audio-lnx: Rename folders to new flat structure.
Kernel audio drivers can be categorised into below folders.
asoc - ALSA based drivers,
asoc/codecs - codec drivers,
ipc - APR IPC communication drivers,
dsp - DSP low level drivers/Audio ION/ADSP Loader,
dsp/codecs - Native encoders and decoders,
soc - SoC based drivers(pinctrl/regmap/soundwire)
Restructure drivers to above folder format.
Include directories also follow above format.
Change-Id: I8fa0857baaacd47db126fb5c1f1f5ed7e886dbc0
Signed-off-by: Laxminath Kasam <lkasam@codeaurora.org>
diff --git a/asoc/msm-pcm-q6-v2.c b/asoc/msm-pcm-q6-v2.c
new file mode 100644
index 0000000..4910dec
--- /dev/null
+++ b/asoc/msm-pcm-q6-v2.c
@@ -0,0 +1,1884 @@
+/* Copyright (c) 2012-2017, The Linux Foundation. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 and
+ * only 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.
+ */
+
+
+#include <linux/init.h>
+#include <linux/err.h>
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/time.h>
+#include <linux/wait.h>
+#include <linux/platform_device.h>
+#include <linux/slab.h>
+#include <sound/core.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/control.h>
+#include <sound/timer.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/msm_audio.h>
+
+#include <linux/of_device.h>
+#include <sound/tlv.h>
+#include <sound/pcm_params.h>
+#include <dsp/msm_audio_ion.h>
+#include <dsp/q6audio-v2.h>
+
+#include "msm-pcm-q6-v2.h"
+#include "msm-pcm-routing-v2.h"
+#include "msm-qti-pp-config.h"
+
+enum stream_state {
+ IDLE = 0,
+ STOPPED,
+ RUNNING,
+};
+
+static struct audio_locks the_locks;
+
+#define PCM_MASTER_VOL_MAX_STEPS 0x2000
+static const DECLARE_TLV_DB_LINEAR(msm_pcm_vol_gain, 0,
+ PCM_MASTER_VOL_MAX_STEPS);
+
+struct snd_msm {
+ struct snd_card *card;
+ struct snd_pcm *pcm;
+};
+
+#define CMD_EOS_MIN_TIMEOUT_LENGTH 50
+#define CMD_EOS_TIMEOUT_MULTIPLIER (HZ * 50)
+#define MAX_PB_COPY_RETRIES 3
+
+static struct snd_pcm_hardware msm_pcm_hardware_capture = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = SNDRV_PCM_RATE_8000_384000,
+ .rate_min = 8000,
+ .rate_max = 384000,
+ .channels_min = 1,
+ .channels_max = 4,
+ .buffer_bytes_max = CAPTURE_MAX_NUM_PERIODS *
+ CAPTURE_MAX_PERIOD_SIZE,
+ .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE,
+ .period_bytes_max = CAPTURE_MAX_PERIOD_SIZE,
+ .periods_min = CAPTURE_MIN_NUM_PERIODS,
+ .periods_max = CAPTURE_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_pcm_hardware_playback = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME),
+ .formats = (SNDRV_PCM_FMTBIT_S16_LE |
+ SNDRV_PCM_FMTBIT_S24_LE |
+ SNDRV_PCM_FMTBIT_S24_3LE |
+ SNDRV_PCM_FMTBIT_S32_LE),
+ .rates = SNDRV_PCM_RATE_8000_384000,
+ .rate_min = 8000,
+ .rate_max = 384000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = PLAYBACK_MAX_NUM_PERIODS *
+ PLAYBACK_MAX_PERIOD_SIZE,
+ .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE,
+ .period_bytes_max = PLAYBACK_MAX_PERIOD_SIZE,
+ .periods_min = PLAYBACK_MIN_NUM_PERIODS,
+ .periods_max = PLAYBACK_MAX_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000,
+ 88200, 96000, 176400, 192000, 352800, 384000
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static void msm_pcm_route_event_handler(enum msm_pcm_routing_event event,
+ void *priv_data)
+{
+ struct msm_audio *prtd = priv_data;
+
+ WARN_ON(!prtd);
+
+ pr_debug("%s: event %x\n", __func__, event);
+
+ switch (event) {
+ case MSM_PCM_RT_EVT_BUF_RECFG:
+ q6asm_cmd(prtd->audio_client, CMD_PAUSE);
+ q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+ q6asm_run(prtd->audio_client, 0, 0, 0);
+ /* fallthrough */
+ default:
+ break;
+ }
+}
+
+static void event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_audio *prtd = priv;
+ struct snd_pcm_substream *substream = prtd->substream;
+ uint32_t *ptrmem = (uint32_t *)payload;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+ uint8_t buf_index;
+ struct snd_soc_pcm_runtime *rtd;
+ int ret = 0;
+
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2: {
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2\n");
+ pr_debug("Buffer Consumed = 0x%08x\n", *ptrmem);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start))
+ break;
+ if (!prtd->mmap_flag || prtd->reset_event)
+ break;
+ if (q6asm_is_cpu_buf_avail_nolock(IN,
+ prtd->audio_client,
+ &size, &idx)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp 2\n",
+ __func__, prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count, 0, 0, NO_TIMESTAMP);
+ }
+ break;
+ }
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ pr_debug("ASM_DATA_EVENT_RENDERED_EOS\n");
+ clear_bit(CMD_EOS, &prtd->cmd_pending);
+ wake_up(&the_locks.eos_wait);
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2: {
+ pr_debug("ASM_DATA_EVENT_READ_DONE_V2\n");
+ buf_index = q6asm_get_buf_index_from_token(token);
+ pr_debug("%s: token=0x%08x buf_index=0x%08x\n",
+ __func__, token, buf_index);
+ prtd->in_frame_info[buf_index].size = payload[4];
+ prtd->in_frame_info[buf_index].offset = payload[5];
+ /* assume data size = 0 during flushing */
+ if (prtd->in_frame_info[buf_index].size) {
+ prtd->pcm_irq_pos +=
+ prtd->in_frame_info[buf_index].size;
+ pr_debug("pcm_irq_pos=%d\n", prtd->pcm_irq_pos);
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ if (atomic_read(&prtd->in_count) <= prtd->periods)
+ atomic_inc(&prtd->in_count);
+ wake_up(&the_locks.read_wait);
+ if (prtd->mmap_flag &&
+ q6asm_is_cpu_buf_avail_nolock(OUT,
+ prtd->audio_client,
+ &size, &idx) &&
+ (substream->runtime->status->state ==
+ SNDRV_PCM_STATE_RUNNING))
+ q6asm_read_nolock(prtd->audio_client);
+ } else {
+ pr_debug("%s: reclaim flushed buf in_count %x\n",
+ __func__, atomic_read(&prtd->in_count));
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ if (prtd->mmap_flag) {
+ if (q6asm_is_cpu_buf_avail_nolock(OUT,
+ prtd->audio_client,
+ &size, &idx) &&
+ (substream->runtime->status->state ==
+ SNDRV_PCM_STATE_RUNNING))
+ q6asm_read_nolock(prtd->audio_client);
+ } else {
+ atomic_inc(&prtd->in_count);
+ }
+ if (atomic_read(&prtd->in_count) == prtd->periods) {
+ pr_info("%s: reclaimed all bufs\n", __func__);
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ wake_up(&the_locks.read_wait);
+ }
+ }
+ break;
+ }
+ case ASM_STREAM_PP_EVENT:
+ case ASM_STREAM_CMD_ENCDEC_EVENTS: {
+ pr_debug("%s: ASM_STREAM_EVENT (0x%x)\n", __func__, opcode);
+ if (!substream) {
+ pr_err("%s: substream is NULL.\n", __func__);
+ return;
+ }
+
+ rtd = substream->private_data;
+ if (!rtd) {
+ pr_err("%s: rtd is NULL\n", __func__);
+ return;
+ }
+
+ ret = msm_adsp_inform_mixer_ctl(rtd, payload);
+ if (ret) {
+ pr_err("%s: failed to inform mixer ctl. err = %d\n",
+ __func__, ret);
+ return;
+ }
+
+ break;
+ }
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN_V2:
+ if (substream->stream
+ != SNDRV_PCM_STREAM_PLAYBACK) {
+ atomic_set(&prtd->start, 1);
+ break;
+ }
+ if (prtd->mmap_flag) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ } else {
+ while (atomic_read(&prtd->out_needed)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__,
+ prtd->pcm_count);
+ q6asm_write_nolock(prtd->audio_client,
+ prtd->pcm_count,
+ 0, 0, NO_TIMESTAMP);
+ atomic_dec(&prtd->out_needed);
+ wake_up(&the_locks.write_wait);
+ };
+ }
+ atomic_set(&prtd->start, 1);
+ break;
+ case ASM_STREAM_CMD_REGISTER_PP_EVENTS:
+ pr_debug("%s: ASM_STREAM_CMD_REGISTER_PP_EVENTS:",
+ __func__);
+ break;
+ default:
+ pr_debug("%s:Payload = [0x%x]stat[0x%x]\n",
+ __func__, payload[0], payload[1]);
+ break;
+ }
+ }
+ break;
+ case RESET_EVENTS:
+ pr_debug("%s RESET_EVENTS\n", __func__);
+ prtd->pcm_irq_pos += prtd->pcm_count;
+ atomic_inc(&prtd->out_count);
+ atomic_inc(&prtd->in_count);
+ prtd->reset_event = true;
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+ wake_up(&the_locks.eos_wait);
+ wake_up(&the_locks.write_wait);
+ wake_up(&the_locks.read_wait);
+ break;
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_pcm_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ struct msm_plat_data *pdata;
+ struct snd_pcm_hw_params *params;
+ int ret;
+ uint32_t fmt_type = FORMAT_LINEAR_PCM;
+ uint16_t bits_per_sample;
+ uint16_t sample_word_size;
+
+ pdata = (struct msm_plat_data *)
+ dev_get_drvdata(soc_prtd->platform->dev);
+ if (!pdata) {
+ pr_err("%s: platform data not populated\n", __func__);
+ return -EINVAL;
+ }
+ if (!prtd || !prtd->audio_client) {
+ pr_err("%s: private data null or audio client freed\n",
+ __func__);
+ return -EINVAL;
+ }
+ params = &soc_prtd->dpcm[substream->stream].hw_params;
+
+ pr_debug("%s\n", __func__);
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+ if (prtd->enabled)
+ return 0;
+
+ prtd->audio_client->perf_mode = pdata->perf_mode;
+ pr_debug("%s: perf: %x\n", __func__, pdata->perf_mode);
+
+ switch (params_format(params)) {
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bits_per_sample = 32;
+ sample_word_size = 32;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bits_per_sample = 24;
+ sample_word_size = 32;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ bits_per_sample = 24;
+ sample_word_size = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ default:
+ bits_per_sample = 16;
+ sample_word_size = 16;
+ break;
+ }
+ if (prtd->compress_enable) {
+ fmt_type = FORMAT_GEN_COMPR;
+ pr_debug("%s: Compressed enabled!\n", __func__);
+ ret = q6asm_open_write_compressed(prtd->audio_client, fmt_type,
+ COMPRESSED_PASSTHROUGH_GEN);
+ if (ret < 0) {
+ pr_err("%s: q6asm_open_write_compressed failed (%d)\n",
+ __func__, ret);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return -ENOMEM;
+ }
+ } else {
+ ret = q6asm_open_write_v4(prtd->audio_client,
+ fmt_type, bits_per_sample);
+
+ if (ret < 0) {
+ pr_err("%s: q6asm_open_write_v4 failed (%d)\n",
+ __func__, ret);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return -ENOMEM;
+ }
+
+ ret = q6asm_send_cal(prtd->audio_client);
+ if (ret < 0)
+ pr_debug("%s : Send cal failed : %d", __func__, ret);
+ }
+ pr_debug("%s: session ID %d\n", __func__,
+ prtd->audio_client->session);
+ prtd->session_id = prtd->audio_client->session;
+
+ if (prtd->compress_enable) {
+ ret = msm_pcm_routing_reg_phy_compr_stream(
+ soc_prtd->dai_link->id,
+ prtd->audio_client->perf_mode,
+ prtd->session_id,
+ SNDRV_PCM_STREAM_PLAYBACK,
+ COMPRESSED_PASSTHROUGH_GEN);
+ } else {
+ ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->id,
+ prtd->audio_client->perf_mode,
+ prtd->session_id, substream->stream);
+ }
+ if (ret) {
+ pr_err("%s: stream reg failed ret:%d\n", __func__, ret);
+ return ret;
+ }
+ if (prtd->compress_enable) {
+ ret = q6asm_media_format_block_gen_compr(
+ prtd->audio_client, runtime->rate,
+ runtime->channels, !prtd->set_channel_map,
+ prtd->channel_map, bits_per_sample);
+ } else {
+ ret = q6asm_media_format_block_multi_ch_pcm_v4(
+ prtd->audio_client, runtime->rate,
+ runtime->channels, !prtd->set_channel_map,
+ prtd->channel_map, bits_per_sample,
+ sample_word_size, ASM_LITTLE_ENDIAN,
+ DEFAULT_QF);
+ }
+ if (ret < 0)
+ pr_info("%s: CMD Format block failed\n", __func__);
+
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ prtd->enabled = 1;
+ prtd->cmd_pending = 0;
+ prtd->cmd_interrupt = 0;
+
+ return 0;
+}
+
+static int msm_pcm_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_plat_data *pdata;
+ struct snd_pcm_hw_params *params;
+ struct msm_pcm_routing_evt event;
+ int ret = 0;
+ int i = 0;
+ uint16_t bits_per_sample = 16;
+ uint16_t sample_word_size;
+
+ pdata = (struct msm_plat_data *)
+ dev_get_drvdata(soc_prtd->platform->dev);
+ if (!pdata) {
+ pr_err("%s: platform data not populated\n", __func__);
+ return -EINVAL;
+ }
+ if (!prtd || !prtd->audio_client) {
+ pr_err("%s: private data null or audio client freed\n",
+ __func__);
+ return -EINVAL;
+ }
+
+ if (prtd->enabled == IDLE) {
+ pr_debug("%s:perf_mode=%d periods=%d\n", __func__,
+ pdata->perf_mode, runtime->periods);
+ params = &soc_prtd->dpcm[substream->stream].hw_params;
+ if ((params_format(params) == SNDRV_PCM_FORMAT_S24_LE) ||
+ (params_format(params) == SNDRV_PCM_FORMAT_S24_3LE))
+ bits_per_sample = 24;
+ else if (params_format(params) == SNDRV_PCM_FORMAT_S32_LE)
+ bits_per_sample = 32;
+
+ /* ULL mode is not supported in capture path */
+ if (pdata->perf_mode == LEGACY_PCM_MODE)
+ prtd->audio_client->perf_mode = LEGACY_PCM_MODE;
+ else
+ prtd->audio_client->perf_mode = LOW_LATENCY_PCM_MODE;
+
+ pr_debug("%s Opening %d-ch PCM read stream, perf_mode %d\n",
+ __func__, params_channels(params),
+ prtd->audio_client->perf_mode);
+
+ ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,
+ bits_per_sample, false);
+ if (ret < 0) {
+ pr_err("%s: q6asm_open_read failed\n", __func__);
+ q6asm_audio_client_free(prtd->audio_client);
+ prtd->audio_client = NULL;
+ return -ENOMEM;
+ }
+
+ ret = q6asm_send_cal(prtd->audio_client);
+ if (ret < 0)
+ pr_debug("%s : Send cal failed : %d", __func__, ret);
+
+ pr_debug("%s: session ID %d\n",
+ __func__, prtd->audio_client->session);
+ prtd->session_id = prtd->audio_client->session;
+ event.event_func = msm_pcm_route_event_handler;
+ event.priv_data = (void *) prtd;
+ ret = msm_pcm_routing_reg_phy_stream_v2(
+ soc_prtd->dai_link->id,
+ prtd->audio_client->perf_mode,
+ prtd->session_id, substream->stream,
+ event);
+ if (ret) {
+ pr_err("%s: stream reg failed ret:%d\n", __func__, ret);
+ return ret;
+ }
+ }
+
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+
+ if (prtd->enabled == IDLE || prtd->enabled == STOPPED) {
+ for (i = 0; i < runtime->periods; i++)
+ q6asm_read(prtd->audio_client);
+ prtd->periods = runtime->periods;
+ }
+
+ if (prtd->enabled != IDLE)
+ return 0;
+
+ switch (runtime->format) {
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bits_per_sample = 32;
+ sample_word_size = 32;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bits_per_sample = 24;
+ sample_word_size = 32;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ bits_per_sample = 24;
+ sample_word_size = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ default:
+ bits_per_sample = 16;
+ sample_word_size = 16;
+ break;
+ }
+
+ pr_debug("%s: Samp_rate = %d Channel = %d bit width = %d, word size = %d\n",
+ __func__, prtd->samp_rate, prtd->channel_mode,
+ bits_per_sample, sample_word_size);
+ ret = q6asm_enc_cfg_blk_pcm_format_support_v4(prtd->audio_client,
+ prtd->samp_rate,
+ prtd->channel_mode,
+ bits_per_sample,
+ sample_word_size,
+ ASM_LITTLE_ENDIAN,
+ DEFAULT_QF);
+ if (ret < 0)
+ pr_debug("%s: cmd cfg pcm was block failed", __func__);
+
+ prtd->enabled = RUNNING;
+
+ return ret;
+}
+
+static int msm_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ ret = q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ atomic_set(&prtd->start, 0);
+ if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) {
+ prtd->enabled = STOPPED;
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ break;
+ }
+ /* pending CMD_EOS isn't expected */
+ WARN_ON_ONCE(test_bit(CMD_EOS, &prtd->cmd_pending));
+ set_bit(CMD_EOS, &prtd->cmd_pending);
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ if (ret)
+ clear_bit(CMD_EOS, &prtd->cmd_pending);
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ ret = q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_pcm_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd;
+ int ret = 0;
+
+ prtd = kzalloc(sizeof(struct msm_audio), GFP_KERNEL);
+ if (prtd == NULL)
+ return -ENOMEM;
+
+ prtd->substream = substream;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+
+ prtd->audio_client->dev = soc_prtd->platform->dev;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ runtime->hw = msm_pcm_hardware_playback;
+
+ /* Capture path */
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ runtime->hw = msm_pcm_hardware_capture;
+ else {
+ pr_err("Invalid Stream type %d\n", substream->stream);
+ return -EINVAL;
+ }
+
+ ret = snd_pcm_hw_constraint_list(runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_sample_rates);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_list failed\n");
+ /* Ensure that buffer size is a multiple of period size */
+ ret = snd_pcm_hw_constraint_integer(runtime,
+ SNDRV_PCM_HW_PARAM_PERIODS);
+ if (ret < 0)
+ pr_info("snd_pcm_hw_constraint_integer failed\n");
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ PLAYBACK_MIN_NUM_PERIODS * PLAYBACK_MIN_PERIOD_SIZE,
+ PLAYBACK_MAX_NUM_PERIODS * PLAYBACK_MAX_PERIOD_SIZE);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes min max ret = %d\n",
+ ret);
+ }
+ }
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ ret = snd_pcm_hw_constraint_minmax(runtime,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+ CAPTURE_MIN_NUM_PERIODS * CAPTURE_MIN_PERIOD_SIZE,
+ CAPTURE_MAX_NUM_PERIODS * CAPTURE_MAX_PERIOD_SIZE);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes min max ret = %d\n",
+ ret);
+ }
+ }
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 32);
+ if (ret < 0) {
+ pr_err("constraint for period bytes step ret = %d\n",
+ ret);
+ }
+ ret = snd_pcm_hw_constraint_step(runtime, 0,
+ SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 32);
+ if (ret < 0) {
+ pr_err("constraint for buffer bytes step ret = %d\n",
+ ret);
+ }
+
+ prtd->enabled = IDLE;
+ prtd->dsp_cnt = 0;
+ prtd->set_channel_map = false;
+ prtd->reset_event = false;
+ runtime->private_data = prtd;
+ msm_adsp_init_mixer_ctl_pp_event_queue(soc_prtd);
+
+ return 0;
+}
+
+static int msm_pcm_playback_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer = 0;
+ char *bufptr = NULL;
+ void *data = NULL;
+ uint32_t idx = 0;
+ uint32_t size = 0;
+ uint32_t retries = 0;
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ fbytes = frames_to_bytes(runtime, frames);
+ pr_debug("%s: prtd->out_count = %d\n",
+ __func__, atomic_read(&prtd->out_count));
+
+ while ((fbytes > 0) && (retries < MAX_PB_COPY_RETRIES)) {
+ if (prtd->reset_event) {
+ pr_err("%s: In SSR return ENETRESET before wait\n",
+ __func__);
+ return -ENETRESET;
+ }
+
+ ret = wait_event_timeout(the_locks.write_wait,
+ (atomic_read(&prtd->out_count)), 5 * HZ);
+ if (!ret) {
+ pr_err("%s: wait_event_timeout failed\n", __func__);
+ ret = -ETIMEDOUT;
+ goto fail;
+ }
+ ret = 0;
+
+ if (prtd->reset_event) {
+ pr_err("%s: In SSR return ENETRESET after wait\n",
+ __func__);
+ return -ENETRESET;
+ }
+
+ if (!atomic_read(&prtd->out_count)) {
+ pr_err("%s: pcm stopped out_count 0\n", __func__);
+ return 0;
+ }
+
+ data = q6asm_is_cpu_buf_avail(IN, prtd->audio_client, &size,
+ &idx);
+ if (data == NULL) {
+ retries++;
+ continue;
+ } else {
+ retries = 0;
+ }
+
+ if (fbytes > size)
+ xfer = size;
+ else
+ xfer = fbytes;
+
+ bufptr = data;
+ if (bufptr) {
+ pr_debug("%s:fbytes =%d: xfer=%d size=%d\n",
+ __func__, fbytes, xfer, size);
+ if (copy_from_user(bufptr, buf, xfer)) {
+ ret = -EFAULT;
+ pr_err("%s: copy_from_user failed\n",
+ __func__);
+ q6asm_cpu_buf_release(IN, prtd->audio_client);
+ goto fail;
+ }
+ buf += xfer;
+ fbytes -= xfer;
+ pr_debug("%s:fbytes = %d: xfer=%d\n", __func__, fbytes,
+ xfer);
+ if (atomic_read(&prtd->start)) {
+ pr_debug("%s:writing %d bytes of buffer to dsp\n",
+ __func__, xfer);
+ ret = q6asm_write(prtd->audio_client, xfer,
+ 0, 0, NO_TIMESTAMP);
+ if (ret < 0) {
+ ret = -EFAULT;
+ q6asm_cpu_buf_release(IN,
+ prtd->audio_client);
+ goto fail;
+ }
+ } else
+ atomic_inc(&prtd->out_needed);
+ atomic_dec(&prtd->out_count);
+ }
+ }
+fail:
+ if (retries >= MAX_PB_COPY_RETRIES)
+ ret = -ENOMEM;
+
+ return ret;
+}
+
+static int msm_pcm_playback_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ uint32_t timeout;
+ int dir = 0;
+ int ret = 0;
+
+ pr_debug("%s: cmd_pending 0x%lx\n", __func__, prtd->cmd_pending);
+
+ if (prtd->audio_client) {
+ dir = IN;
+
+ /* determine timeout length */
+ if (runtime->frame_bits == 0 || runtime->rate == 0) {
+ timeout = CMD_EOS_MIN_TIMEOUT_LENGTH;
+ } else {
+ timeout = (runtime->period_size *
+ CMD_EOS_TIMEOUT_MULTIPLIER) /
+ ((runtime->frame_bits / 8) *
+ runtime->rate);
+ if (timeout < CMD_EOS_MIN_TIMEOUT_LENGTH)
+ timeout = CMD_EOS_MIN_TIMEOUT_LENGTH;
+ }
+ pr_debug("%s: CMD_EOS timeout is %d\n", __func__, timeout);
+
+ ret = wait_event_timeout(the_locks.eos_wait,
+ !test_bit(CMD_EOS, &prtd->cmd_pending),
+ timeout);
+ if (!ret)
+ pr_err("%s: CMD_EOS failed, cmd_pending 0x%lx\n",
+ __func__, prtd->cmd_pending);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ }
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ msm_adsp_clean_mixer_ctl_pp_event_queue(soc_prtd);
+ kfree(prtd);
+ runtime->private_data = NULL;
+
+ return 0;
+}
+
+static int msm_pcm_capture_copy(struct snd_pcm_substream *substream,
+ int channel, snd_pcm_uframes_t hwoff, void __user *buf,
+ snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+ int fbytes = 0;
+ int xfer;
+ char *bufptr;
+ void *data = NULL;
+ static uint32_t idx;
+ static uint32_t size;
+ uint32_t offset = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = substream->runtime->private_data;
+
+
+ pr_debug("%s\n", __func__);
+ fbytes = frames_to_bytes(runtime, frames);
+
+ pr_debug("appl_ptr %d\n", (int)runtime->control->appl_ptr);
+ pr_debug("hw_ptr %d\n", (int)runtime->status->hw_ptr);
+ pr_debug("avail_min %d\n", (int)runtime->control->avail_min);
+
+ if (prtd->reset_event) {
+ pr_err("%s: In SSR return ENETRESET before wait\n", __func__);
+ return -ENETRESET;
+ }
+ ret = wait_event_timeout(the_locks.read_wait,
+ (atomic_read(&prtd->in_count)), 5 * HZ);
+ if (!ret) {
+ pr_debug("%s: wait_event_timeout failed\n", __func__);
+ goto fail;
+ }
+ if (prtd->reset_event) {
+ pr_err("%s: In SSR return ENETRESET after wait\n", __func__);
+ return -ENETRESET;
+ }
+ if (!atomic_read(&prtd->in_count)) {
+ pr_debug("%s: pcm stopped in_count 0\n", __func__);
+ return 0;
+ }
+ pr_debug("Checking if valid buffer is available...%pK\n",
+ data);
+ data = q6asm_is_cpu_buf_avail(OUT, prtd->audio_client, &size, &idx);
+ bufptr = data;
+ pr_debug("Size = %d\n", size);
+ pr_debug("fbytes = %d\n", fbytes);
+ pr_debug("idx = %d\n", idx);
+ if (bufptr) {
+ xfer = fbytes;
+ if (xfer > size)
+ xfer = size;
+ offset = prtd->in_frame_info[idx].offset;
+ pr_debug("Offset value = %d\n", offset);
+ if (copy_to_user(buf, bufptr+offset, xfer)) {
+ pr_err("Failed to copy buf to user\n");
+ ret = -EFAULT;
+ q6asm_cpu_buf_release(OUT, prtd->audio_client);
+ goto fail;
+ }
+ fbytes -= xfer;
+ size -= xfer;
+ prtd->in_frame_info[idx].offset += xfer;
+ pr_debug("%s:fbytes = %d: size=%d: xfer=%d\n",
+ __func__, fbytes, size, xfer);
+ pr_debug(" Sending next buffer to dsp\n");
+ memset(&prtd->in_frame_info[idx], 0,
+ sizeof(struct msm_audio_in_frame_info));
+ atomic_dec(&prtd->in_count);
+ ret = q6asm_read(prtd->audio_client);
+ if (ret < 0) {
+ pr_err("q6asm read failed\n");
+ ret = -EFAULT;
+ q6asm_cpu_buf_release(OUT, prtd->audio_client);
+ goto fail;
+ }
+ } else
+ pr_err("No valid buffer\n");
+
+ pr_debug("Returning from capture_copy... %d\n", ret);
+fail:
+ return ret;
+}
+
+static int msm_pcm_capture_close(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = runtime->private_data;
+ int dir = OUT;
+
+ pr_debug("%s\n", __func__);
+ if (prtd->audio_client) {
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ q6asm_audio_client_free(prtd->audio_client);
+ }
+
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ kfree(prtd);
+ runtime->private_data = NULL;
+
+ return 0;
+}
+
+static int msm_pcm_copy(struct snd_pcm_substream *substream, int a,
+ snd_pcm_uframes_t hwoff, void __user *buf, snd_pcm_uframes_t frames)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_copy(substream, a, hwoff, buf, frames);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_copy(substream, a, hwoff, buf, frames);
+ return ret;
+}
+
+static int msm_pcm_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_close(substream);
+ return ret;
+}
+
+static int msm_pcm_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_pcm_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_pcm_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_pcm_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("pcm_irq_pos = %d\n", prtd->pcm_irq_pos);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_pcm_mmap(struct snd_pcm_substream *substream,
+ struct vm_area_struct *vma)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct audio_client *ac = prtd->audio_client;
+ struct audio_port_data *apd = ac->port;
+ struct audio_buffer *ab;
+ int dir = -1;
+
+ prtd->mmap_flag = 1;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ dir = OUT;
+ ab = &(apd[dir].buf[0]);
+
+ return msm_audio_ion_mmap(ab, vma);
+}
+
+static int msm_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct snd_dma_buffer *dma_buf = &substream->dma_buffer;
+ struct audio_buffer *buf;
+ int dir, ret;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ dir = IN;
+ else
+ dir = OUT;
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ (params_buffer_bytes(params) / params_periods(params)),
+ params_periods(params));
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
+ ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].buf;
+ if (buf == NULL || buf[0].data == NULL)
+ return -ENOMEM;
+
+ pr_debug("%s:buf = %pK\n", __func__, buf);
+ dma_buf->dev.type = SNDRV_DMA_TYPE_DEV;
+ dma_buf->dev.dev = substream->pcm->card->dev;
+ dma_buf->private_data = NULL;
+ dma_buf->area = buf[0].data;
+ dma_buf->addr = buf[0].phys;
+ dma_buf->bytes = params_buffer_bytes(params);
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static const struct snd_pcm_ops msm_pcm_ops = {
+ .open = msm_pcm_open,
+ .copy = msm_pcm_copy,
+ .hw_params = msm_pcm_hw_params,
+ .close = msm_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .prepare = msm_pcm_prepare,
+ .trigger = msm_pcm_trigger,
+ .pointer = msm_pcm_pointer,
+ .mmap = msm_pcm_mmap,
+};
+
+static int msm_pcm_adsp_stream_cmd_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *pcm = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_platform *platform = snd_soc_component_to_platform(pcm);
+ struct msm_plat_data *pdata = dev_get_drvdata(platform->dev);
+ struct snd_pcm_substream *substream;
+ struct msm_audio *prtd;
+ int ret = 0;
+ struct msm_adsp_event_data *event_data = NULL;
+
+ if (!pdata) {
+ pr_err("%s pdata is NULL\n", __func__);
+ ret = -ENODEV;
+ goto done;
+ }
+
+ substream = pdata->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (!substream) {
+ pr_err("%s substream not found\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if (!substream->runtime) {
+ pr_err("%s substream runtime not found\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ prtd = substream->runtime->private_data;
+ if (prtd->audio_client == NULL) {
+ pr_err("%s prtd is null.\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ event_data = (struct msm_adsp_event_data *)ucontrol->value.bytes.data;
+ if ((event_data->event_type < ADSP_STREAM_PP_EVENT) ||
+ (event_data->event_type >= ADSP_STREAM_EVENT_MAX)) {
+ pr_err("%s: invalid event_type=%d",
+ __func__, event_data->event_type);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if ((sizeof(struct msm_adsp_event_data) + event_data->payload_len) >=
+ sizeof(ucontrol->value.bytes.data)) {
+ pr_err("%s param length=%d exceeds limit",
+ __func__, event_data->payload_len);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ret = q6asm_send_stream_cmd(prtd->audio_client, event_data);
+ if (ret < 0)
+ pr_err("%s: failed to send stream event cmd, err = %d\n",
+ __func__, ret);
+done:
+ return ret;
+}
+
+static int msm_pcm_add_audio_adsp_stream_cmd_control(
+ struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = DSP_STREAM_CMD;
+ const char *deviceNo = "NN";
+ char *mixer_str = NULL;
+ int ctl_len = 0, ret = 0;
+ struct snd_kcontrol_new fe_audio_adsp_stream_cmd_config_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_adsp_stream_cmd_info,
+ .put = msm_pcm_adsp_stream_cmd_put,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s rtd is NULL\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1;
+ mixer_str = kzalloc(ctl_len, GFP_KERNEL);
+ if (!mixer_str) {
+ ret = -ENOMEM;
+ goto done;
+ }
+
+ snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device);
+ fe_audio_adsp_stream_cmd_config_control[0].name = mixer_str;
+ fe_audio_adsp_stream_cmd_config_control[0].private_value =
+ rtd->dai_link->id;
+ pr_debug("Registering new mixer ctl %s\n", mixer_str);
+ ret = snd_soc_add_platform_controls(rtd->platform,
+ fe_audio_adsp_stream_cmd_config_control,
+ ARRAY_SIZE(fe_audio_adsp_stream_cmd_config_control));
+ if (ret < 0)
+ pr_err("%s: failed add ctl %s. err = %d\n",
+ __func__, mixer_str, ret);
+
+ kfree(mixer_str);
+done:
+ return ret;
+}
+
+static int msm_pcm_add_audio_adsp_stream_callback_control(
+ struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = DSP_STREAM_CALLBACK;
+ const char *deviceNo = "NN";
+ char *mixer_str = NULL;
+ int ctl_len = 0, ret = 0;
+ struct snd_kcontrol *kctl;
+
+ struct snd_kcontrol_new fe_audio_adsp_callback_config_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_adsp_stream_callback_info,
+ .get = msm_adsp_stream_callback_get,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s NULL rtd\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ pr_debug("%s: added new pcm FE with name %s, id %d, cpu dai %s, device no %d\n",
+ __func__, rtd->dai_link->name, rtd->dai_link->id,
+ rtd->dai_link->cpu_dai_name, rtd->pcm->device);
+ ctl_len = strlen(mixer_ctl_name) + 1 + strlen(deviceNo) + 1;
+ mixer_str = kzalloc(ctl_len, GFP_KERNEL);
+ if (!mixer_str) {
+ ret = -ENOMEM;
+ goto done;
+ }
+
+ snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device);
+ fe_audio_adsp_callback_config_control[0].name = mixer_str;
+ fe_audio_adsp_callback_config_control[0].private_value =
+ rtd->dai_link->id;
+ pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str);
+ ret = snd_soc_add_platform_controls(rtd->platform,
+ fe_audio_adsp_callback_config_control,
+ ARRAY_SIZE(fe_audio_adsp_callback_config_control));
+ if (ret < 0) {
+ pr_err("%s: failed to add ctl %s. err = %d\n",
+ __func__, mixer_str, ret);
+ ret = -EINVAL;
+ goto free_mixer_str;
+ }
+
+ kctl = snd_soc_card_get_kcontrol(rtd->card, mixer_str);
+ if (!kctl) {
+ pr_err("%s: failed to get kctl %s.\n", __func__, mixer_str);
+ ret = -EINVAL;
+ goto free_mixer_str;
+ }
+
+ kctl->private_data = NULL;
+
+free_mixer_str:
+ kfree(mixer_str);
+done:
+ return ret;
+}
+
+static int msm_pcm_set_volume(struct msm_audio *prtd, uint32_t volume)
+{
+ int rc = 0;
+
+ if (prtd && prtd->audio_client) {
+ pr_debug("%s: channels %d volume 0x%x\n", __func__,
+ prtd->channel_mode, volume);
+ rc = q6asm_set_volume(prtd->audio_client, volume);
+ if (rc < 0) {
+ pr_err("%s: Send Volume command failed rc=%d\n",
+ __func__, rc);
+ }
+ }
+ return rc;
+}
+
+static int msm_pcm_volume_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
+ struct snd_pcm_substream *substream =
+ vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ struct msm_audio *prtd;
+
+ pr_debug("%s\n", __func__);
+ if (!substream) {
+ pr_err("%s substream not found\n", __func__);
+ return -ENODEV;
+ }
+ if (!substream->runtime) {
+ pr_err("%s substream runtime not found\n", __func__);
+ return 0;
+ }
+ prtd = substream->runtime->private_data;
+ if (prtd)
+ ucontrol->value.integer.value[0] = prtd->volume;
+ return 0;
+}
+
+static int msm_pcm_volume_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ struct snd_pcm_volume *vol = snd_kcontrol_chip(kcontrol);
+ struct snd_pcm_substream *substream =
+ vol->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ struct msm_audio *prtd;
+ int volume = ucontrol->value.integer.value[0];
+
+ pr_debug("%s: volume : 0x%x\n", __func__, volume);
+ if (!substream) {
+ pr_err("%s substream not found\n", __func__);
+ return -ENODEV;
+ }
+ if (!substream->runtime) {
+ pr_err("%s substream runtime not found\n", __func__);
+ return 0;
+ }
+ prtd = substream->runtime->private_data;
+ if (prtd) {
+ rc = msm_pcm_set_volume(prtd, volume);
+ prtd->volume = volume;
+ }
+ return rc;
+}
+
+static int msm_pcm_add_volume_control(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret = 0;
+ struct snd_pcm *pcm = rtd->pcm;
+ struct snd_pcm_volume *volume_info;
+ struct snd_kcontrol *kctl;
+
+ dev_dbg(rtd->dev, "%s, Volume control add\n", __func__);
+ ret = snd_pcm_add_volume_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ NULL, 1, rtd->dai_link->id,
+ &volume_info);
+ if (ret < 0) {
+ pr_err("%s volume control failed ret %d\n", __func__, ret);
+ return ret;
+ }
+ kctl = volume_info->kctl;
+ kctl->put = msm_pcm_volume_ctl_put;
+ kctl->get = msm_pcm_volume_ctl_get;
+ kctl->tlv.p = msm_pcm_vol_gain;
+ return 0;
+}
+
+static int msm_pcm_compress_ctl_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0x2000;
+ return 0;
+}
+
+static int msm_pcm_compress_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_platform *platform = snd_soc_component_to_platform(comp);
+ struct msm_plat_data *pdata = dev_get_drvdata(platform->dev);
+ struct snd_pcm_substream *substream;
+ struct msm_audio *prtd;
+
+ if (!pdata) {
+ pr_err("%s pdata is NULL\n", __func__);
+ return -ENODEV;
+ }
+ substream = pdata->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ if (!substream) {
+ pr_err("%s substream not found\n", __func__);
+ return -EINVAL;
+ }
+ if (!substream->runtime) {
+ pr_err("%s substream runtime not found\n", __func__);
+ return 0;
+ }
+ prtd = substream->runtime->private_data;
+ if (prtd)
+ ucontrol->value.integer.value[0] = prtd->compress_enable;
+ return 0;
+}
+
+static int msm_pcm_compress_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int rc = 0;
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ struct snd_soc_platform *platform = snd_soc_component_to_platform(comp);
+ struct msm_plat_data *pdata = dev_get_drvdata(platform->dev);
+ struct snd_pcm_substream *substream;
+ struct msm_audio *prtd;
+ int compress = ucontrol->value.integer.value[0];
+
+ if (!pdata) {
+ pr_err("%s pdata is NULL\n", __func__);
+ return -ENODEV;
+ }
+ substream = pdata->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream;
+ pr_debug("%s: compress : 0x%x\n", __func__, compress);
+ if (!substream) {
+ pr_err("%s substream not found\n", __func__);
+ return -EINVAL;
+ }
+ if (!substream->runtime) {
+ pr_err("%s substream runtime not found\n", __func__);
+ return 0;
+ }
+ prtd = substream->runtime->private_data;
+ if (prtd) {
+ pr_debug("%s: setting compress flag to 0x%x\n",
+ __func__, compress);
+ prtd->compress_enable = compress;
+ }
+ return rc;
+}
+
+static int msm_pcm_add_compress_control(struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = "Playback ";
+ const char *mixer_ctl_end_name = " Compress";
+ const char *deviceNo = "NN";
+ char *mixer_str = NULL;
+ int ctl_len;
+ int ret = 0;
+ struct msm_plat_data *pdata;
+ struct snd_kcontrol_new pcm_compress_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_pcm_compress_ctl_info,
+ .get = msm_pcm_compress_ctl_get,
+ .put = msm_pcm_compress_ctl_put,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s: NULL rtd\n", __func__);
+ return -EINVAL;
+ }
+
+ ctl_len = strlen(mixer_ctl_name) + strlen(deviceNo) +
+ strlen(mixer_ctl_end_name) + 1;
+ mixer_str = kzalloc(ctl_len, GFP_KERNEL);
+
+ if (!mixer_str)
+ return -ENOMEM;
+
+ snprintf(mixer_str, ctl_len, "%s%d%s", mixer_ctl_name,
+ rtd->pcm->device, mixer_ctl_end_name);
+
+ pcm_compress_control[0].name = mixer_str;
+ pcm_compress_control[0].private_value = rtd->dai_link->id;
+ pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str);
+ pdata = dev_get_drvdata(rtd->platform->dev);
+ if (pdata) {
+ if (!pdata->pcm) {
+ pdata->pcm = rtd->pcm;
+ snd_soc_add_platform_controls(rtd->platform,
+ pcm_compress_control,
+ ARRAY_SIZE
+ (pcm_compress_control));
+ pr_debug("%s: add control success plt = %pK\n",
+ __func__, rtd->platform);
+ }
+ } else {
+ pr_err("%s: NULL pdata\n", __func__);
+ ret = -EINVAL;
+ }
+ kfree(mixer_str);
+ return ret;
+}
+
+static int msm_pcm_chmap_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int i;
+ struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ struct snd_pcm_substream *substream;
+ struct msm_audio *prtd;
+
+ pr_debug("%s", __func__);
+ substream = snd_pcm_chmap_substream(info, idx);
+ if (!substream)
+ return -ENODEV;
+ if (!substream->runtime)
+ return 0;
+
+ prtd = substream->runtime->private_data;
+ if (prtd) {
+ prtd->set_channel_map = true;
+ for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++)
+ prtd->channel_map[i] =
+ (char)(ucontrol->value.integer.value[i]);
+ }
+ return 0;
+}
+
+static int msm_pcm_chmap_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ int i;
+ struct snd_pcm_chmap *info = snd_kcontrol_chip(kcontrol);
+ unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id);
+ struct snd_pcm_substream *substream;
+ struct msm_audio *prtd;
+
+ pr_debug("%s", __func__);
+ substream = snd_pcm_chmap_substream(info, idx);
+ if (!substream)
+ return -ENODEV;
+ memset(ucontrol->value.integer.value, 0,
+ sizeof(ucontrol->value.integer.value));
+ if (!substream->runtime)
+ return 0; /* no channels set */
+
+ prtd = substream->runtime->private_data;
+
+ if (prtd && prtd->set_channel_map == true) {
+ for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++)
+ ucontrol->value.integer.value[i] =
+ (int)prtd->channel_map[i];
+ } else {
+ for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++)
+ ucontrol->value.integer.value[i] = 0;
+ }
+
+ return 0;
+}
+
+static int msm_pcm_add_chmap_controls(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ struct snd_pcm_chmap *chmap_info;
+ struct snd_kcontrol *kctl;
+ char device_num[12];
+ int i, ret = 0;
+
+ pr_debug("%s, Channel map cntrl add\n", __func__);
+ ret = snd_pcm_add_chmap_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ snd_pcm_std_chmaps,
+ PCM_FORMAT_MAX_NUM_CHANNEL, 0,
+ &chmap_info);
+ if (ret < 0) {
+ pr_err("%s, channel map cntrl add failed\n", __func__);
+ return ret;
+ }
+ kctl = chmap_info->kctl;
+ for (i = 0; i < kctl->count; i++)
+ kctl->vd[i].access |= SNDRV_CTL_ELEM_ACCESS_WRITE;
+ snprintf(device_num, sizeof(device_num), "%d", pcm->device);
+ strlcat(kctl->id.name, device_num, sizeof(kctl->id.name));
+ pr_debug("%s, Overwriting channel map control name to: %s\n",
+ __func__, kctl->id.name);
+ kctl->put = msm_pcm_chmap_ctl_put;
+ kctl->get = msm_pcm_chmap_ctl_get;
+ return 0;
+}
+
+static int msm_pcm_playback_app_type_cfg_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u64 fe_id = kcontrol->private_value;
+ int session_type = SESSION_TYPE_RX;
+ int be_id = ucontrol->value.integer.value[3];
+ struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000};
+ int ret = 0;
+
+ cfg_data.app_type = ucontrol->value.integer.value[0];
+ cfg_data.acdb_dev_id = ucontrol->value.integer.value[1];
+ if (ucontrol->value.integer.value[2] != 0)
+ cfg_data.sample_rate = ucontrol->value.integer.value[2];
+ pr_debug("%s: fe_id- %llu session_type- %d be_id- %d app_type- %d acdb_dev_id- %d sample_rate- %d\n",
+ __func__, fe_id, session_type, be_id,
+ cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate);
+ ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type,
+ be_id, &cfg_data);
+ if (ret < 0)
+ pr_err("%s: msm_pcm_routing_reg_stream_app_type_cfg failed returned %d\n",
+ __func__, ret);
+
+ return ret;
+}
+
+static int msm_pcm_playback_app_type_cfg_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u64 fe_id = kcontrol->private_value;
+ int session_type = SESSION_TYPE_RX;
+ int be_id = 0;
+ struct msm_pcm_stream_app_type_cfg cfg_data = {0};
+ int ret = 0;
+
+ ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type,
+ &be_id, &cfg_data);
+ if (ret < 0) {
+ pr_err("%s: msm_pcm_routing_get_stream_app_type_cfg failed returned %d\n",
+ __func__, ret);
+ goto done;
+ }
+
+ ucontrol->value.integer.value[0] = cfg_data.app_type;
+ ucontrol->value.integer.value[1] = cfg_data.acdb_dev_id;
+ ucontrol->value.integer.value[2] = cfg_data.sample_rate;
+ ucontrol->value.integer.value[3] = be_id;
+ pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n",
+ __func__, fe_id, session_type, be_id,
+ cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate);
+done:
+ return ret;
+}
+
+static int msm_pcm_capture_app_type_cfg_ctl_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u64 fe_id = kcontrol->private_value;
+ int session_type = SESSION_TYPE_TX;
+ int be_id = ucontrol->value.integer.value[3];
+ struct msm_pcm_stream_app_type_cfg cfg_data = {0, 0, 48000};
+ int ret = 0;
+
+ cfg_data.app_type = ucontrol->value.integer.value[0];
+ cfg_data.acdb_dev_id = ucontrol->value.integer.value[1];
+ if (ucontrol->value.integer.value[2] != 0)
+ cfg_data.sample_rate = ucontrol->value.integer.value[2];
+ pr_debug("%s: fe_id- %llu session_type- %d be_id- %d app_type- %d acdb_dev_id- %d sample_rate- %d\n",
+ __func__, fe_id, session_type, be_id,
+ cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate);
+ ret = msm_pcm_routing_reg_stream_app_type_cfg(fe_id, session_type,
+ be_id, &cfg_data);
+ if (ret < 0)
+ pr_err("%s: msm_pcm_routing_reg_stream_app_type_cfg failed returned %d\n",
+ __func__, ret);
+
+ return ret;
+}
+
+static int msm_pcm_capture_app_type_cfg_ctl_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ u64 fe_id = kcontrol->private_value;
+ int session_type = SESSION_TYPE_TX;
+ int be_id = 0;
+ struct msm_pcm_stream_app_type_cfg cfg_data = {0};
+ int ret = 0;
+
+ ret = msm_pcm_routing_get_stream_app_type_cfg(fe_id, session_type,
+ &be_id, &cfg_data);
+ if (ret < 0) {
+ pr_err("%s: msm_pcm_routing_get_stream_app_type_cfg failed returned %d\n",
+ __func__, ret);
+ goto done;
+ }
+
+ ucontrol->value.integer.value[0] = cfg_data.app_type;
+ ucontrol->value.integer.value[1] = cfg_data.acdb_dev_id;
+ ucontrol->value.integer.value[2] = cfg_data.sample_rate;
+ ucontrol->value.integer.value[3] = be_id;
+ pr_debug("%s: fedai_id %llu, session_type %d, be_id %d, app_type %d, acdb_dev_id %d, sample_rate %d\n",
+ __func__, fe_id, session_type, be_id,
+ cfg_data.app_type, cfg_data.acdb_dev_id, cfg_data.sample_rate);
+done:
+ return ret;
+}
+
+static int msm_pcm_add_app_type_controls(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_pcm *pcm = rtd->pcm;
+ struct snd_pcm_usr *app_type_info;
+ struct snd_kcontrol *kctl;
+ const char *playback_mixer_ctl_name = "Audio Stream";
+ const char *capture_mixer_ctl_name = "Audio Stream Capture";
+ const char *deviceNo = "NN";
+ const char *suffix = "App Type Cfg";
+ int ctl_len, ret = 0;
+
+ if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+ ctl_len = strlen(playback_mixer_ctl_name) + 1 +
+ strlen(deviceNo) + 1 + strlen(suffix) + 1;
+ pr_debug("%s: Playback app type cntrl add\n", __func__);
+ ret = snd_pcm_add_usr_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ NULL, 1, ctl_len, rtd->dai_link->id,
+ &app_type_info);
+ if (ret < 0) {
+ pr_err("%s: playback app type cntrl add failed: %d\n",
+ __func__, ret);
+ return ret;
+ }
+ kctl = app_type_info->kctl;
+ snprintf(kctl->id.name, ctl_len, "%s %d %s",
+ playback_mixer_ctl_name, rtd->pcm->device, suffix);
+ kctl->put = msm_pcm_playback_app_type_cfg_ctl_put;
+ kctl->get = msm_pcm_playback_app_type_cfg_ctl_get;
+ }
+
+ if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+ ctl_len = strlen(capture_mixer_ctl_name) + 1 +
+ strlen(deviceNo) + 1 + strlen(suffix) + 1;
+ pr_debug("%s: Capture app type cntrl add\n", __func__);
+ ret = snd_pcm_add_usr_ctls(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ NULL, 1, ctl_len, rtd->dai_link->id,
+ &app_type_info);
+ if (ret < 0) {
+ pr_err("%s: capture app type cntrl add failed: %d\n",
+ __func__, ret);
+ return ret;
+ }
+ kctl = app_type_info->kctl;
+ snprintf(kctl->id.name, ctl_len, "%s %d %s",
+ capture_mixer_ctl_name, rtd->pcm->device, suffix);
+ kctl->put = msm_pcm_capture_app_type_cfg_ctl_put;
+ kctl->get = msm_pcm_capture_app_type_cfg_ctl_get;
+ }
+
+ return 0;
+}
+
+static int msm_pcm_add_controls(struct snd_soc_pcm_runtime *rtd)
+{
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ ret = msm_pcm_add_chmap_controls(rtd);
+ if (ret)
+ pr_err("%s: pcm add controls failed:%d\n", __func__, ret);
+ ret = msm_pcm_add_app_type_controls(rtd);
+ if (ret)
+ pr_err("%s: pcm add app type controls failed:%d\n",
+ __func__, ret);
+ return ret;
+}
+
+static int msm_asoc_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+ struct snd_card *card = rtd->card->snd_card;
+ int ret = 0;
+
+ if (!card->dev->coherent_dma_mask)
+ card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+
+ ret = msm_pcm_add_controls(rtd);
+ if (ret) {
+ pr_err("%s, kctl add failed:%d\n", __func__, ret);
+ return ret;
+ }
+
+ ret = msm_pcm_add_volume_control(rtd);
+ if (ret)
+ pr_err("%s: Could not add pcm Volume Control %d\n",
+ __func__, ret);
+
+ ret = msm_pcm_add_compress_control(rtd);
+ if (ret)
+ pr_err("%s: Could not add pcm Compress Control %d\n",
+ __func__, ret);
+
+ ret = msm_pcm_add_audio_adsp_stream_cmd_control(rtd);
+ if (ret)
+ pr_err("%s: Could not add pcm ADSP Stream Cmd Control\n",
+ __func__);
+
+ ret = msm_pcm_add_audio_adsp_stream_callback_control(rtd);
+ if (ret)
+ pr_err("%s: Could not add pcm ADSP Stream Callback Control\n",
+ __func__);
+
+ return ret;
+}
+
+static snd_pcm_sframes_t msm_pcm_delay_blk(struct snd_pcm_substream *substream,
+ struct snd_soc_dai *dai)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct msm_audio *prtd = runtime->private_data;
+ struct audio_client *ac = prtd->audio_client;
+ snd_pcm_sframes_t frames;
+ int ret;
+
+ ret = q6asm_get_path_delay(prtd->audio_client);
+ if (ret) {
+ pr_err("%s: get_path_delay failed, ret=%d\n", __func__, ret);
+ return 0;
+ }
+
+ /* convert microseconds to frames */
+ frames = ac->path_delay / 1000 * runtime->rate / 1000;
+
+ /* also convert the remainder from the initial division */
+ frames += ac->path_delay % 1000 * runtime->rate / 1000000;
+
+ /* overcompensate for the loss of precision (empirical) */
+ frames += 2;
+
+ return frames;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_pcm_ops,
+ .pcm_new = msm_asoc_pcm_new,
+ .delay_blk = msm_pcm_delay_blk,
+};
+
+static int msm_pcm_probe(struct platform_device *pdev)
+{
+ int rc;
+ int id;
+ struct msm_plat_data *pdata;
+ const char *latency_level;
+
+ rc = of_property_read_u32(pdev->dev.of_node,
+ "qcom,msm-pcm-dsp-id", &id);
+ if (rc) {
+ dev_err(&pdev->dev, "%s: qcom,msm-pcm-dsp-id missing in DT node\n",
+ __func__);
+ return rc;
+ }
+
+ pdata = kzalloc(sizeof(struct msm_plat_data), GFP_KERNEL);
+ if (!pdata)
+ return -ENOMEM;
+
+ if (of_property_read_bool(pdev->dev.of_node,
+ "qcom,msm-pcm-low-latency")) {
+
+ pdata->perf_mode = LOW_LATENCY_PCM_MODE;
+ rc = of_property_read_string(pdev->dev.of_node,
+ "qcom,latency-level", &latency_level);
+ if (!rc) {
+ if (!strcmp(latency_level, "ultra"))
+ pdata->perf_mode = ULTRA_LOW_LATENCY_PCM_MODE;
+ else if (!strcmp(latency_level, "ull-pp"))
+ pdata->perf_mode =
+ ULL_POST_PROCESSING_PCM_MODE;
+ }
+ } else {
+ pdata->perf_mode = LEGACY_PCM_MODE;
+ }
+
+ dev_set_drvdata(&pdev->dev, pdata);
+
+
+ dev_dbg(&pdev->dev, "%s: dev name %s\n",
+ __func__, dev_name(&pdev->dev));
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_pcm_remove(struct platform_device *pdev)
+{
+ struct msm_plat_data *pdata;
+
+ pdata = dev_get_drvdata(&pdev->dev);
+ kfree(pdata);
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+static const struct of_device_id msm_pcm_dt_match[] = {
+ {.compatible = "qcom,msm-pcm-dsp"},
+ {}
+};
+MODULE_DEVICE_TABLE(of, msm_pcm_dt_match);
+
+static struct platform_driver msm_pcm_driver = {
+ .driver = {
+ .name = "msm-pcm-dsp",
+ .owner = THIS_MODULE,
+ .of_match_table = msm_pcm_dt_match,
+ },
+ .probe = msm_pcm_probe,
+ .remove = msm_pcm_remove,
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ init_waitqueue_head(&the_locks.enable_wait);
+ init_waitqueue_head(&the_locks.eos_wait);
+ init_waitqueue_head(&the_locks.write_wait);
+ init_waitqueue_head(&the_locks.read_wait);
+
+ return platform_driver_register(&msm_pcm_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_pcm_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");