Remove unnecessary parentheses around simple boolean expressions in the if condition. The kernel coding style prefers not to use parentheses around simple conditions.
Signed-off-by: YiCheng Lu 01luyicheng@gmail.com --- drivers/staging/greybus/audio_codec.c | 1100 +++++++++++++++++++++++++ 1 file changed, 1100 insertions(+) create mode 100644 drivers/staging/greybus/audio_codec.c
diff --git a/drivers/staging/greybus/audio_codec.c b/drivers/staging/greybus/audio_codec.c new file mode 100644 index 0000000..44aee82 --- /dev/null +++ b/drivers/staging/greybus/audio_codec.c @@ -0,0 +1,1100 @@ +// SPDX-License-Identifier: GPL-2.0 +/* + * APBridge ALSA SoC dummy codec driver + * Copyright 2016 Google Inc. + * Copyright 2016 Linaro Ltd. + */ +#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" +#include "audio_helper.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; + u32 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; + u32 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); +} + +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; + u32 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); + gb_pm_runtime_put_noidle(bundle); + mutex_unlock(&codec->lock); + return ret; + } + + gb_pm_runtime_put_noidle(bundle); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + sig_bits = dai->driver->playback.sig_bits; + else + sig_bits = dai->driver->capture.sig_bits; + + 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 = NULL, *iter; + 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(iter, &codec->module_list, list) { + /* find the dai */ + data = find_data(iter, dai->id); + if (data) { + module = iter; + 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) { + gb_pm_runtime_put_noidle(bundle); + 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 = NULL, *iter; + 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(iter, &codec->module_list, list) { + /* find the dai */ + data = find_data(iter, dai->id); + if (data) { + module = iter; + break; + } + } + if (!data) { + dev_err(dai->dev, "%s DATA connection missing\n", + dai->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 const 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_FMTBIT_S16_LE, + .rate_max = 48000, + .rate_min = 48000, + .channels_min = 1, + .channels_max = 2, + .sig_bits = 16, + }, + .capture = { + .stream_name = "I2S 0 Capture", + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_max = 48000, + .rate_min = 48000, + .channels_min = 1, + .channels_max = 2, + .sig_bits = 16, + }, + .ops = &gbcodec_dai_ops, + }, +}; + +static int gbaudio_init_jack(struct gbaudio_module_info *module, + struct snd_soc_card *card) +{ + int ret; + struct gbaudio_jack *jack, *n; + struct snd_soc_jack_pin *headset, *button; + + if (!module->jack_mask) + return 0; + + snprintf(module->jack_name, NAME_SIZE, "GB %d Headset Jack", + module->dev_id); + + headset = devm_kzalloc(module->dev, sizeof(*headset), GFP_KERNEL); + if (!headset) + return -ENOMEM; + + headset->pin = module->jack_name; + headset->mask = module->jack_mask; + ret = snd_soc_card_jack_new_pins(card, module->jack_name, + module->jack_mask, + &module->headset.jack, headset, 1); + if (ret) { + dev_err(module->dev, "Failed to create new jack\n"); + return ret; + } + + /* Add to module's jack list */ + list_add(&module->headset.list, &module->jack_list); + + if (!module->button_mask) + return 0; + + snprintf(module->button_name, NAME_SIZE, "GB %d Button Jack", + module->dev_id); + button = devm_kzalloc(module->dev, sizeof(*button), GFP_KERNEL); + if (!button) { + ret = -ENOMEM; + goto free_jacks; + } + + button->pin = module->button_name; + button->mask = module->button_mask; + ret = snd_soc_card_jack_new_pins(card, module->button_name, + module->button_mask, + &module->button.jack, + button, 1); + if (ret) { + dev_err(module->dev, "Failed to create button jack\n"); + goto free_jacks; + } + + /* Add to module's jack list */ + list_add(&module->button.list, &module->jack_list); + + /* + * 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"); + goto free_jacks; + } + } + + 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"); + goto free_jacks; + } + } + + 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"); + goto free_jacks; + } + } + + 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"); + goto free_jacks; + } + } + + /* FIXME + * verify if this is really required + set_bit(INPUT_PROP_NO_DUMMY_RELEASE, + module->button.jack.jack->input_dev->propbit); + */ + + return 0; + +free_jacks: + list_for_each_entry_safe(jack, n, &module->jack_list, list) { + snd_device_free(card->snd_card, jack->jack.jack); + list_del(&jack->list); + } + + return ret; +} + +int gbaudio_register_module(struct gbaudio_module_info *module) +{ + int ret; + struct snd_soc_component *comp; + struct snd_soc_dapm_context *dapm; + struct gbaudio_jack *jack = NULL; + + if (!gbcodec) { + dev_err(module->dev, "GB Codec not yet probed\n"); + return -EAGAIN; + } + + comp = gbcodec->component; + dapm = snd_soc_component_to_dapm(comp); + + mutex_lock(&gbcodec->register_mutex); + + if (module->num_dais) { + dev_err(gbcodec->dev, + "%d:DAIs not supported via gbcodec driver\n", + module->num_dais); + mutex_unlock(&gbcodec->register_mutex); + return -EINVAL; + } + + ret = gbaudio_init_jack(module, comp->card); + if (ret) { + mutex_unlock(&gbcodec->register_mutex); + return ret; + } + + if (module->dapm_widgets) + snd_soc_dapm_new_controls(dapm, module->dapm_widgets, + module->num_dapm_widgets); + if (module->controls) + snd_soc_add_component_controls(comp, module->controls, + module->num_controls); + if (module->dapm_routes) + snd_soc_dapm_add_routes(dapm, module->dapm_routes, + module->num_dapm_routes); + + /* card already instantiated, create widgets here only */ + if (comp->card->instantiated) { + gbaudio_dapm_link_component_dai_widgets(comp->card, dapm); +#ifdef CONFIG_SND_JACK + /* + * register jack devices for this module + * from codec->jack_list + */ + list_for_each_entry(jack, &module->jack_list, list) { + snd_device_register(comp->card->snd_card, + jack->jack.jack); + } +#endif + } + + mutex_lock(&gbcodec->lock); + list_add(&module->list, &gbcodec->module_list); + mutex_unlock(&gbcodec->lock); + + if (comp->card->instantiated) + ret = snd_soc_dapm_new_widgets(comp->card); + dev_dbg(comp->dev, "Registered %s module\n", module->name); + + mutex_unlock(&gbcodec->register_mutex); + 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_component *comp = gbcodec->component; + struct gbaudio_jack *jack, *n; + int mask; + + dev_dbg(comp->dev, "Unregister %s module\n", module->name); + + mutex_lock(&gbcodec->register_mutex); + mutex_lock(&gbcodec->lock); + gbaudio_codec_cleanup(module); + list_del(&module->list); + dev_dbg(comp->dev, "Process Unregister %s module\n", module->name); + mutex_unlock(&gbcodec->lock); + +#ifdef CONFIG_SND_JACK + /* free jack devices for this module jack_list */ + list_for_each_entry_safe(jack, n, &module->jack_list, list) { + if (jack == &module->headset) + mask = GBCODEC_JACK_MASK; + else if (jack == &module->button) + mask = GBCODEC_JACK_BUTTON_MASK; + else + mask = 0; + if (mask) { + dev_dbg(module->dev, "Report %s removal\n", + jack->jack.jack->id); + snd_soc_jack_report(&jack->jack, 0, mask); + snd_device_free(comp->card->snd_card, + jack->jack.jack); + list_del(&jack->list); + } + } +#endif + + if (module->dapm_routes) { + struct snd_soc_dapm_context *dapm = snd_soc_component_to_dapm(comp); + + dev_dbg(comp->dev, "Removing %d routes\n", + module->num_dapm_routes); + snd_soc_dapm_del_routes(dapm, module->dapm_routes, + module->num_dapm_routes); + } + if (module->controls) { + dev_dbg(comp->dev, "Removing %d controls\n", + module->num_controls); + /* release control semaphore */ + gbaudio_remove_component_controls(comp, module->controls, + module->num_controls); + } + if (module->dapm_widgets) { + struct snd_soc_dapm_context *dapm = snd_soc_component_to_dapm(comp); + + dev_dbg(comp->dev, "Removing %d widgets\n", + module->num_dapm_widgets); + gbaudio_dapm_free_controls(dapm, module->dapm_widgets, + module->num_dapm_widgets); + } + + dev_dbg(comp->dev, "Unregistered %s module\n", module->name); + + mutex_unlock(&gbcodec->register_mutex); +} +EXPORT_SYMBOL(gbaudio_unregister_module); + +/* + * component driver ops + */ +static int gbcodec_probe(struct snd_soc_component *comp) +{ + int i; + struct gbaudio_codec_info *info; + struct gbaudio_codec_dai *dai; + + info = devm_kzalloc(comp->dev, sizeof(*info), GFP_KERNEL); + if (!info) + return -ENOMEM; + + info->dev = comp->dev; + INIT_LIST_HEAD(&info->module_list); + mutex_init(&info->lock); + mutex_init(&info->register_mutex); + 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(comp->dev, sizeof(*dai), GFP_KERNEL); + if (!dai) + return -ENOMEM; + dai->id = gbaudio_dai[i].id; + list_add(&dai->list, &info->dai_list); + } + + info->component = comp; + snd_soc_component_set_drvdata(comp, info); + gbcodec = info; + + device_init_wakeup(comp->dev, 1); + return 0; +} + +static int gbcodec_write(struct snd_soc_component *comp, unsigned int reg, + unsigned int value) +{ + return 0; +} + +static unsigned int gbcodec_read(struct snd_soc_component *comp, + unsigned int reg) +{ + return 0; +} + +static const struct snd_soc_component_driver soc_codec_dev_gbaudio = { + .probe = gbcodec_probe, + .read = gbcodec_read, + .write = gbcodec_write, +}; + +#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 devm_snd_soc_register_component(&pdev->dev, + &soc_codec_dev_gbaudio, + gbaudio_dai, ARRAY_SIZE(gbaudio_dai)); +} + +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", +#ifdef CONFIG_PM + .pm = &gbaudio_codec_pm_ops, +#endif + .of_match_table = greybus_asoc_machine_of_match, + }, + .probe = gbaudio_codec_probe, +}; +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");