+static int snd_wl1273_fm_set_i2s_mode(struct wl1273_core *core,
+ int rate, int width)
+{
+ struct device *dev = &core->i2c_dev->dev;
+ int r = 0;
+ u16 mode;
+
+ dev_dbg(dev, "rate: %d\n", rate);
+ dev_dbg(dev, "width: %d\n", width);
+
+ mutex_lock(&core->lock);
+
+ mode = core->i2s_mode & ~WL1273_IS2_WIDTH & ~WL1273_IS2_RATE;
+
+ switch (rate) {
+ case 48000:
+ mode |= WL1273_IS2_RATE_48K;
+ break;
+ case 44100:
+ mode |= WL1273_IS2_RATE_44_1K;
+ break;
+ case 32000:
+ mode |= WL1273_IS2_RATE_32K;
+ break;
+ case 22050:
+ mode |= WL1273_IS2_RATE_22_05K;
+ break;
+ case 16000:
+ mode |= WL1273_IS2_RATE_16K;
+ break;
+ case 12000:
+ mode |= WL1273_IS2_RATE_12K;
+ break;
+ case 11025:
+ mode |= WL1273_IS2_RATE_11_025;
+ break;
+ case 8000:
+ mode |= WL1273_IS2_RATE_8K;
+ break;
+ default:
+ dev_err(dev, "Sampling rate: %d not supported\n", rate);
+ r = -EINVAL;
+ goto out;
+ }
+
+ switch (width) {
+ case 16:
+ mode |= WL1273_IS2_WIDTH_32;
+ break;
+ case 20:
+ mode |= WL1273_IS2_WIDTH_40;
+ break;
+ case 24:
+ mode |= WL1273_IS2_WIDTH_48;
+ break;
+ case 25:
+ mode |= WL1273_IS2_WIDTH_50;
+ break;
+ case 30:
+ mode |= WL1273_IS2_WIDTH_60;
+ break;
+ case 32:
+ mode |= WL1273_IS2_WIDTH_64;
+ break;
+ case 40:
+ mode |= WL1273_IS2_WIDTH_80;
+ break;
+ case 48:
+ mode |= WL1273_IS2_WIDTH_96;
+ break;
+ case 64:
+ mode |= WL1273_IS2_WIDTH_128;
+ break;
+ default:
+ dev_err(dev, "Data width: %d not supported\n", width);
+ r = -EINVAL;
+ goto out;
+ }
+
+ dev_dbg(dev, "WL1273_I2S_DEF_MODE: 0x%04x\n", WL1273_I2S_DEF_MODE);
+ dev_dbg(dev, "core->i2s_mode: 0x%04x\n", core->i2s_mode);
+ dev_dbg(dev, "mode: 0x%04x\n", mode);
+
+ if (core->i2s_mode != mode) {
+ r = wl1273_fm_write_cmd(core, WL1273_I2S_MODE_CONFIG_SET, mode);
+ if (r)
+ goto out;
+
+ core->i2s_mode = mode;
+ r = wl1273_fm_write_cmd(core, WL1273_AUDIO_ENABLE,
+ WL1273_AUDIO_ENABLE_I2S);
+ if (r)
+ goto out;
+ }
+out:
+ mutex_unlock(&core->lock);
+
+ return r;
+}
+
+static int snd_wl1273_fm_set_channel_number(struct wl1273_core *core,
+ int channel_number)
+{
+ struct i2c_client *client = core->i2c_dev;
+ struct device *dev = &client->dev;
+ int r = 0;
+
+ dev_dbg(dev, "%s\n", __func__);
+
+ mutex_lock(&core->lock);
+
+ if (core->channel_number == channel_number)
+ goto out;
+
+ if (channel_number == 1 && core->mode == WL1273_MODE_RX)
+ r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
+ WL1273_RX_MONO);
+ else if (channel_number == 1 && core->mode == WL1273_MODE_TX)
+ r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
+ WL1273_TX_MONO);
+ else if (channel_number == 2 && core->mode == WL1273_MODE_RX)
+ r = wl1273_fm_write_cmd(core, WL1273_MOST_MODE_SET,
+ WL1273_RX_STEREO);
+ else if (channel_number == 2 && core->mode == WL1273_MODE_TX)
+ r = wl1273_fm_write_cmd(core, WL1273_MONO_SET,
+ WL1273_TX_STEREO);
+ else
+ r = -EINVAL;
+out:
+ mutex_unlock(&core->lock);
+
+ return r;
+}
+
+static int snd_wl1273_get_audio_route(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ ucontrol->value.integer.value[0] = wl1273->mode;
+
+ return 0;
+}
+
+static const char *wl1273_audio_route[] = { "Bt", "FmRx", "FmTx" };
+
+static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ /* Do not allow changes while stream is running */
+ if (codec->active)
+ return -EPERM;
+
+ if (ucontrol->value.integer.value[0] < 0 ||
+ ucontrol->value.integer.value[0] >= ARRAY_SIZE(wl1273_audio_route))
+ return -EINVAL;
+
+ wl1273->mode = ucontrol->value.integer.value[0];
+
+ return 1;
+}
+
+static const struct soc_enum wl1273_enum =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route);
+
+static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+ ucontrol->value.integer.value[0] = wl1273->core->audio_mode;
+
+ return 0;
+}
+
+static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+ int val, r = 0;
+
+ dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+ val = ucontrol->value.integer.value[0];
+ if (wl1273->core->audio_mode == val)
+ return 0;
+
+ r = wl1273_fm_set_audio(wl1273->core, val);
+ if (r < 0)
+ return r;
+
+ return 1;
+}
+
+static const char *wl1273_audio_strings[] = { "Digital", "Analog" };
+
+static const struct soc_enum wl1273_audio_enum =
+ SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings),
+ wl1273_audio_strings);
+
+static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+ ucontrol->value.integer.value[0] = wl1273->core->volume;
+
+ return 0;
+}
+
+static int snd_wl1273_fm_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+ int r;
+
+ dev_dbg(codec->dev, "%s: enter.\n", __func__);
+
+ r = wl1273_fm_set_volume(wl1273->core,
+ ucontrol->value.integer.value[0]);
+ if (r)
+ return r;
+
+ return 1;
+}
+
+static const struct snd_kcontrol_new wl1273_controls[] = {
+ SOC_ENUM_EXT("Codec Mode", wl1273_enum,
+ snd_wl1273_get_audio_route, snd_wl1273_set_audio_route),
+ SOC_ENUM_EXT("Audio Switch", wl1273_audio_enum,
+ snd_wl1273_fm_audio_get, snd_wl1273_fm_audio_put),
+ SOC_SINGLE_EXT("Volume", 0, 0, WL1273_MAX_VOLUME, 0,
+ snd_wl1273_fm_volume_get, snd_wl1273_fm_volume_put),
+};
+
+static int wl1273_startup(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct snd_soc_codec *codec = rtd->codec;
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ switch (wl1273->mode) {
+ case WL1273_MODE_BT:
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_RATE,
+ 8000, 8000);
+ snd_pcm_hw_constraint_minmax(substream->runtime,
+ SNDRV_PCM_HW_PARAM_CHANNELS, 1, 1);
+ break;
+ case WL1273_MODE_FM_RX:
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ pr_err("Cannot play in RX mode.\n");
+ return -EINVAL;
+ }
+ break;
+ case WL1273_MODE_FM_TX:
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ pr_err("Cannot capture in TX mode.\n");
+ return -EINVAL;
+ }
+ break;
+ default:
+ return -EINVAL;
+ break;
+ }
+
+ return 0;
+}
+
+static int wl1273_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_pcm_runtime *rtd = substream->private_data;
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(rtd->codec);
+ struct wl1273_core *core = wl1273->core;
+ unsigned int rate, width, r;
+
+ if (params_format(params) != SNDRV_PCM_FORMAT_S16_LE) {
+ pr_err("Only SNDRV_PCM_FORMAT_S16_LE supported.\n");
+ return -EINVAL;
+ }
+
+ rate = params_rate(params);
+ width = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min;
+
+ if (wl1273->mode == WL1273_MODE_BT) {
+ if (rate != 8000) {
+ pr_err("Rate %d not supported.\n", params_rate(params));
+ return -EINVAL;
+ }
+
+ if (params_channels(params) != 1) {
+ pr_err("Only mono supported.\n");
+ return -EINVAL;
+ }
+
+ return 0;
+ }
+
+ if (wl1273->mode == WL1273_MODE_FM_TX &&
+ substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ pr_err("Only playback supported with TX.\n");
+ return -EINVAL;
+ }
+
+ if (wl1273->mode == WL1273_MODE_FM_RX &&
+ substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ pr_err("Only capture supported with RX.\n");
+ return -EINVAL;
+ }
+
+ if (wl1273->mode != WL1273_MODE_FM_RX &&
+ wl1273->mode != WL1273_MODE_FM_TX) {
+ pr_err("Unexpected mode: %d.\n", wl1273->mode);
+ return -EINVAL;
+ }
+
+ r = snd_wl1273_fm_set_i2s_mode(core, rate, width);
+ if (r)
+ return r;
+
+ wl1273->channels = params_channels(params);
+ r = snd_wl1273_fm_set_channel_number(core, wl1273->channels);
+ if (r)
+ return r;
+
+ return 0;
+}
+
+static struct snd_soc_dai_ops wl1273_dai_ops = {
+ .startup = wl1273_startup,
+ .hw_params = wl1273_hw_params,
+};
+
+static struct snd_soc_dai_driver wl1273_dai = {
+ .name = "wl1273-fm",
+ .playback = {
+ .stream_name = "Playback",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE},
+ .capture = {
+ .stream_name = "Capture",
+ .channels_min = 1,
+ .channels_max = 2,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .formats = SNDRV_PCM_FMTBIT_S16_LE},
+ .ops = &wl1273_dai_ops,
+};
+
+/* Audio interface format for the soc_card driver */
+int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt)
+{
+ struct wl1273_priv *wl1273;
+
+ if (codec == NULL || fmt == NULL)
+ return -EINVAL;
+
+ wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ switch (wl1273->mode) {
+ case WL1273_MODE_FM_RX:
+ case WL1273_MODE_FM_TX:
+ *fmt = SND_SOC_DAIFMT_I2S |
+ SND_SOC_DAIFMT_NB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ break;
+ case WL1273_MODE_BT:
+ *fmt = SND_SOC_DAIFMT_DSP_A |
+ SND_SOC_DAIFMT_IB_NF |
+ SND_SOC_DAIFMT_CBM_CFM;
+
+ break;
+ default:
+ return -EINVAL;
+ }
+
+ return 0;
+}
+EXPORT_SYMBOL_GPL(wl1273_get_format);
+
+static int wl1273_probe(struct snd_soc_codec *codec)
+{
+ struct wl1273_core **core = codec->dev->platform_data;
+ struct wl1273_priv *wl1273;
+ int r;
+
+ dev_dbg(codec->dev, "%s.\n", __func__);
+
+ if (!core) {
+ dev_err(codec->dev, "Platform data is missing.\n");
+ return -EINVAL;
+ }
+
+ wl1273 = kzalloc(sizeof(struct wl1273_priv), GFP_KERNEL);
+ if (wl1273 == NULL) {
+ dev_err(codec->dev, "Cannot allocate memory.\n");
+ return -ENOMEM;
+ }
+
+ wl1273->mode = WL1273_MODE_BT;
+ wl1273->core = *core;
+
+ snd_soc_codec_set_drvdata(codec, wl1273);
+ mutex_init(&codec->mutex);
+
+ r = snd_soc_add_controls(codec, wl1273_controls,
+ ARRAY_SIZE(wl1273_controls));
+ if (r)
+ kfree(wl1273);
+
+ return r;
+}
+
+static int wl1273_remove(struct snd_soc_codec *codec)
+{
+ struct wl1273_priv *wl1273 = snd_soc_codec_get_drvdata(codec);
+
+ dev_dbg(codec->dev, "%s\n", __func__);
+ kfree(wl1273);
+
+ return 0;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_wl1273 = {
+ .probe = wl1273_probe,
+ .remove = wl1273_remove,
+};
+
+static int __devinit wl1273_platform_probe(struct platform_device *pdev)
+{
+ return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_wl1273,
+ &wl1273_dai, 1);
+}
+
+static int __devexit wl1273_platform_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_codec(&pdev->dev);
+ return 0;
+}
+
+MODULE_ALIAS("platform:wl1273-codec");
+
+static struct platform_driver wl1273_platform_driver = {
+ .driver = {
+ .name = "wl1273-codec",
+ .owner = THIS_MODULE,
+ },
+ .probe = wl1273_platform_probe,
+ .remove = __devexit_p(wl1273_platform_remove),
+};
+
+static int __init wl1273_init(void)
+{
+ return platform_driver_register(&wl1273_platform_driver);
+}
+module_init(wl1273_init);
+
+static void __exit wl1273_exit(void)
+{
+ platform_driver_unregister(&wl1273_platform_driver);
+}
+module_exit(wl1273_exit);
+
+MODULE_AUTHOR("Matti Aaltonen <matti.j.aaltonen@nokia.com>");
+MODULE_DESCRIPTION("ASoC WL1273 codec driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/codecs/wl1273.h b/sound/soc/codecs/wl1273.h
new file mode 100644
index 0000000..14ed027
--- /dev/null
+++ b/sound/soc/codecs/wl1273.h
@@ -0,0 +1,101 @@
+/*
+ * sound/soc/codec/wl1273.h
+ *
+ * ALSA SoC WL1273 codec driver
+ *
+ * Copyright (C) Nokia Corporation
+ * Author: Matti Aaltonen <matti.j.aaltonen@nokia.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __WL1273_CODEC_H__
+#define __WL1273_CODEC_H__
+
+/* I2S protocol, left channel first, data width 16 bits */
+#define WL1273_PCM_DEF_MODE 0x00
+
+/* Rx */
+#define WL1273_AUDIO_ENABLE_I2S (1 << 0)
+#define WL1273_AUDIO_ENABLE_ANALOG (1 << 1)
+
+/* Tx */
+#define WL1273_AUDIO_IO_SET_ANALOG 0
+#define WL1273_AUDIO_IO_SET_I2S 1
+
+#define WL1273_POWER_SET_OFF 0
+#define WL1273_POWER_SET_FM (1 << 0)
+#define WL1273_POWER_SET_RDS (1 << 1)
+#define WL1273_POWER_SET_RETENTION (1 << 4)
+
+#define WL1273_PUPD_SET_OFF 0x00
+#define WL1273_PUPD_SET_ON 0x01
+#define WL1273_PUPD_SET_RETENTION 0x10
+
+/* I2S mode */
+#define WL1273_IS2_WIDTH_32 0x0
+#define WL1273_IS2_WIDTH_40 0x1
+#define WL1273_IS2_WIDTH_22_23 0x2
+#define WL1273_IS2_WIDTH_23_22 0x3
+#define WL1273_IS2_WIDTH_48 0x4
+#define WL1273_IS2_WIDTH_50 0x5
+#define WL1273_IS2_WIDTH_60 0x6
+#define WL1273_IS2_WIDTH_64 0x7
+#define WL1273_IS2_WIDTH_80 0x8
+#define WL1273_IS2_WIDTH_96 0x9
+#define WL1273_IS2_WIDTH_128 0xa
+#define WL1273_IS2_WIDTH 0xf
+
+#define WL1273_IS2_FORMAT_STD (0x0 << 4)
+#define WL1273_IS2_FORMAT_LEFT (0x1 << 4)
+#define WL1273_IS2_FORMAT_RIGHT (0x2 << 4)
+#define WL1273_IS2_FORMAT_USER (0x3 << 4)
+
+#define WL1273_IS2_MASTER (0x0 << 6)
+#define WL1273_IS2_SLAVEW (0x1 << 6)
+
+#define WL1273_IS2_TRI_AFTER_SENDING (0x0 << 7)
+#define WL1273_IS2_TRI_ALWAYS_ACTIVE (0x1 << 7)
+
+#define WL1273_IS2_SDOWS_RR (0x0 << 8)
+#define WL1273_IS2_SDOWS_RF (0x1 << 8)
+#define WL1273_IS2_SDOWS_FR (0x2 << 8)
+#define WL1273_IS2_SDOWS_FF (0x3 << 8)
+
+#define WL1273_IS2_TRI_OPT (0x0 << 10)
+#define WL1273_IS2_TRI_ALWAYS (0x1 << 10)
+
+#define WL1273_IS2_RATE_48K (0x0 << 12)
+#define WL1273_IS2_RATE_44_1K (0x1 << 12)
+#define WL1273_IS2_RATE_32K (0x2 << 12)
+#define WL1273_IS2_RATE_22_05K (0x4 << 12)
+#define WL1273_IS2_RATE_16K (0x5 << 12)
+#define WL1273_IS2_RATE_12K (0x8 << 12)
+#define WL1273_IS2_RATE_11_025 (0x9 << 12)
+#define WL1273_IS2_RATE_8K (0xa << 12)
+#define WL1273_IS2_RATE (0xf << 12)
+
+#define WL1273_I2S_DEF_MODE (WL1273_IS2_WIDTH_32 | \
+ WL1273_IS2_FORMAT_STD | \
+ WL1273_IS2_MASTER | \
+ WL1273_IS2_TRI_AFTER_SENDING | \
+ WL1273_IS2_SDOWS_RR | \
+ WL1273_IS2_TRI_OPT | \
+ WL1273_IS2_RATE_48K)
+
+int wl1273_get_format(struct snd_soc_codec *codec, unsigned int *fmt);
+
+#endif /* End of __WL1273_CODEC_H__ */
diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c
index 5a6f56d..f039e8d 100644
--- a/sound/soc/fsl/fsl_dma.c
+++ b/sound/soc/fsl/fsl_dma.c
@@ -60,6 +60,7 @@
struct snd_soc_platform_driver dai;
dma_addr_t ssi_stx_phys;
dma_addr_t ssi_srx_phys;
+ unsigned int ssi_fifo_depth;
struct ccsr_dma_channel __iomem *channel;
unsigned int irq;
bool assigned;
@@ -99,6 +100,7 @@
unsigned int irq;
struct snd_pcm_substream *substream;
dma_addr_t ssi_sxx_phys;
+ unsigned int ssi_fifo_depth;
dma_addr_t ld_buf_phys;
unsigned int current_link;
dma_addr_t dma_buf_phys;
@@ -439,6 +441,7 @@
else
dma_private->ssi_sxx_phys = dma->ssi_srx_phys;
+ dma_private->ssi_fifo_depth = dma->ssi_fifo_depth;
dma_private->dma_channel = dma->channel;
dma_private->irq = dma->irq;
dma_private->substream = substream;
@@ -552,11 +555,11 @@
struct device *dev = rtd->platform->dev;
/* Number of bits per sample */
- unsigned int sample_size =
+ unsigned int sample_bits =
snd_pcm_format_physical_width(params_format(hw_params));
/* Number of bytes per frame */
- unsigned int frame_size = 2 * (sample_size / 8);
+ unsigned int sample_bytes = sample_bits / 8;
/* Bus address of SSI STX register */
dma_addr_t ssi_sxx_phys = dma_private->ssi_sxx_phys;
@@ -596,7 +599,7 @@
* that offset here. While we're at it, also tell the DMA controller
* how much data to transfer per sample.
*/
- switch (sample_size) {
+ switch (sample_bits) {
case 8:
mr |= CCSR_DMA_MR_DAHTS_1 | CCSR_DMA_MR_SAHTS_1;
ssi_sxx_phys += 3;
@@ -610,22 +613,42 @@
break;
default:
/* We should never get here */
- dev_err(dev, "unsupported sample size %u\n", sample_size);
+ dev_err(dev, "unsupported sample size %u\n", sample_bits);
return -EINVAL;
}
/*
- * BWC should always be a multiple of the frame size. BWC determines
- * how many bytes are sent/received before the DMA controller checks the
- * SSI to see if it needs to stop. For playback, the transmit FIFO can
- * hold three frames, so we want to send two frames at a time. For
- * capture, the receive FIFO is triggered when it contains one frame, so
- * we want to receive one frame at a time.
+ * BWC determines how many bytes are sent/received before the DMA
+ * controller checks the SSI to see if it needs to stop. BWC should
+ * always be a multiple of the frame size, so that we always transmit
+ * whole frames. Each frame occupies two slots in the FIFO. The
+ * parameter for CCSR_DMA_MR_BWC() is rounded down the next power of two
+ * (MR[BWC] can only represent even powers of two).
+ *
+ * To simplify the process, we set BWC to the largest value that is
+ * less than or equal to the FIFO watermark. For playback, this ensures
+ * that we transfer the maximum amount without overrunning the FIFO.
+ * For capture, this ensures that we transfer the maximum amount without
+ * underrunning the FIFO.
+ *
+ * f = SSI FIFO depth
+ * w = SSI watermark value (which equals f - 2)
+ * b = DMA bandwidth count (in bytes)
+ * s = sample size (in bytes, which equals frame_size * 2)
+ *
+ * For playback, we never transmit more than the transmit FIFO
+ * watermark, otherwise we might write more data than the FIFO can hold.
+ * The watermark is equal to the FIFO depth minus two.
+ *
+ * For capture, two equations must hold:
+ * w > f - (b / s)
+ * w >= b / s
+ *
+ * So, b > 2 * s, but b must also be <= s * w. To simplify, we set
+ * b = s * w, which is equal to
+ * (dma_private->ssi_fifo_depth - 2) * sample_bytes.
*/
- if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- mr |= CCSR_DMA_MR_BWC(2 * frame_size);
- else
- mr |= CCSR_DMA_MR_BWC(frame_size);
+ mr |= CCSR_DMA_MR_BWC((dma_private->ssi_fifo_depth - 2) * sample_bytes);
out_be32(&dma_channel->mr, mr);
@@ -879,6 +902,7 @@
struct device_node *np = of_dev->dev.of_node;
struct device_node *ssi_np;
struct resource res;
+ const uint32_t *iprop;
int ret;
/* Find the SSI node that points to us. */
@@ -889,15 +913,17 @@
}
ret = of_address_to_resource(ssi_np, 0, &res);
- of_node_put(ssi_np);
if (ret) {
- dev_err(&of_dev->dev, "could not determine device resources\n");
+ dev_err(&of_dev->dev, "could not determine resources for %s\n",
+ ssi_np->full_name);
+ of_node_put(ssi_np);
return ret;
}
dma = kzalloc(sizeof(*dma) + strlen(np->full_name), GFP_KERNEL);
if (!dma) {
dev_err(&of_dev->dev, "could not allocate dma object\n");
+ of_node_put(ssi_np);
return -ENOMEM;
}
@@ -910,6 +936,15 @@
dma->ssi_stx_phys = res.start + offsetof(struct ccsr_ssi, stx0);
dma->ssi_srx_phys = res.start + offsetof(struct ccsr_ssi, srx0);
+ iprop = of_get_property(ssi_np, "fsl,fifo-depth", NULL);
+ if (iprop)
+ dma->ssi_fifo_depth = *iprop;
+ else
+ /* Older 8610 DTs didn't have the fifo-depth property */
+ dma->ssi_fifo_depth = 8;
+
+ of_node_put(ssi_np);
+
ret = snd_soc_register_platform(&of_dev->dev, &dma->dai);
if (ret) {
dev_err(&of_dev->dev, "could not register platform\n");
diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c
index 7939c33..d1c855a 100644
--- a/sound/soc/fsl/fsl_ssi.c
+++ b/sound/soc/fsl/fsl_ssi.c
@@ -93,6 +93,7 @@
unsigned int playback;
unsigned int capture;
int asynchronous;
+ unsigned int fifo_depth;
struct snd_soc_dai_driver cpu_dai_drv;
struct device_attribute dev_attr;
struct platform_device *pdev;
@@ -337,11 +338,20 @@
/*
* Set the watermark for transmit FIFI 0 and receive FIFO 0. We
- * don't use FIFO 1. Since the SSI only supports stereo, the
- * watermark should never be an odd number.
+ * don't use FIFO 1. We program the transmit water to signal a
+ * DMA transfer if there are only two (or fewer) elements left
+ * in the FIFO. Two elements equals one frame (left channel,
+ * right channel). This value, however, depends on the depth of
+ * the transmit buffer.
+ *
+ * We program the receive FIFO to notify us if at least two
+ * elements (one frame) have been written to the FIFO. We could
+ * make this value larger (and maybe we should), but this way
+ * data will be written to memory as soon as it's available.
*/
out_be32(&ssi->sfcsr,
- CCSR_SSI_SFCSR_TFWM0(6) | CCSR_SSI_SFCSR_RFWM0(2));
+ CCSR_SSI_SFCSR_TFWM0(ssi_private->fifo_depth - 2) |
+ CCSR_SSI_SFCSR_RFWM0(ssi_private->fifo_depth - 2));
/*
* We keep the SSI disabled because if we enable it, then the
@@ -622,6 +632,7 @@
struct device_attribute *dev_attr = NULL;
struct device_node *np = of_dev->dev.of_node;
const char *p, *sprop;
+ const uint32_t *iprop;
struct resource res;
char name[64];
@@ -678,6 +689,14 @@
else
ssi_private->cpu_dai_drv.symmetric_rates = 1;
+ /* Determine the FIFO depth. */
+ iprop = of_get_property(np, "fsl,fifo-depth", NULL);
+ if (iprop)
+ ssi_private->fifo_depth = *iprop;
+ else
+ /* Older 8610 DTs didn't have the fifo-depth property */
+ ssi_private->fifo_depth = 8;
+
/* Initialize the the device_attribute structure */
dev_attr = &ssi_private->dev_attr;
dev_attr->attr.name = "statistics";