Merge tag 'staging-4.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/staging

Pull staging and IIO updates from Greg KH:
 "Here is the big staging and IIO driver pull request for 4.9-rc1.

  There are a lot of patches in here, the majority due to the
  drivers/staging/greybus/ subsystem being merged in with full
  development history that went back a few years, in order to preserve
  the work that those developers did over time.

  Lots and lots of tiny cleanups happened in the tree as well, due to
  the Outreachy application process and lots of other developers showing
  up for the first time to clean code up.  Along with those changes, we
  deleted a wireless driver, and added a raspberrypi driver (currently
  marked broken), and lots of new iio drivers.

  Overall the tree still shrunk with more lines removed than added,
  about 10 thousand lines removed in total. Full details are in the very
  long shortlog below.

  All of this has been in the linux-next tree with no issues. There will
  be some merge problems with other subsystem trees, but those are all
  minor problems and shouldn't be hard to work out when they happen
  (MAINTAINERS and some lustre build problems with the IB tree)"

And furter from me asking for clarification about greybus:
 "Right now there is a phone from Motorola shipping with this code (a
  slightly older version, but the same tree), so even though Ara is not
  alive in the same form, the functionality is happening. We are working
  with the developers of that phone to merge the newer stuff in with
  their fork so they can use the upstream version in future versions of
  their phone product line.

  Toshiba has at least one chip shipping in their catalog that
  needs/uses this protocol over a Unipro link, and rumor has it that
  there might be more in the future.

  There are also other users of the greybus protocols, there is a talk
  next week at ELC that shows how it is being used across a network
  connection to control a device, and previous ELC talks have showed the
  protocol stack being used over USB to drive embedded Linux boards.
  I've also talked to some people who are starting to work to add a host
  controller driver to control arduinos as the greybus PHY protocols are
  very useful to control a serial/i2c/spio/whatever device across a
  random physical link, as it is a way to have a self-describing device
  be attached to a host without needing manual configuration.

  So yes, people are using it, and there is still the chance that it
  will show up in a phone/laptop/tablet/whatever from Google in the
  future as well, the tech isn't dead, even if the original large phone
  project happens to be"

* tag 'staging-4.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/gregkh/staging: (3703 commits)
  Staging: fbtft: Fix bug in fbtft-core
  staging: rtl8188eu: fix double unlock error in rtw_resume_process()
  staging:r8188eu: remove GEN_MLME_EXT_HANDLER macro
  staging:r8188eu: remove GEN_DRV_CMD_HANDLER macro
  staging:r8188eu: remove GEN_EVT_CODE macro
  staging:r8188eu: remove GEN_CMD_CODE macro
  staging:r8188eu: remove pkt_newalloc member of the recv_buf structure
  staging:r8188eu: remove rtw_handle_dualmac declaration
  staging:r8188eu: remove (RGTRY|BSSID)_(OFT|SZ) macros
  staging:r8188eu: change rtl8188e_process_phy_info function argument type
  Staging: fsl-mc: Remove blank lines
  Staging: fsl-mc: Fix unaligned * in block comments
  Staging: comedi: Align the * in block comments
  Staging : ks7010 : Fix block comments warninig
  Staging: vt6655: Remove explicit NULL comparison using Coccinelle
  staging: rtl8188eu: core: rtw_xmit: Use macros instead of constants
  staging: rtl8188eu: core: rtw_xmit: Move constant of the right side
  staging: dgnc: Fix lines longer than 80 characters
  Staging: dgnc: constify attribute_group structures
  Staging: most: hdm-dim2: constify attribute_group structures
  ...

1 files changed, 1132 insertions, 0 deletions

diff --git a/drivers/staging/greybus/audio_codec.c b/drivers/staging/greybus/audio_codec.c
new file mode 100644
index 0000000..8a0744b
--- /dev/null
+++ b/drivers/staging/greybus/audio_codec.c
@@ -0,0 +1,1132 @@
+/*
+ * APBridge ALSA SoC dummy codec driver
+ * Copyright 2016 Google Inc.
+ * Copyright 2016 Linaro Ltd.
+ *
+ * Released under the GPLv2 only.
+ */
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/pm_runtime.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <uapi/linux/input.h>
+
+#include "audio_codec.h"
+#include "audio_apbridgea.h"
+#include "audio_manager.h"
+
+static struct gbaudio_codec_info *gbcodec;
+
+static struct gbaudio_data_connection *
+find_data(struct gbaudio_module_info *module, int id)
+{
+	struct gbaudio_data_connection *data;
+
+	list_for_each_entry(data, &module->data_list, list) {
+		if (id == data->id)
+			return data;
+	}
+	return NULL;
+}
+
+static struct gbaudio_stream_params *
+find_dai_stream_params(struct gbaudio_codec_info *codec, int id, int stream)
+{
+	struct gbaudio_codec_dai *dai;
+
+	list_for_each_entry(dai, &codec->dai_list, list) {
+		if (dai->id == id)
+			return &dai->params[stream];
+	}
+	return NULL;
+}
+
+static int gbaudio_module_enable_tx(struct gbaudio_codec_info *codec,
+				    struct gbaudio_module_info *module, int id)
+{
+	int module_state, ret = 0;
+	u16 data_cport, i2s_port, cportid;
+	u8 sig_bits, channels;
+	uint32_t format, rate;
+	struct gbaudio_data_connection *data;
+	struct gbaudio_stream_params *params;
+
+	/* find the dai */
+	data = find_data(module, id);
+	if (!data) {
+		dev_err(module->dev, "%d:DATA connection missing\n", id);
+		return -ENODEV;
+	}
+	module_state = data->state[SNDRV_PCM_STREAM_PLAYBACK];
+
+	params = find_dai_stream_params(codec, id, SNDRV_PCM_STREAM_PLAYBACK);
+	if (!params) {
+		dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
+		return -EINVAL;
+	}
+
+	/* register cport */
+	if (module_state < GBAUDIO_CODEC_STARTUP) {
+		i2s_port = 0;	/* fixed for now */
+		cportid = data->connection->hd_cport_id;
+		ret = gb_audio_apbridgea_register_cport(data->connection,
+						i2s_port, cportid,
+						AUDIO_APBRIDGEA_DIRECTION_TX);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "reg_cport failed:%d\n", ret);
+			return ret;
+		}
+		data->state[SNDRV_PCM_STREAM_PLAYBACK] =
+			GBAUDIO_CODEC_STARTUP;
+		dev_dbg(module->dev, "Dynamic Register %d DAI\n", cportid);
+	}
+
+	/* hw_params */
+	if (module_state < GBAUDIO_CODEC_HWPARAMS) {
+		format = params->format;
+		channels = params->channels;
+		rate = params->rate;
+		sig_bits = params->sig_bits;
+		data_cport = data->connection->intf_cport_id;
+		ret = gb_audio_gb_set_pcm(module->mgmt_connection, data_cport,
+					  format, rate, channels, sig_bits);
+		if (ret) {
+			dev_err_ratelimited(module->dev, "set_pcm failed:%d\n",
+					    ret);
+			return ret;
+		}
+		data->state[SNDRV_PCM_STREAM_PLAYBACK] =
+			GBAUDIO_CODEC_HWPARAMS;
+		dev_dbg(module->dev, "Dynamic hw_params %d DAI\n", data_cport);
+	}
+
+	/* prepare */
+	if (module_state < GBAUDIO_CODEC_PREPARE) {
+		data_cport = data->connection->intf_cport_id;
+		ret = gb_audio_gb_set_tx_data_size(module->mgmt_connection,
+						   data_cport, 192);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "set_tx_data_size failed:%d\n",
+					    ret);
+			return ret;
+		}
+		ret = gb_audio_gb_activate_tx(module->mgmt_connection,
+					      data_cport);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "activate_tx failed:%d\n", ret);
+			return ret;
+		}
+		data->state[SNDRV_PCM_STREAM_PLAYBACK] =
+			GBAUDIO_CODEC_PREPARE;
+		dev_dbg(module->dev, "Dynamic prepare %d DAI\n", data_cport);
+	}
+
+	return 0;
+}
+
+static int gbaudio_module_disable_tx(struct gbaudio_module_info *module, int id)
+{
+	int ret;
+	u16 data_cport, cportid, i2s_port;
+	int module_state;
+	struct gbaudio_data_connection *data;
+
+	/* find the dai */
+	data = find_data(module, id);
+	if (!data) {
+		dev_err(module->dev, "%d:DATA connection missing\n", id);
+		return -ENODEV;
+	}
+	module_state = data->state[SNDRV_PCM_STREAM_PLAYBACK];
+
+	if (module_state > GBAUDIO_CODEC_HWPARAMS) {
+		data_cport = data->connection->intf_cport_id;
+		ret = gb_audio_gb_deactivate_tx(module->mgmt_connection,
+						data_cport);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "deactivate_tx failed:%d\n", ret);
+			return ret;
+		}
+		dev_dbg(module->dev, "Dynamic deactivate %d DAI\n", data_cport);
+		data->state[SNDRV_PCM_STREAM_PLAYBACK] =
+			GBAUDIO_CODEC_HWPARAMS;
+	}
+
+	if (module_state > GBAUDIO_CODEC_SHUTDOWN) {
+		i2s_port = 0;	/* fixed for now */
+		cportid = data->connection->hd_cport_id;
+		ret = gb_audio_apbridgea_unregister_cport(data->connection,
+						i2s_port, cportid,
+						AUDIO_APBRIDGEA_DIRECTION_TX);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "unregister_cport failed:%d\n",
+					    ret);
+			return ret;
+		}
+		dev_dbg(module->dev, "Dynamic Unregister %d DAI\n", cportid);
+		data->state[SNDRV_PCM_STREAM_PLAYBACK] =
+			GBAUDIO_CODEC_SHUTDOWN;
+	}
+
+	return 0;
+}
+
+static int gbaudio_module_enable_rx(struct gbaudio_codec_info *codec,
+				    struct gbaudio_module_info *module, int id)
+{
+	int module_state, ret = 0;
+	u16 data_cport, i2s_port, cportid;
+	u8 sig_bits, channels;
+	uint32_t format, rate;
+	struct gbaudio_data_connection *data;
+	struct gbaudio_stream_params *params;
+
+	/* find the dai */
+	data = find_data(module, id);
+	if (!data) {
+		dev_err(module->dev, "%d:DATA connection missing\n", id);
+		return -ENODEV;
+	}
+	module_state = data->state[SNDRV_PCM_STREAM_CAPTURE];
+
+	params = find_dai_stream_params(codec, id, SNDRV_PCM_STREAM_CAPTURE);
+	if (!params) {
+		dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
+		return -EINVAL;
+	}
+
+	/* register cport */
+	if (module_state < GBAUDIO_CODEC_STARTUP) {
+		i2s_port = 0;	/* fixed for now */
+		cportid = data->connection->hd_cport_id;
+		ret = gb_audio_apbridgea_register_cport(data->connection,
+						i2s_port, cportid,
+						AUDIO_APBRIDGEA_DIRECTION_RX);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "reg_cport failed:%d\n", ret);
+			return ret;
+		}
+		data->state[SNDRV_PCM_STREAM_CAPTURE] =
+			GBAUDIO_CODEC_STARTUP;
+		dev_dbg(module->dev, "Dynamic Register %d DAI\n", cportid);
+	}
+
+	/* hw_params */
+	if (module_state < GBAUDIO_CODEC_HWPARAMS) {
+		format = params->format;
+		channels = params->channels;
+		rate = params->rate;
+		sig_bits = params->sig_bits;
+		data_cport = data->connection->intf_cport_id;
+		ret = gb_audio_gb_set_pcm(module->mgmt_connection, data_cport,
+					  format, rate, channels, sig_bits);
+		if (ret) {
+			dev_err_ratelimited(module->dev, "set_pcm failed:%d\n",
+					    ret);
+			return ret;
+		}
+		data->state[SNDRV_PCM_STREAM_CAPTURE] =
+			GBAUDIO_CODEC_HWPARAMS;
+		dev_dbg(module->dev, "Dynamic hw_params %d DAI\n", data_cport);
+	}
+
+	/* prepare */
+	if (module_state < GBAUDIO_CODEC_PREPARE) {
+		data_cport = data->connection->intf_cport_id;
+		ret = gb_audio_gb_set_rx_data_size(module->mgmt_connection,
+						   data_cport, 192);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "set_rx_data_size failed:%d\n",
+					    ret);
+			return ret;
+		}
+		ret = gb_audio_gb_activate_rx(module->mgmt_connection,
+					      data_cport);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "activate_rx failed:%d\n", ret);
+			return ret;
+		}
+		data->state[SNDRV_PCM_STREAM_CAPTURE] =
+			GBAUDIO_CODEC_PREPARE;
+		dev_dbg(module->dev, "Dynamic prepare %d DAI\n", data_cport);
+	}
+
+	return 0;
+}
+
+static int gbaudio_module_disable_rx(struct gbaudio_module_info *module, int id)
+{
+	int ret;
+	u16 data_cport, cportid, i2s_port;
+	int module_state;
+	struct gbaudio_data_connection *data;
+
+	/* find the dai */
+	data = find_data(module, id);
+	if (!data) {
+		dev_err(module->dev, "%d:DATA connection missing\n", id);
+		return -ENODEV;
+	}
+	module_state = data->state[SNDRV_PCM_STREAM_CAPTURE];
+
+	if (module_state > GBAUDIO_CODEC_HWPARAMS) {
+		data_cport = data->connection->intf_cport_id;
+		ret = gb_audio_gb_deactivate_rx(module->mgmt_connection,
+						data_cport);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "deactivate_rx failed:%d\n", ret);
+			return ret;
+		}
+		dev_dbg(module->dev, "Dynamic deactivate %d DAI\n", data_cport);
+		data->state[SNDRV_PCM_STREAM_CAPTURE] =
+			GBAUDIO_CODEC_HWPARAMS;
+	}
+
+	if (module_state > GBAUDIO_CODEC_SHUTDOWN) {
+		i2s_port = 0;	/* fixed for now */
+		cportid = data->connection->hd_cport_id;
+		ret = gb_audio_apbridgea_unregister_cport(data->connection,
+						i2s_port, cportid,
+						AUDIO_APBRIDGEA_DIRECTION_RX);
+		if (ret) {
+			dev_err_ratelimited(module->dev,
+					    "unregister_cport failed:%d\n",
+					    ret);
+			return ret;
+		}
+		dev_dbg(module->dev, "Dynamic Unregister %d DAI\n", cportid);
+		data->state[SNDRV_PCM_STREAM_CAPTURE] =
+			GBAUDIO_CODEC_SHUTDOWN;
+	}
+
+	return 0;
+}
+
+int gbaudio_module_update(struct gbaudio_codec_info *codec,
+			  struct snd_soc_dapm_widget *w,
+			  struct gbaudio_module_info *module, int enable)
+{
+	int dai_id, ret;
+	char intf_name[NAME_SIZE], dir[NAME_SIZE];
+
+	dev_dbg(module->dev, "%s:Module update %s sequence\n", w->name,
+		enable ? "Enable":"Disable");
+
+	if ((w->id != snd_soc_dapm_aif_in) && (w->id != snd_soc_dapm_aif_out)) {
+		dev_dbg(codec->dev, "No action required for %s\n", w->name);
+		return 0;
+	}
+
+	/* parse dai_id from AIF widget's stream_name */
+	ret = sscanf(w->sname, "%s %d %s", intf_name, &dai_id, dir);
+	if (ret < 3) {
+		dev_err(codec->dev, "Error while parsing dai_id for %s\n",
+			w->name);
+		return -EINVAL;
+	}
+
+	mutex_lock(&codec->lock);
+	if (w->id == snd_soc_dapm_aif_in) {
+		if (enable)
+			ret = gbaudio_module_enable_tx(codec, module, dai_id);
+		else
+			ret = gbaudio_module_disable_tx(module, dai_id);
+	} else if (w->id == snd_soc_dapm_aif_out) {
+		if (enable)
+			ret = gbaudio_module_enable_rx(codec, module, dai_id);
+		else
+			ret = gbaudio_module_disable_rx(module, dai_id);
+	}
+
+	mutex_unlock(&codec->lock);
+
+	return ret;
+}
+EXPORT_SYMBOL(gbaudio_module_update);
+
+/*
+ * codec DAI ops
+ */
+static int gbcodec_startup(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
+	struct gbaudio_stream_params *params;
+
+	mutex_lock(&codec->lock);
+
+	if (list_empty(&codec->module_list)) {
+		dev_err(codec->dev, "No codec module available\n");
+		mutex_unlock(&codec->lock);
+		return -ENODEV;
+	}
+
+	params = find_dai_stream_params(codec, dai->id, substream->stream);
+	if (!params) {
+		dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
+		mutex_unlock(&codec->lock);
+		return -EINVAL;
+	}
+	params->state = GBAUDIO_CODEC_STARTUP;
+	mutex_unlock(&codec->lock);
+	/* to prevent suspend in case of active audio */
+	pm_stay_awake(dai->dev);
+
+	return 0;
+}
+
+static void gbcodec_shutdown(struct snd_pcm_substream *substream,
+			     struct snd_soc_dai *dai)
+{
+	struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
+	struct gbaudio_stream_params *params;
+
+	mutex_lock(&codec->lock);
+
+	if (list_empty(&codec->module_list))
+		dev_info(codec->dev, "No codec module available during shutdown\n");
+
+	params = find_dai_stream_params(codec, dai->id, substream->stream);
+	if (!params) {
+		dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
+		mutex_unlock(&codec->lock);
+		return;
+	}
+	params->state = GBAUDIO_CODEC_SHUTDOWN;
+	mutex_unlock(&codec->lock);
+	pm_relax(dai->dev);
+	return;
+}
+
+static int gbcodec_hw_params(struct snd_pcm_substream *substream,
+			     struct snd_pcm_hw_params *hwparams,
+			     struct snd_soc_dai *dai)
+{
+	int ret;
+	u8 sig_bits, channels;
+	uint32_t format, rate;
+	struct gbaudio_module_info *module;
+	struct gbaudio_data_connection *data;
+	struct gb_bundle *bundle;
+	struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
+	struct gbaudio_stream_params *params;
+
+	mutex_lock(&codec->lock);
+
+	if (list_empty(&codec->module_list)) {
+		dev_err(codec->dev, "No codec module available\n");
+		mutex_unlock(&codec->lock);
+		return -ENODEV;
+	}
+
+	/*
+	 * assuming, currently only 48000 Hz, 16BIT_LE, stereo
+	 * is supported, validate params before configuring codec
+	 */
+	if (params_channels(hwparams) != 2) {
+		dev_err(dai->dev, "Invalid channel count:%d\n",
+			params_channels(hwparams));
+		mutex_unlock(&codec->lock);
+		return -EINVAL;
+	}
+	channels = params_channels(hwparams);
+
+	if (params_rate(hwparams) != 48000) {
+		dev_err(dai->dev, "Invalid sampling rate:%d\n",
+			params_rate(hwparams));
+		mutex_unlock(&codec->lock);
+		return -EINVAL;
+	}
+	rate = GB_AUDIO_PCM_RATE_48000;
+
+	if (params_format(hwparams) != SNDRV_PCM_FORMAT_S16_LE) {
+		dev_err(dai->dev, "Invalid format:%d\n",
+			params_format(hwparams));
+		mutex_unlock(&codec->lock);
+		return -EINVAL;
+	}
+	format = GB_AUDIO_PCM_FMT_S16_LE;
+
+	/* find the data connection */
+	list_for_each_entry(module, &codec->module_list, list) {
+		data = find_data(module, dai->id);
+		if (data)
+			break;
+	}
+
+	if (!data) {
+		dev_err(dai->dev, "DATA connection missing\n");
+		mutex_unlock(&codec->lock);
+		return -EINVAL;
+	}
+
+	params = find_dai_stream_params(codec, dai->id, substream->stream);
+	if (!params) {
+		dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
+		mutex_unlock(&codec->lock);
+		return -EINVAL;
+	}
+
+	bundle = to_gb_bundle(module->dev);
+	ret = gb_pm_runtime_get_sync(bundle);
+	if (ret) {
+		mutex_unlock(&codec->lock);
+		return ret;
+	}
+
+	ret = gb_audio_apbridgea_set_config(data->connection, 0,
+					    AUDIO_APBRIDGEA_PCM_FMT_16,
+					    AUDIO_APBRIDGEA_PCM_RATE_48000,
+					    6144000);
+	if (ret) {
+		dev_err_ratelimited(dai->dev, "%d: Error during set_config\n",
+				    ret);
+		mutex_unlock(&codec->lock);
+		return ret;
+	}
+
+	gb_pm_runtime_put_noidle(bundle);
+
+	params->state = GBAUDIO_CODEC_HWPARAMS;
+	params->format = format;
+	params->rate = rate;
+	params->channels = channels;
+	params->sig_bits = sig_bits;
+
+	mutex_unlock(&codec->lock);
+	return 0;
+}
+
+static int gbcodec_prepare(struct snd_pcm_substream *substream,
+			   struct snd_soc_dai *dai)
+{
+	int ret;
+	struct gbaudio_module_info *module;
+	struct gbaudio_data_connection *data;
+	struct gb_bundle *bundle;
+	struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
+	struct gbaudio_stream_params *params;
+
+	mutex_lock(&codec->lock);
+
+	if (list_empty(&codec->module_list)) {
+		dev_err(codec->dev, "No codec module available\n");
+		mutex_unlock(&codec->lock);
+		return -ENODEV;
+	}
+
+	list_for_each_entry(module, &codec->module_list, list) {
+		/* find the dai */
+		data = find_data(module, dai->id);
+		if (data)
+			break;
+	}
+	if (!data) {
+		dev_err(dai->dev, "DATA connection missing\n");
+		mutex_unlock(&codec->lock);
+		return -ENODEV;
+	}
+
+	params = find_dai_stream_params(codec, dai->id, substream->stream);
+	if (!params) {
+		dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
+		mutex_unlock(&codec->lock);
+		return -EINVAL;
+	}
+
+	bundle = to_gb_bundle(module->dev);
+	ret = gb_pm_runtime_get_sync(bundle);
+	if (ret) {
+		mutex_unlock(&codec->lock);
+		return ret;
+	}
+
+	switch (substream->stream) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		ret = gb_audio_apbridgea_set_tx_data_size(data->connection, 0,
+							  192);
+		break;
+	case SNDRV_PCM_STREAM_CAPTURE:
+		ret = gb_audio_apbridgea_set_rx_data_size(data->connection, 0,
+							  192);
+		break;
+	}
+	if (ret) {
+		mutex_unlock(&codec->lock);
+		dev_err_ratelimited(dai->dev, "set_data_size failed:%d\n",
+				     ret);
+		return ret;
+	}
+
+	gb_pm_runtime_put_noidle(bundle);
+
+	params->state = GBAUDIO_CODEC_PREPARE;
+	mutex_unlock(&codec->lock);
+	return 0;
+}
+
+static int gbcodec_mute_stream(struct snd_soc_dai *dai, int mute, int stream)
+{
+	int ret;
+	struct gbaudio_data_connection *data;
+	struct gbaudio_module_info *module;
+	struct gb_bundle *bundle;
+	struct gbaudio_codec_info *codec = dev_get_drvdata(dai->dev);
+	struct gbaudio_stream_params *params;
+
+
+	dev_dbg(dai->dev, "Mute:%d, Direction:%s\n", mute,
+		stream ? "CAPTURE":"PLAYBACK");
+
+	mutex_lock(&codec->lock);
+
+	params = find_dai_stream_params(codec, dai->id, stream);
+	if (!params) {
+		dev_err(codec->dev, "Failed to fetch dai_stream pointer\n");
+		mutex_unlock(&codec->lock);
+		return -EINVAL;
+	}
+
+	if (list_empty(&codec->module_list)) {
+		dev_err(codec->dev, "No codec module available\n");
+		if (mute) {
+			params->state = GBAUDIO_CODEC_STOP;
+			ret = 0;
+		} else {
+			ret = -ENODEV;
+		}
+		mutex_unlock(&codec->lock);
+		return ret;
+	}
+
+	list_for_each_entry(module, &codec->module_list, list) {
+		/* find the dai */
+		data = find_data(module, dai->id);
+		if (data)
+			break;
+	}
+	if (!data) {
+		dev_err(dai->dev, "%s:%s DATA connection missing\n",
+			dai->name, module->name);
+		mutex_unlock(&codec->lock);
+		return -ENODEV;
+	}
+
+	bundle = to_gb_bundle(module->dev);
+	ret = gb_pm_runtime_get_sync(bundle);
+	if (ret) {
+		mutex_unlock(&codec->lock);
+		return ret;
+	}
+
+	if (!mute && !stream) {/* start playback */
+		ret = gb_audio_apbridgea_prepare_tx(data->connection,
+						    0);
+		if (!ret)
+			ret = gb_audio_apbridgea_start_tx(data->connection,
+							  0, 0);
+		params->state = GBAUDIO_CODEC_START;
+	} else if (!mute && stream) {/* start capture */
+		ret = gb_audio_apbridgea_prepare_rx(data->connection,
+						    0);
+		if (!ret)
+			ret = gb_audio_apbridgea_start_rx(data->connection,
+							  0);
+		params->state = GBAUDIO_CODEC_START;
+	} else if (mute && !stream) {/* stop playback */
+		ret = gb_audio_apbridgea_stop_tx(data->connection, 0);
+		if (!ret)
+			ret = gb_audio_apbridgea_shutdown_tx(data->connection,
+							     0);
+		params->state = GBAUDIO_CODEC_STOP;
+	} else if (mute && stream) {/* stop capture */
+		ret = gb_audio_apbridgea_stop_rx(data->connection, 0);
+		if (!ret)
+			ret = gb_audio_apbridgea_shutdown_rx(data->connection,
+							     0);
+		params->state = GBAUDIO_CODEC_STOP;
+	} else
+		ret = -EINVAL;
+	if (ret)
+		dev_err_ratelimited(dai->dev,
+				    "%s:Error during %s %s stream:%d\n",
+				    module->name, mute ? "Mute" : "Unmute",
+				    stream ? "Capture" : "Playback", ret);
+
+	gb_pm_runtime_put_noidle(bundle);
+	mutex_unlock(&codec->lock);
+	return ret;
+}
+
+static struct snd_soc_dai_ops gbcodec_dai_ops = {
+	.startup = gbcodec_startup,
+	.shutdown = gbcodec_shutdown,
+	.hw_params = gbcodec_hw_params,
+	.prepare = gbcodec_prepare,
+	.mute_stream = gbcodec_mute_stream,
+};
+
+static struct snd_soc_dai_driver gbaudio_dai[] = {
+	{
+		.name = "apb-i2s0",
+		.id = 0,
+		.playback = {
+			.stream_name = "I2S 0 Playback",
+			.rates = SNDRV_PCM_RATE_48000,
+			.formats = SNDRV_PCM_FORMAT_S16_LE,
+			.rate_max = 48000,
+			.rate_min = 48000,
+			.channels_min = 1,
+			.channels_max = 2,
+		},
+		.capture = {
+			.stream_name = "I2S 0 Capture",
+			.rates = SNDRV_PCM_RATE_48000,
+			.formats = SNDRV_PCM_FORMAT_S16_LE,
+			.rate_max = 48000,
+			.rate_min = 48000,
+			.channels_min = 1,
+			.channels_max = 2,
+		},
+		.ops = &gbcodec_dai_ops,
+	},
+};
+
+static int gbaudio_init_jack(struct gbaudio_module_info *module,
+			     struct snd_soc_codec *codec)
+{
+	int ret;
+
+	if (!module->jack_mask)
+		return 0;
+
+	snprintf(module->jack_name, NAME_SIZE, "GB %d Headset Jack",
+		 module->dev_id);
+	ret = snd_soc_jack_new(codec, module->jack_name, module->jack_mask,
+			       &module->headset_jack);
+	if (ret) {
+		dev_err(module->dev, "Failed to create new jack\n");
+		return ret;
+	}
+
+	if (!module->button_mask)
+		return 0;
+
+	snprintf(module->button_name, NAME_SIZE, "GB %d Button Jack",
+		 module->dev_id);
+	ret = snd_soc_jack_new(codec, module->button_name, module->button_mask,
+			       &module->button_jack);
+	if (ret) {
+		dev_err(module->dev, "Failed to create button jack\n");
+		return ret;
+	}
+
+	/*
+	 * Currently, max 4 buttons are supported with following key mapping
+	 * BTN_0 = KEY_MEDIA
+	 * BTN_1 = KEY_VOICECOMMAND
+	 * BTN_2 = KEY_VOLUMEUP
+	 * BTN_3 = KEY_VOLUMEDOWN
+	 */
+
+	if (module->button_mask & SND_JACK_BTN_0) {
+		ret = snd_jack_set_key(module->button_jack.jack, SND_JACK_BTN_0,
+				       KEY_MEDIA);
+		if (ret) {
+			dev_err(module->dev, "Failed to set BTN_0\n");
+			return ret;
+		}
+	}
+
+	if (module->button_mask & SND_JACK_BTN_1) {
+		ret = snd_jack_set_key(module->button_jack.jack, SND_JACK_BTN_1,
+				       KEY_VOICECOMMAND);
+		if (ret) {
+			dev_err(module->dev, "Failed to set BTN_1\n");
+			return ret;
+		}
+	}
+
+	if (module->button_mask & SND_JACK_BTN_2) {
+		ret = snd_jack_set_key(module->button_jack.jack, SND_JACK_BTN_2,
+				       KEY_VOLUMEUP);
+		if (ret) {
+			dev_err(module->dev, "Failed to set BTN_2\n");
+			return ret;
+		}
+	}
+
+	if (module->button_mask & SND_JACK_BTN_3) {
+		ret = snd_jack_set_key(module->button_jack.jack, SND_JACK_BTN_3,
+				       KEY_VOLUMEDOWN);
+		if (ret) {
+			dev_err(module->dev, "Failed to set BTN_0\n");
+			return ret;
+		}
+	}
+
+	/* FIXME
+	 * verify if this is really required
+	set_bit(INPUT_PROP_NO_DUMMY_RELEASE,
+		module->button_jack.jack->input_dev->propbit);
+	*/
+
+	return 0;
+}
+
+int gbaudio_register_module(struct gbaudio_module_info *module)
+{
+	int ret;
+	struct snd_soc_codec *codec;
+	struct snd_card *card;
+	struct snd_soc_jack *jack = NULL;
+
+	if (!gbcodec) {
+		dev_err(module->dev, "GB Codec not yet probed\n");
+		return -EAGAIN;
+	}
+
+	codec = gbcodec->codec;
+	card = codec->card->snd_card;
+
+	down_write(&card->controls_rwsem);
+
+	if (module->num_dais) {
+		dev_err(gbcodec->dev,
+			"%d:DAIs not supported via gbcodec driver\n",
+			module->num_dais);
+		up_write(&card->controls_rwsem);
+		return -EINVAL;
+	}
+
+	ret = gbaudio_init_jack(module, codec);
+	if (ret) {
+		up_write(&card->controls_rwsem);
+		return ret;
+	}
+
+	if (module->dapm_widgets)
+		snd_soc_dapm_new_controls(&codec->dapm, module->dapm_widgets,
+					  module->num_dapm_widgets);
+	if (module->controls)
+		snd_soc_add_codec_controls(codec, module->controls,
+				     module->num_controls);
+	if (module->dapm_routes)
+		snd_soc_dapm_add_routes(&codec->dapm, module->dapm_routes,
+					module->num_dapm_routes);
+
+	/* card already instantiated, create widgets here only */
+	if (codec->card->instantiated) {
+		snd_soc_dapm_link_component_dai_widgets(codec->card,
+							&codec->dapm);
+#ifdef CONFIG_SND_JACK
+		/* register jack devices for this module from codec->jack_list */
+		list_for_each_entry(jack, &codec->jack_list, list) {
+			if ((jack == &module->headset_jack)
+			    || (jack == &module->button_jack))
+				snd_device_register(codec->card->snd_card,
+						    jack->jack);
+		}
+#endif
+	}
+
+	mutex_lock(&gbcodec->lock);
+	list_add(&module->list, &gbcodec->module_list);
+	mutex_unlock(&gbcodec->lock);
+
+	if (codec->card->instantiated)
+		ret = snd_soc_dapm_new_widgets(&codec->dapm);
+	dev_dbg(codec->dev, "Registered %s module\n", module->name);
+
+	up_write(&card->controls_rwsem);
+	return ret;
+}
+EXPORT_SYMBOL(gbaudio_register_module);
+
+static void gbaudio_codec_clean_data_tx(struct gbaudio_data_connection *data)
+{
+	u16 i2s_port, cportid;
+	int ret;
+
+	if (list_is_singular(&gbcodec->module_list)) {
+		ret = gb_audio_apbridgea_stop_tx(data->connection, 0);
+		if (ret)
+			return;
+		ret = gb_audio_apbridgea_shutdown_tx(data->connection,
+						     0);
+		if (ret)
+			return;
+	}
+	i2s_port = 0;	/* fixed for now */
+	cportid = data->connection->hd_cport_id;
+	ret = gb_audio_apbridgea_unregister_cport(data->connection,
+						  i2s_port, cportid,
+						  AUDIO_APBRIDGEA_DIRECTION_TX);
+	data->state[0] = GBAUDIO_CODEC_SHUTDOWN;
+}
+
+static void gbaudio_codec_clean_data_rx(struct gbaudio_data_connection *data)
+{
+	u16 i2s_port, cportid;
+	int ret;
+
+	if (list_is_singular(&gbcodec->module_list)) {
+		ret = gb_audio_apbridgea_stop_rx(data->connection, 0);
+		if (ret)
+			return;
+		ret = gb_audio_apbridgea_shutdown_rx(data->connection,
+						     0);
+		if (ret)
+			return;
+	}
+	i2s_port = 0;	/* fixed for now */
+	cportid = data->connection->hd_cport_id;
+	ret = gb_audio_apbridgea_unregister_cport(data->connection,
+						  i2s_port, cportid,
+						  AUDIO_APBRIDGEA_DIRECTION_RX);
+	data->state[1] = GBAUDIO_CODEC_SHUTDOWN;
+}
+
+
+static void gbaudio_codec_cleanup(struct gbaudio_module_info *module)
+{
+	struct gbaudio_data_connection *data;
+	int pb_state, cap_state;
+
+	dev_dbg(gbcodec->dev, "%s: removed, cleanup APBridge\n", module->name);
+	list_for_each_entry(data, &module->data_list, list) {
+		pb_state = data->state[0];
+		cap_state = data->state[1];
+
+		if (pb_state > GBAUDIO_CODEC_SHUTDOWN)
+			gbaudio_codec_clean_data_tx(data);
+
+		if (cap_state > GBAUDIO_CODEC_SHUTDOWN)
+			gbaudio_codec_clean_data_rx(data);
+
+	}
+}
+
+void gbaudio_unregister_module(struct gbaudio_module_info *module)
+{
+	struct snd_soc_codec *codec = gbcodec->codec;
+	struct snd_card *card = codec->card->snd_card;
+	struct snd_soc_jack *jack, *next_j;
+	int mask;
+
+	dev_dbg(codec->dev, "Unregister %s module\n", module->name);
+
+	down_write(&card->controls_rwsem);
+	mutex_lock(&gbcodec->lock);
+	gbaudio_codec_cleanup(module);
+	list_del(&module->list);
+	dev_dbg(codec->dev, "Process Unregister %s module\n", module->name);
+	mutex_unlock(&gbcodec->lock);
+
+#ifdef CONFIG_SND_JACK
+	/* free jack devices for this module from codec->jack_list */
+	list_for_each_entry_safe(jack, next_j, &codec->jack_list, list) {
+		if (jack == &module->headset_jack)
+			mask = GBCODEC_JACK_MASK;
+		else if (jack == &module->button_jack)
+			mask = GBCODEC_JACK_BUTTON_MASK;
+		else
+			mask = 0;
+		if (mask) {
+			dev_dbg(module->dev, "Report %s removal\n",
+				jack->jack->id);
+			snd_soc_jack_report(jack, 0, mask);
+			snd_device_free(codec->card->snd_card, jack->jack);
+			list_del(&jack->list);
+		}
+	}
+#endif
+
+	if (module->dapm_routes) {
+		dev_dbg(codec->dev, "Removing %d routes\n",
+			module->num_dapm_routes);
+		snd_soc_dapm_del_routes(&codec->dapm, module->dapm_routes,
+					module->num_dapm_routes);
+	}
+	if (module->controls) {
+		dev_dbg(codec->dev, "Removing %d controls\n",
+			module->num_controls);
+		snd_soc_remove_codec_controls(codec, module->controls,
+					  module->num_controls);
+	}
+	if (module->dapm_widgets) {
+		dev_dbg(codec->dev, "Removing %d widgets\n",
+			module->num_dapm_widgets);
+		snd_soc_dapm_free_controls(&codec->dapm, module->dapm_widgets,
+					   module->num_dapm_widgets);
+	}
+
+	dev_dbg(codec->dev, "Unregistered %s module\n", module->name);
+
+	up_write(&card->controls_rwsem);
+}
+EXPORT_SYMBOL(gbaudio_unregister_module);
+
+/*
+ * codec driver ops
+ */
+static int gbcodec_probe(struct snd_soc_codec *codec)
+{
+	int i;
+	struct gbaudio_codec_info *info;
+	struct gbaudio_codec_dai *dai;
+
+	info = devm_kzalloc(codec->dev, sizeof(*info), GFP_KERNEL);
+	if (!info)
+		return -ENOMEM;
+
+	info->dev = codec->dev;
+	INIT_LIST_HEAD(&info->module_list);
+	mutex_init(&info->lock);
+	INIT_LIST_HEAD(&info->dai_list);
+
+	/* init dai_list used to maintain runtime stream info */
+	for (i = 0; i < ARRAY_SIZE(gbaudio_dai); i++) {
+		dai = devm_kzalloc(codec->dev, sizeof(*dai), GFP_KERNEL);
+		if (!dai)
+			return -ENOMEM;
+		dai->id = gbaudio_dai[i].id;
+		list_add(&dai->list, &info->dai_list);
+	}
+
+	info->codec = codec;
+	snd_soc_codec_set_drvdata(codec, info);
+	gbcodec = info;
+
+	device_init_wakeup(codec->dev, 1);
+	return 0;
+}
+
+static int gbcodec_remove(struct snd_soc_codec *codec)
+{
+	/* Empty function for now */
+	return 0;
+}
+
+static u8 gbcodec_reg[GBCODEC_REG_COUNT] = {
+	[GBCODEC_CTL_REG] = GBCODEC_CTL_REG_DEFAULT,
+	[GBCODEC_MUTE_REG] = GBCODEC_MUTE_REG_DEFAULT,
+	[GBCODEC_PB_LVOL_REG] = GBCODEC_PB_VOL_REG_DEFAULT,
+	[GBCODEC_PB_RVOL_REG] = GBCODEC_PB_VOL_REG_DEFAULT,
+	[GBCODEC_CAP_LVOL_REG] = GBCODEC_CAP_VOL_REG_DEFAULT,
+	[GBCODEC_CAP_RVOL_REG] = GBCODEC_CAP_VOL_REG_DEFAULT,
+	[GBCODEC_APB1_MUX_REG] = GBCODEC_APB1_MUX_REG_DEFAULT,
+	[GBCODEC_APB2_MUX_REG] = GBCODEC_APB2_MUX_REG_DEFAULT,
+};
+
+static int gbcodec_write(struct snd_soc_codec *codec, unsigned int reg,
+			 unsigned int value)
+{
+	int ret = 0;
+
+	if (reg == SND_SOC_NOPM)
+		return 0;
+
+	BUG_ON(reg >= GBCODEC_REG_COUNT);
+
+	gbcodec_reg[reg] = value;
+	dev_dbg(codec->dev, "reg[%d] = 0x%x\n", reg, value);
+
+	return ret;
+}
+
+static unsigned int gbcodec_read(struct snd_soc_codec *codec,
+				 unsigned int reg)
+{
+	unsigned int val = 0;
+
+	if (reg == SND_SOC_NOPM)
+		return 0;
+
+	BUG_ON(reg >= GBCODEC_REG_COUNT);
+
+	val = gbcodec_reg[reg];
+	dev_dbg(codec->dev, "reg[%d] = 0x%x\n", reg, val);
+
+	return val;
+}
+
+static struct snd_soc_codec_driver soc_codec_dev_gbaudio = {
+	.probe	= gbcodec_probe,
+	.remove	= gbcodec_remove,
+
+	.read = gbcodec_read,
+	.write = gbcodec_write,
+
+	.reg_cache_size = GBCODEC_REG_COUNT,
+	.reg_cache_default = gbcodec_reg_defaults,
+	.reg_word_size = 1,
+
+	.idle_bias_off = true,
+	.ignore_pmdown_time = 1,
+};
+
+#ifdef CONFIG_PM
+static int gbaudio_codec_suspend(struct device *dev)
+{
+	dev_dbg(dev, "%s: suspend\n", __func__);
+	return 0;
+}
+
+static int gbaudio_codec_resume(struct device *dev)
+{
+	dev_dbg(dev, "%s: resume\n", __func__);
+	return 0;
+}
+
+static const struct dev_pm_ops gbaudio_codec_pm_ops = {
+	.suspend	= gbaudio_codec_suspend,
+	.resume		= gbaudio_codec_resume,
+};
+#endif
+
+static int gbaudio_codec_probe(struct platform_device *pdev)
+{
+	return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_gbaudio,
+			gbaudio_dai, ARRAY_SIZE(gbaudio_dai));
+}
+
+static int gbaudio_codec_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_codec(&pdev->dev);
+	return 0;
+}
+
+static const struct of_device_id greybus_asoc_machine_of_match[]  = {
+	{ .compatible = "toshiba,apb-dummy-codec", },
+	{},
+};
+
+static struct platform_driver gbaudio_codec_driver = {
+	.driver = {
+		.name = "apb-dummy-codec",
+		.owner = THIS_MODULE,
+#ifdef CONFIG_PM
+		.pm = &gbaudio_codec_pm_ops,
+#endif
+		.of_match_table = greybus_asoc_machine_of_match,
+	},
+	.probe = gbaudio_codec_probe,
+	.remove = gbaudio_codec_remove,
+};
+module_platform_driver(gbaudio_codec_driver);
+
+MODULE_DESCRIPTION("APBridge ALSA SoC dummy codec driver");
+MODULE_AUTHOR("Vaibhav Agarwal <vaibhav.agarwal@linaro.org>");
+MODULE_LICENSE("GPL v2");
+MODULE_ALIAS("platform:apb-dummy-codec");