diff --git a/sound/soc/msm/qdsp6v2/Makefile b/sound/soc/msm/qdsp6v2/Makefile
index 8f9e67eda54bf3f8d1d0736ca6f9e05d118d257c..8e318cb1b0eb3dceefd45b36cdcf7e45ed70e953 100644
--- a/sound/soc/msm/qdsp6v2/Makefile
+++ b/sound/soc/msm/qdsp6v2/Makefile
@@ -1,5 +1,5 @@
snd-soc-qdsp6v2-objs += msm-dai-q6-v2.o msm-pcm-q6-v2.o msm-pcm-routing-v2.o \
- msm-compress-q6-v2.o \
+ msm-compress-q6-v2.o msm-compr-q6-v2.o \
msm-pcm-lpa-v2.o \
msm-pcm-afe-v2.o msm-pcm-voip-v2.o \
msm-pcm-voice-v2.o msm-dai-q6-hdmi-v2.o \
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
new file mode 100644
index 0000000000000000000000000000000000000000..87523ab84c1b7650c03b61cec527d0cf70404584
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.c
@@ -0,0 +1,1732 @@
+/* Copyright (c) 2012-2014, 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/q6asm-v2.h>
+#include <sound/pcm_params.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/msm_audio_ion.h>
+
+#include <sound/timer.h>
+
+#include "msm-compr-q6-v2.h"
+#include "msm-pcm-routing-v2.h"
+#include "audio_ocmem.h"
+#include <sound/tlv.h>
+
+#define COMPRE_CAPTURE_NUM_PERIODS 16
+/* Allocate the worst case frame size for compressed audio */
+#define COMPRE_CAPTURE_HEADER_SIZE (sizeof(struct snd_compr_audio_info))
+/* Changing period size to 4032. 4032 will make sure COMPRE_CAPTURE_PERIOD_SIZE
+ * is 4096 with meta data size of 64 and MAX_NUM_FRAMES_PER_BUFFER 1
+ */
+#define COMPRE_CAPTURE_MAX_FRAME_SIZE (4032)
+#define COMPRE_CAPTURE_PERIOD_SIZE ((COMPRE_CAPTURE_MAX_FRAME_SIZE + \
+ COMPRE_CAPTURE_HEADER_SIZE) * \
+ MAX_NUM_FRAMES_PER_BUFFER)
+#define COMPRE_OUTPUT_METADATA_SIZE (sizeof(struct output_meta_data_st))
+#define COMPRESSED_LR_VOL_MAX_STEPS 0x20002000
+
+#define MAX_AC3_PARAM_SIZE (18*2*sizeof(int))
+#define AMR_WB_BAND_MODE 8
+#define AMR_WB_DTX_MODE 0
+
+
+const DECLARE_TLV_DB_LINEAR(compr_rx_vol_gain, 0,
+ COMPRESSED_LR_VOL_MAX_STEPS);
+struct snd_msm {
+ atomic_t audio_ocmem_req;
+};
+static struct snd_msm compressed_audio;
+
+static struct audio_locks the_locks;
+
+static struct snd_pcm_hardware msm_compr_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,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max =
+ COMPRE_CAPTURE_PERIOD_SIZE * COMPRE_CAPTURE_NUM_PERIODS ,
+ .period_bytes_min = COMPRE_CAPTURE_PERIOD_SIZE,
+ .period_bytes_max = COMPRE_CAPTURE_PERIOD_SIZE,
+ .periods_min = COMPRE_CAPTURE_NUM_PERIODS,
+ .periods_max = COMPRE_CAPTURE_NUM_PERIODS,
+ .fifo_size = 0,
+};
+
+static struct snd_pcm_hardware msm_compr_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,
+ .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 1,
+ .channels_max = 8,
+ .buffer_bytes_max = 1024 * 1024,
+ .period_bytes_min = 128 * 1024,
+ .period_bytes_max = 256 * 1024,
+ .periods_min = 4,
+ .periods_max = 8,
+ .fifo_size = 0,
+};
+
+/* Conventional and unconventional sample rate supported */
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000
+};
+
+/* Add supported codecs for compress capture path */
+static uint32_t supported_compr_capture_codecs[] = {
+ SND_AUDIOCODEC_AMRWB
+};
+
+static struct snd_pcm_hw_constraint_list constraints_sample_rates = {
+ .count = ARRAY_SIZE(supported_sample_rates),
+ .list = supported_sample_rates,
+ .mask = 0,
+};
+
+static bool msm_compr_capture_codecs(uint32_t req_codec)
+{
+ int i;
+ pr_debug("%s req_codec:%d\n", __func__, req_codec);
+ if (req_codec == 0)
+ return false;
+ for (i = 0; i < ARRAY_SIZE(supported_compr_capture_codecs); i++) {
+ if (req_codec == supported_compr_capture_codecs[i])
+ return true;
+ }
+ return false;
+}
+
+static void compr_event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct compr_audio *compr = priv;
+ struct msm_audio *prtd = &compr->prtd;
+ struct snd_pcm_substream *substream = prtd->substream;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct audio_aio_write_param param;
+ struct audio_aio_read_param read_param;
+ struct audio_buffer *buf = NULL;
+ phys_addr_t temp;
+ struct output_meta_data_st output_meta_data;
+ uint32_t *ptrmem = (uint32_t *)payload;
+ int i = 0;
+ int time_stamp_flag = 0;
+ int buffer_length = 0;
+ int stop_playback = 0;
+
+ pr_debug("%s opcode =%08x\n", __func__, opcode);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2: {
+ uint32_t *ptrmem = (uint32_t *)¶m;
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE\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);
+ else
+ if (substream->timer_running)
+ snd_timer_interrupt(substream->timer, 1);
+ atomic_inc(&prtd->out_count);
+ wake_up(&the_locks.write_wait);
+ if (!atomic_read(&prtd->start)) {
+ atomic_set(&prtd->pending_buffer, 1);
+ break;
+ } else
+ atomic_set(&prtd->pending_buffer, 0);
+
+ /*
+ * check for underrun
+ */
+ snd_pcm_stream_lock_irq(substream);
+ if (runtime->status->hw_ptr >= runtime->control->appl_ptr) {
+ runtime->render_flag |= SNDRV_RENDER_STOPPED;
+ stop_playback = 1;
+ }
+ snd_pcm_stream_unlock_irq(substream);
+
+ if (stop_playback) {
+ pr_err("underrun! render stopped\n");
+ break;
+ }
+
+ buf = prtd->audio_client->port[IN].buf;
+ pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
+ __func__, prtd->pcm_count, prtd->out_head);
+ temp = buf[0].phys + (prtd->out_head * prtd->pcm_count);
+ pr_debug("%s:writing buffer[%d] from 0x%pK\n",
+ __func__, prtd->out_head, &temp);
+
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ time_stamp_flag = SET_TIMESTAMP;
+ else
+ time_stamp_flag = NO_TIMESTAMP;
+ memcpy(&output_meta_data, (char *)(buf->data +
+ prtd->out_head * prtd->pcm_count),
+ COMPRE_OUTPUT_METADATA_SIZE);
+
+ buffer_length = output_meta_data.frame_size;
+ pr_debug("meta_data_length: %d, frame_length: %d\n",
+ output_meta_data.meta_data_length,
+ output_meta_data.frame_size);
+ pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
+ output_meta_data.timestamp_msw,
+ output_meta_data.timestamp_lsw);
+ if (buffer_length == 0) {
+ pr_debug("Recieved a zero length buffer-break out");
+ break;
+ }
+ param.paddr = temp + output_meta_data.meta_data_length;
+ param.len = buffer_length;
+ param.msw_ts = output_meta_data.timestamp_msw;
+ param.lsw_ts = output_meta_data.timestamp_lsw;
+ param.flags = time_stamp_flag;
+ param.uid = prtd->session_id;
+ for (i = 0; i < sizeof(struct audio_aio_write_param)/4;
+ i++, ++ptrmem)
+ pr_debug("cmd[%d]=0x%08x\n", i, *ptrmem);
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1) & (runtime->periods - 1);
+ break;
+ }
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ pr_debug("ASM_DATA_CMDRSP_EOS\n");
+ if (atomic_read(&prtd->eos)) {
+ pr_debug("ASM_DATA_CMDRSP_EOS wake up\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.eos_wait);
+ atomic_set(&prtd->eos, 0);
+ }
+ break;
+ case ASM_DATA_EVENT_READ_DONE_V2: {
+ pr_debug("ASM_DATA_EVENT_READ_DONE\n");
+ pr_debug("buf = %pK, data = 0x%X, *data = %pK,\n"
+ "prtd->pcm_irq_pos = %d\n",
+ prtd->audio_client->port[OUT].buf,
+ *(uint32_t *)prtd->audio_client->port[OUT].buf->data,
+ prtd->audio_client->port[OUT].buf->data,
+ prtd->pcm_irq_pos);
+
+ memcpy(prtd->audio_client->port[OUT].buf->data +
+ prtd->pcm_irq_pos, (ptrmem + READDONE_IDX_SIZE),
+ COMPRE_CAPTURE_HEADER_SIZE);
+ pr_debug("buf = %pK, updated data = 0x%X, *data = %pK\n",
+ prtd->audio_client->port[OUT].buf,
+ *(uint32_t *)(prtd->audio_client->port[OUT].buf->data +
+ prtd->pcm_irq_pos),
+ prtd->audio_client->port[OUT].buf->data);
+ if (!atomic_read(&prtd->start))
+ break;
+ pr_debug("frame size=%d, buffer = 0x%X\n",
+ ptrmem[READDONE_IDX_SIZE],
+ ptrmem[READDONE_IDX_BUFADD_LSW]);
+ if (ptrmem[READDONE_IDX_SIZE] > COMPRE_CAPTURE_MAX_FRAME_SIZE) {
+ pr_err("Frame length exceeded the max length");
+ break;
+ }
+ buf = prtd->audio_client->port[OUT].buf;
+
+ pr_debug("pcm_irq_pos=%d, buf[0].phys = 0x%pK\n",
+ prtd->pcm_irq_pos, &buf[0].phys);
+ read_param.len = prtd->pcm_count - COMPRE_CAPTURE_HEADER_SIZE;
+ read_param.paddr = buf[0].phys +
+ prtd->pcm_irq_pos + COMPRE_CAPTURE_HEADER_SIZE;
+ prtd->pcm_irq_pos += prtd->pcm_count;
+
+ if (atomic_read(&prtd->start))
+ snd_pcm_period_elapsed(substream);
+
+ q6asm_async_read(prtd->audio_client, &read_param);
+ 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 (!atomic_read(&prtd->pending_buffer))
+ break;
+ pr_debug("%s: writing %d bytes of buffer[%d] to dsp\n",
+ __func__, prtd->pcm_count, prtd->out_head);
+ buf = prtd->audio_client->port[IN].buf;
+ pr_debug("%s: writing buffer[%d] from 0x%pK head %d count %d\n",
+ __func__, prtd->out_head, &buf[0].phys,
+ prtd->pcm_count, prtd->out_head);
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ time_stamp_flag = SET_TIMESTAMP;
+ else
+ time_stamp_flag = NO_TIMESTAMP;
+ memcpy(&output_meta_data, (char *)(buf->data +
+ prtd->out_head * prtd->pcm_count),
+ COMPRE_OUTPUT_METADATA_SIZE);
+ buffer_length = output_meta_data.frame_size;
+ pr_debug("meta_data_length: %d, frame_length: %d\n",
+ output_meta_data.meta_data_length,
+ output_meta_data.frame_size);
+ pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
+ output_meta_data.timestamp_msw,
+ output_meta_data.timestamp_lsw);
+ param.paddr = buf[prtd->out_head].phys
+ + output_meta_data.meta_data_length;
+ param.len = buffer_length;
+ param.msw_ts = output_meta_data.timestamp_msw;
+ param.lsw_ts = output_meta_data.timestamp_lsw;
+ param.flags = time_stamp_flag;
+ param.uid = prtd->session_id;
+ param.metadata_len = COMPRE_OUTPUT_METADATA_SIZE;
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1)
+ & (runtime->periods - 1);
+ atomic_set(&prtd->pending_buffer, 0);
+ }
+ break;
+ case ASM_STREAM_CMD_FLUSH:
+ pr_debug("ASM_STREAM_CMD_FLUSH\n");
+ prtd->cmd_ack = 1;
+ wake_up(&the_locks.flush_wait);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ default:
+ pr_debug("Not Supported Event opcode[0x%x]\n", opcode);
+ break;
+ }
+}
+
+static int msm_compr_send_ddp_cfg(struct audio_client *ac,
+ struct snd_dec_ddp *ddp)
+{
+ int i, rc;
+ pr_debug("%s\n", __func__);
+ for (i = 0; i < ddp->params_length/2; i++) {
+ rc = q6asm_ds1_set_endp_params(ac, ddp->params_id[i],
+ ddp->params_value[i]);
+ if (rc) {
+ pr_err("sending params_id: %d failed\n",
+ ddp->params_id[i]);
+ return rc;
+ }
+ }
+ return 0;
+}
+
+static int msm_compr_playback_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ struct snd_pcm_hw_params *params;
+ struct asm_aac_cfg aac_cfg;
+ uint16_t bits_per_sample = 16;
+ int ret;
+
+ struct asm_softpause_params softpause = {
+ .enable = SOFT_PAUSE_ENABLE,
+ .period = SOFT_PAUSE_PERIOD,
+ .step = SOFT_PAUSE_STEP,
+ .rampingcurve = SOFT_PAUSE_CURVE_LINEAR,
+ };
+ struct asm_softvolume_params softvol = {
+ .period = SOFT_VOLUME_PERIOD,
+ .step = SOFT_VOLUME_STEP,
+ .rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+ };
+
+ pr_debug("%s\n", __func__);
+
+ params = &soc_prtd->dpcm[substream->stream].hw_params;
+ if (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
+ bits_per_sample = 24;
+
+ ret = q6asm_open_write_v2(prtd->audio_client,
+ compr->codec, bits_per_sample);
+ if (ret < 0) {
+ pr_err("%s: Session out open failed\n",
+ __func__);
+ return -ENOMEM;
+ }
+ msm_pcm_routing_reg_phy_stream(
+ soc_prtd->dai_link->be_id,
+ prtd->audio_client->perf_mode,
+ prtd->session_id,
+ substream->stream);
+ /*
+ * the number of channels are required to call volume api
+ * accoridngly. So, get channels from hw params
+ */
+ if ((params_channels(params) > 0) &&
+ (params_periods(params) <= runtime->hw.channels_max))
+ prtd->channel_mode = params_channels(params);
+
+ ret = q6asm_set_softpause(prtd->audio_client, &softpause);
+ if (ret < 0)
+ pr_err("%s: Send SoftPause Param failed ret=%d\n",
+ __func__, ret);
+ ret = q6asm_set_softvolume(prtd->audio_client, &softvol);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+
+ ret = q6asm_set_io_mode(prtd->audio_client,
+ (COMPRESSED_IO | ASYNC_IO_MODE));
+ if (ret < 0) {
+ pr_err("%s: Set IO mode failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ 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;
+ prtd->out_head = 0;
+ atomic_set(&prtd->out_count, runtime->periods);
+
+ if (prtd->enabled)
+ return 0;
+
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_MP3:
+ /* No media format block for mp3 */
+ break;
+ case SND_AUDIOCODEC_AAC:
+ pr_debug("%s: SND_AUDIOCODEC_AAC\n", __func__);
+ memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg));
+ aac_cfg.aot = AAC_ENC_MODE_EAAC_P;
+ aac_cfg.format = 0x03;
+ aac_cfg.ch_cfg = runtime->channels;
+ aac_cfg.sample_rate = runtime->rate;
+ ret = q6asm_media_format_block_aac(prtd->audio_client,
+ &aac_cfg);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+ break;
+ case SND_AUDIOCODEC_AC3: {
+ struct snd_dec_ddp *ddp =
+ &compr->info.codec_param.codec.options.ddp;
+ pr_debug("%s: SND_AUDIOCODEC_AC3\n", __func__);
+ ret = msm_compr_send_ddp_cfg(prtd->audio_client, ddp);
+ if (ret < 0)
+ pr_err("%s: DDP CMD CFG failed\n", __func__);
+ break;
+ }
+ case SND_AUDIOCODEC_EAC3: {
+ struct snd_dec_ddp *ddp =
+ &compr->info.codec_param.codec.options.ddp;
+ pr_debug("%s: SND_AUDIOCODEC_EAC3\n", __func__);
+ ret = msm_compr_send_ddp_cfg(prtd->audio_client, ddp);
+ if (ret < 0)
+ pr_err("%s: DDP CMD CFG failed\n", __func__);
+ break;
+ }
+ default:
+ return -EINVAL;
+ }
+
+ prtd->enabled = 1;
+ prtd->cmd_ack = 0;
+ prtd->cmd_interrupt = 0;
+
+ return 0;
+}
+
+static int msm_compr_capture_prepare(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ struct audio_buffer *buf = prtd->audio_client->port[OUT].buf;
+ struct snd_codec *codec = &compr->info.codec_param.codec;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct audio_aio_read_param read_param;
+ uint16_t bits_per_sample = 16;
+ int ret = 0;
+ int i;
+
+ prtd->pcm_size = snd_pcm_lib_buffer_bytes(substream);
+ prtd->pcm_count = snd_pcm_lib_period_bytes(substream);
+ prtd->pcm_irq_pos = 0;
+
+ if (runtime->format == SNDRV_PCM_FORMAT_S24_LE)
+ bits_per_sample = 24;
+
+ if (!msm_compr_capture_codecs(
+ compr->info.codec_param.codec.id)) {
+ /*
+ * request codec invalid or not supported,
+ * use default compress format
+ */
+ compr->info.codec_param.codec.id =
+ SND_AUDIOCODEC_AMRWB;
+ }
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_AMRWB:
+ pr_debug("q6asm_open_read(FORMAT_AMRWB)\n");
+ ret = q6asm_open_read(prtd->audio_client,
+ FORMAT_AMRWB);
+ if (ret < 0) {
+ pr_err("%s: compressed Session out open failed\n",
+ __func__);
+ return -ENOMEM;
+ }
+ pr_debug("msm_pcm_routing_reg_phy_stream\n");
+ msm_pcm_routing_reg_phy_stream(
+ soc_prtd->dai_link->be_id,
+ prtd->audio_client->perf_mode,
+ prtd->session_id, substream->stream);
+ break;
+ default:
+ pr_debug("q6asm_open_read_compressed(COMPRESSED_META_DATA_MODE)\n");
+ /*
+ ret = q6asm_open_read_compressed(prtd->audio_client,
+ MAX_NUM_FRAMES_PER_BUFFER,
+ COMPRESSED_META_DATA_MODE);
+ */
+ ret = -EINVAL;
+ break;
+ }
+
+ if (ret < 0) {
+ pr_err("%s: compressed Session out open failed\n",
+ __func__);
+ return -ENOMEM;
+ }
+
+ ret = q6asm_set_io_mode(prtd->audio_client,
+ (COMPRESSED_IO | ASYNC_IO_MODE));
+ if (ret < 0) {
+ pr_err("%s: Set IO mode failed\n", __func__);
+ return -ENOMEM;
+ }
+
+ if (!msm_compr_capture_codecs(codec->id)) {
+ /*
+ * request codec invalid or not supported,
+ * use default compress format
+ */
+ codec->id = SND_AUDIOCODEC_AMRWB;
+ }
+ /* rate and channels are sent to audio driver */
+ prtd->samp_rate = runtime->rate;
+ prtd->channel_mode = runtime->channels;
+
+ if (prtd->enabled)
+ return ret;
+ read_param.len = prtd->pcm_count;
+
+ switch (codec->id) {
+ case SND_AUDIOCODEC_AMRWB:
+ pr_debug("SND_AUDIOCODEC_AMRWB\n");
+ ret = q6asm_enc_cfg_blk_amrwb(prtd->audio_client,
+ MAX_NUM_FRAMES_PER_BUFFER,
+ /*
+ * use fixed band mode and dtx mode
+ * band mode - 23.85 kbps
+ */
+ AMR_WB_BAND_MODE,
+ /* dtx mode - disable */
+ AMR_WB_DTX_MODE);
+ if (ret < 0)
+ pr_err("%s: CMD Format block" \
+ "failed: %d\n", __func__, ret);
+ break;
+ default:
+ pr_debug("No config for codec %d\n", codec->id);
+ }
+ pr_debug("%s: Samp_rate = %d, Channel = %d, pcm_size = %d,\n"
+ "pcm_count = %d, periods = %d\n",
+ __func__, prtd->samp_rate, prtd->channel_mode,
+ prtd->pcm_size, prtd->pcm_count, runtime->periods);
+
+ for (i = 0; i < runtime->periods; i++) {
+ read_param.uid = i;
+ switch (codec->id) {
+ case SND_AUDIOCODEC_AMRWB:
+ read_param.len = prtd->pcm_count
+ - COMPRE_CAPTURE_HEADER_SIZE;
+ read_param.paddr = buf[i].phys
+ + COMPRE_CAPTURE_HEADER_SIZE;
+ pr_debug("Push buffer [%d] to DSP, "\
+ "paddr: %pK, vaddr: %pK\n",
+ i, &read_param.paddr,
+ buf[i].data);
+ q6asm_async_read(prtd->audio_client, &read_param);
+ break;
+ default:
+ read_param.paddr = buf[i].phys;
+ /*q6asm_async_read_compressed(prtd->audio_client,
+ &read_param);*/
+ pr_debug("%s: To add support for read compressed\n",
+ __func__);
+ ret = -EINVAL;
+ break;
+ }
+ }
+ prtd->periods = runtime->periods;
+
+ prtd->enabled = 1;
+
+ return ret;
+}
+
+static int msm_compr_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+ int ret = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct snd_soc_pcm_runtime *soc_prtd = substream->private_data;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+
+ pr_debug("%s\n", __func__);
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ prtd->pcm_irq_pos = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ if (!msm_compr_capture_codecs(
+ compr->info.codec_param.codec.id)) {
+ /*
+ * request codec invalid or not supported,
+ * use default compress format
+ */
+ compr->info.codec_param.codec.id =
+ SND_AUDIOCODEC_AMRWB;
+ }
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_AMRWB:
+ break;
+ default:
+ msm_pcm_routing_reg_psthr_stream(
+ soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+ break;
+ }
+ }
+ atomic_set(&prtd->pending_buffer, 1);
+ case SNDRV_PCM_TRIGGER_RESUME:
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("%s: Trigger start\n", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ atomic_set(&prtd->start, 1);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ pr_debug("SNDRV_PCM_TRIGGER_STOP\n");
+ if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_AMRWB:
+ break;
+ default:
+ msm_pcm_routing_reg_psthr_stream(
+ soc_prtd->dai_link->be_id,
+ prtd->session_id, substream->stream);
+ break;
+ }
+ }
+ atomic_set(&prtd->start, 0);
+ runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
+ break;
+ case SNDRV_PCM_TRIGGER_SUSPEND:
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE\n");
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->start, 0);
+ runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
+ break;
+ default:
+ ret = -EINVAL;
+ break;
+ }
+
+ return ret;
+}
+
+static void populate_codec_list(struct compr_audio *compr,
+ struct snd_pcm_runtime *runtime)
+{
+ pr_debug("%s\n", __func__);
+ /* MP3 Block */
+ compr->info.compr_cap.num_codecs = 5;
+ compr->info.compr_cap.min_fragment_size = runtime->hw.period_bytes_min;
+ compr->info.compr_cap.max_fragment_size = runtime->hw.period_bytes_max;
+ compr->info.compr_cap.min_fragments = runtime->hw.periods_min;
+ compr->info.compr_cap.max_fragments = runtime->hw.periods_max;
+ compr->info.compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
+ compr->info.compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
+ compr->info.compr_cap.codecs[2] = SND_AUDIOCODEC_AC3;
+ compr->info.compr_cap.codecs[3] = SND_AUDIOCODEC_EAC3;
+ compr->info.compr_cap.codecs[4] = SND_AUDIOCODEC_AMRWB;
+ /* Add new codecs here */
+}
+
+static int msm_compr_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr;
+ struct msm_audio *prtd;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ compr = kzalloc(sizeof(struct compr_audio), GFP_KERNEL);
+ if (compr == NULL) {
+ pr_err("Failed to allocate memory for msm_audio\n");
+ return -ENOMEM;
+ }
+ prtd = &compr->prtd;
+ prtd->substream = substream;
+ runtime->render_flag = SNDRV_DMA_MODE;
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)compr_event_handler, compr);
+ if (!prtd->audio_client) {
+ pr_info("%s: Could not allocate memory\n", __func__);
+ kfree(prtd);
+ return -ENOMEM;
+ }
+
+ prtd->audio_client->perf_mode = false;
+ pr_info("%s: session ID %d\n", __func__, prtd->audio_client->session);
+
+ prtd->session_id = prtd->audio_client->session;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ runtime->hw = msm_compr_hardware_playback;
+ prtd->cmd_ack = 1;
+ } else {
+ runtime->hw = msm_compr_hardware_capture;
+ }
+
+
+ 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");
+
+ prtd->dsp_cnt = 0;
+ atomic_set(&prtd->pending_buffer, 1);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ compr->codec = FORMAT_MP3;
+ populate_codec_list(compr, runtime);
+ runtime->private_data = compr;
+ atomic_set(&prtd->eos, 0);
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+ if (!atomic_cmpxchg(&compressed_audio.audio_ocmem_req, 0, 1))
+ audio_ocmem_process_req(AUDIO, true);
+ else
+ atomic_inc(&compressed_audio.audio_ocmem_req);
+ pr_debug("%s: req: %d\n", __func__,
+ atomic_read(&compressed_audio.audio_ocmem_req));
+ }
+ return 0;
+}
+
+static int compressed_set_volume(struct msm_audio *prtd, uint32_t volume)
+{
+ int rc = 0;
+ int avg_vol = 0;
+ int lgain = (volume >> 16) & 0xFFFF;
+ int rgain = volume & 0xFFFF;
+ if (prtd && prtd->audio_client) {
+ pr_debug("%s: channels %d volume 0x%x\n", __func__,
+ prtd->channel_mode, volume);
+ if ((prtd->channel_mode == 2) &&
+ (lgain != rgain)) {
+ pr_debug("%s: call q6asm_set_lrgain\n", __func__);
+ rc = q6asm_set_lrgain(prtd->audio_client, lgain, rgain);
+ } else {
+ avg_vol = (lgain + rgain)/2;
+ pr_debug("%s: call q6asm_set_volume\n", __func__);
+ rc = q6asm_set_volume(prtd->audio_client, avg_vol);
+ }
+ if (rc < 0) {
+ pr_err("%s: Send Volume command failed rc=%d\n",
+ __func__, rc);
+ }
+ }
+ return rc;
+}
+
+static int msm_compr_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 compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ int dir = 0;
+
+ pr_debug("%s\n", __func__);
+
+ dir = IN;
+ atomic_set(&prtd->pending_buffer, 0);
+
+ if (atomic_read(&compressed_audio.audio_ocmem_req) > 1)
+ atomic_dec(&compressed_audio.audio_ocmem_req);
+ else if (atomic_cmpxchg(&compressed_audio.audio_ocmem_req, 1, 0))
+ audio_ocmem_process_req(AUDIO, false);
+
+ pr_debug("%s: req: %d\n", __func__,
+ atomic_read(&compressed_audio.audio_ocmem_req));
+ prtd->pcm_irq_pos = 0;
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ msm_pcm_routing_dereg_phy_stream(
+ soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_compr_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 compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ int dir = OUT;
+
+ pr_debug("%s\n", __func__);
+ atomic_set(&prtd->pending_buffer, 0);
+ q6asm_cmd(prtd->audio_client, CMD_CLOSE);
+ q6asm_audio_client_buf_free_contiguous(dir,
+ prtd->audio_client);
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->be_id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ q6asm_audio_client_free(prtd->audio_client);
+ kfree(prtd);
+ return 0;
+}
+
+static int msm_compr_close(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_compr_playback_close(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_compr_capture_close(substream);
+ return ret;
+}
+
+static int msm_compr_prepare(struct snd_pcm_substream *substream)
+{
+ int ret = 0;
+
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+ ret = msm_compr_playback_prepare(substream);
+ else if (substream->stream == SNDRV_PCM_STREAM_CAPTURE)
+ ret = msm_compr_capture_prepare(substream);
+ return ret;
+}
+
+static snd_pcm_uframes_t msm_compr_pointer(struct snd_pcm_substream *substream)
+{
+
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+
+ if (prtd->pcm_irq_pos >= prtd->pcm_size)
+ prtd->pcm_irq_pos = 0;
+
+ pr_debug("%s: pcm_irq_pos = %d, pcm_size = %d, sample_bits = %d,\n"
+ "frame_bits = %d\n", __func__, prtd->pcm_irq_pos,
+ prtd->pcm_size, runtime->sample_bits,
+ runtime->frame_bits);
+ return bytes_to_frames(runtime, (prtd->pcm_irq_pos));
+}
+
+static int msm_compr_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;
+ runtime->render_flag = SNDRV_NON_DMA_MODE;
+ 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_compr_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ 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;
+ /* Modifying kernel hardware params based on userspace config */
+ if (params_periods(params) > 0 &&
+ (params_periods(params) != runtime->hw.periods_max)) {
+ runtime->hw.periods_max = params_periods(params);
+ }
+ if (params_period_bytes(params) > 0 &&
+ (params_period_bytes(params) != runtime->hw.period_bytes_min)) {
+ runtime->hw.period_bytes_min = params_period_bytes(params);
+ }
+ runtime->hw.buffer_bytes_max =
+ runtime->hw.period_bytes_min * runtime->hw.periods_max;
+ pr_debug("allocate %zd buffers each of size %d\n",
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir,
+ prtd->audio_client,
+ runtime->hw.period_bytes_min,
+ runtime->hw.periods_max);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n",
+ ret);
+ return -ENOMEM;
+ }
+ buf = prtd->audio_client->port[dir].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 = runtime->hw.buffer_bytes_max;
+
+ pr_debug("%s: buf[%pK]dma_buf->area[%pK]dma_buf->addr[%pK]\n"
+ "dma_buf->bytes[%zd]\n", __func__,
+ (void *)buf, (void *)dma_buf->area,
+ &dma_buf->addr, dma_buf->bytes);
+ if (!dma_buf->area)
+ return -ENOMEM;
+
+ snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+ return 0;
+}
+
+static int msm_compr_ioctl_shared(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ int rc = 0;
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ uint64_t timestamp;
+ uint64_t temp;
+
+ switch (cmd) {
+ case SNDRV_COMPRESS_TSTAMP: {
+ struct snd_compr_tstamp *tstamp;
+ pr_debug("SNDRV_COMPRESS_TSTAMP\n");
+ tstamp = arg;
+ memset(tstamp, 0x0, sizeof(*tstamp));
+ rc = q6asm_get_session_time(prtd->audio_client, ×tamp);
+ if (rc < 0) {
+ pr_err("%s: Get Session Time return value =%lld\n",
+ __func__, timestamp);
+ return -EAGAIN;
+ }
+ temp = (timestamp * 2 * runtime->channels);
+ temp = temp * (runtime->rate/1000);
+ temp = div_u64(temp, 1000);
+ tstamp->sampling_rate = runtime->rate;
+ tstamp->timestamp = timestamp;
+ pr_debug("%s: bytes_consumed:,timestamp = %lld,\n",
+ __func__,
+ tstamp->timestamp);
+ return 0;
+ }
+ case SNDRV_COMPRESS_GET_CAPS: {
+ struct snd_compr_caps *caps;
+ caps = arg;
+ memset(caps, 0, sizeof(*caps));
+ pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
+ memcpy(caps, &compr->info.compr_cap, sizeof(*caps));
+ return 0;
+ }
+ case SNDRV_COMPRESS_SET_PARAMS:
+ pr_debug("SNDRV_COMPRESS_SET_PARAMS:\n");
+ memcpy(&compr->info.codec_param, (void *) arg,
+ sizeof(struct snd_compr_params));
+ switch (compr->info.codec_param.codec.id) {
+ case SND_AUDIOCODEC_MP3:
+ /* For MP3 we dont need any other parameter */
+ pr_debug("SND_AUDIOCODEC_MP3\n");
+ compr->codec = FORMAT_MP3;
+ break;
+ case SND_AUDIOCODEC_AAC:
+ pr_debug("SND_AUDIOCODEC_AAC\n");
+ compr->codec = FORMAT_MPEG4_AAC;
+ break;
+ case SND_AUDIOCODEC_AC3: {
+ char params_value[MAX_AC3_PARAM_SIZE];
+ int *params_value_data = (int *)params_value;
+ /* 36 is the max param length for ddp */
+ int i;
+ struct snd_dec_ddp *ddp =
+ &compr->info.codec_param.codec.options.ddp;
+ uint32_t params_length = 0;
+ memset(params_value, 0, MAX_AC3_PARAM_SIZE);
+ /* check integer overflow */
+ if (ddp->params_length > UINT_MAX/sizeof(int)) {
+ pr_err("%s: Integer overflow ddp->params_length %d\n",
+ __func__, ddp->params_length);
+ return -EINVAL;
+ }
+ params_length = ddp->params_length*sizeof(int);
+ if (params_length > MAX_AC3_PARAM_SIZE) {
+ /*MAX is 36*sizeof(int) this should not happen*/
+ pr_err("%s: params_length(%d) is greater than %zd\n",
+ __func__, params_length, MAX_AC3_PARAM_SIZE);
+ return -EINVAL;
+ }
+ pr_debug("SND_AUDIOCODEC_AC3\n");
+ compr->codec = FORMAT_AC3;
+ if (copy_from_user(params_value, (void *)ddp->params,
+ params_length))
+ pr_err("%s: copy ddp params value, size=%d\n",
+ __func__, params_length);
+ pr_debug("params_length: %d\n", ddp->params_length);
+ for (i = 0; i < params_length/sizeof(int); i++)
+ pr_debug("params_value[%d]: %x\n", i,
+ params_value_data[i]);
+ for (i = 0; i < ddp->params_length/2; i++) {
+ ddp->params_id[i] = params_value_data[2*i];
+ ddp->params_value[i] = params_value_data[2*i+1];
+ }
+ if (atomic_read(&prtd->start)) {
+ rc = msm_compr_send_ddp_cfg(prtd->audio_client,
+ ddp);
+ if (rc < 0)
+ pr_err("%s: DDP CMD CFG failed\n",
+ __func__);
+ }
+ break;
+ }
+ case SND_AUDIOCODEC_EAC3: {
+ char params_value[MAX_AC3_PARAM_SIZE];
+ int *params_value_data = (int *)params_value;
+ /* 36 is the max param length for ddp */
+ int i;
+ struct snd_dec_ddp *ddp =
+ &compr->info.codec_param.codec.options.ddp;
+ uint32_t params_length = 0;
+ memset(params_value, 0, MAX_AC3_PARAM_SIZE);
+ /* check integer overflow */
+ if (ddp->params_length > UINT_MAX/sizeof(int)) {
+ pr_err("%s: Integer overflow ddp->params_length %d\n",
+ __func__, ddp->params_length);
+ return -EINVAL;
+ }
+ if (params_length > MAX_AC3_PARAM_SIZE) {
+ /*MAX is 36*sizeof(int) this should not happen*/
+ pr_err("%s: params_length(%d) is greater than %zd\n",
+ __func__, params_length, MAX_AC3_PARAM_SIZE);
+ return -EINVAL;
+ }
+ pr_debug("SND_AUDIOCODEC_EAC3\n");
+ compr->codec = FORMAT_EAC3;
+ if (copy_from_user(params_value, (void *)ddp->params,
+ params_length))
+ pr_err("%s: copy ddp params value, size=%d\n",
+ __func__, params_length);
+ pr_debug("params_length: %d\n", ddp->params_length);
+ for (i = 0; i < ddp->params_length; i++)
+ pr_debug("params_value[%d]: %x\n", i,
+ params_value_data[i]);
+ for (i = 0; i < ddp->params_length/2; i++) {
+ ddp->params_id[i] = params_value_data[2*i];
+ ddp->params_value[i] = params_value_data[2*i+1];
+ }
+ if (atomic_read(&prtd->start)) {
+ rc = msm_compr_send_ddp_cfg(prtd->audio_client,
+ ddp);
+ if (rc < 0)
+ pr_err("%s: DDP CMD CFG failed\n",
+ __func__);
+ }
+ break;
+ }
+ default:
+ pr_debug("FORMAT_LINEAR_PCM\n");
+ compr->codec = FORMAT_LINEAR_PCM;
+ break;
+ }
+ return 0;
+ case SNDRV_PCM_IOCTL1_RESET:
+ pr_debug("SNDRV_PCM_IOCTL1_RESET\n");
+ /* Flush only when session is started during CAPTURE,
+ while PLAYBACK has no such restriction. */
+ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK ||
+ (substream->stream == SNDRV_PCM_STREAM_CAPTURE &&
+ atomic_read(&prtd->start))) {
+ if (atomic_read(&prtd->eos)) {
+ prtd->cmd_interrupt = 1;
+ wake_up(&the_locks.eos_wait);
+ atomic_set(&prtd->eos, 0);
+ }
+
+ /* A unlikely race condition possible with FLUSH
+ DRAIN if ack is set by flush and reset by drain */
+ prtd->cmd_ack = 0;
+ rc = q6asm_cmd(prtd->audio_client, CMD_FLUSH);
+ if (rc < 0) {
+ pr_err("%s: flush cmd failed rc=%d\n",
+ __func__, rc);
+ return rc;
+ }
+ rc = wait_event_timeout(the_locks.flush_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (!rc)
+ pr_err("Flush cmd timeout\n");
+ prtd->pcm_irq_pos = 0;
+ }
+ break;
+ case SNDRV_COMPRESS_DRAIN:
+ pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__);
+ if (atomic_read(&prtd->pending_buffer)) {
+ pr_debug("%s: no pending writes, drain would block\n",
+ __func__);
+ return -EWOULDBLOCK;
+ }
+
+ atomic_set(&prtd->eos, 1);
+ atomic_set(&prtd->pending_buffer, 0);
+ prtd->cmd_ack = 0;
+ q6asm_cmd_nowait(prtd->audio_client, CMD_EOS);
+ /* Wait indefinitely for DRAIN. Flush can also signal this*/
+ rc = wait_event_interruptible(the_locks.eos_wait,
+ (prtd->cmd_ack || prtd->cmd_interrupt));
+
+ if (rc < 0)
+ pr_err("EOS cmd interrupted\n");
+ pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait\n", __func__);
+
+ if (prtd->cmd_interrupt)
+ rc = -EINTR;
+
+ prtd->cmd_interrupt = 0;
+ return rc;
+ default:
+ break;
+ }
+ return snd_pcm_lib_ioctl(substream, cmd, arg);
+}
+#ifdef CONFIG_COMPAT
+struct snd_enc_wma32 {
+ u32 super_block_align; /* WMA Type-specific data */
+ u32 encodeopt1;
+ u32 encodeopt2;
+};
+
+struct snd_enc_vorbis32 {
+ s32 quality;
+ u32 managed;
+ u32 max_bit_rate;
+ u32 min_bit_rate;
+ u32 downmix;
+};
+
+struct snd_enc_real32 {
+ u32 quant_bits;
+ u32 start_region;
+ u32 num_regions;
+};
+
+struct snd_enc_flac32 {
+ u32 num;
+ u32 gain;
+};
+
+struct snd_enc_generic32 {
+ u32 bw; /* encoder bandwidth */
+ s32 reserved[15];
+};
+struct snd_dec_dts32 {
+ u32 modelIdLength;
+ compat_uptr_t modelId;
+};
+struct snd_dec_ddp32 {
+ u32 params_length;
+ compat_uptr_t params;
+ u32 params_id[18];
+ u32 params_value[18];
+};
+
+union snd_codec_options32 {
+ struct snd_enc_wma32 wma;
+ struct snd_enc_vorbis32 vorbis;
+ struct snd_enc_real32 real;
+ struct snd_enc_flac32 flac;
+ struct snd_enc_generic32 generic;
+ struct snd_dec_dts32 dts;
+ struct snd_dec_ddp32 ddp;
+};
+
+struct snd_codec32 {
+ u32 id;
+ u32 ch_in;
+ u32 ch_out;
+ u32 sample_rate;
+ u32 bit_rate;
+ u32 rate_control;
+ u32 profile;
+ u32 level;
+ u32 ch_mode;
+ u32 format;
+ u32 align;
+ u32 transcode_dts;
+ struct snd_dec_dts32 dts;
+ union snd_codec_options32 options;
+ u32 reserved[3];
+};
+
+struct snd_compressed_buffer32 {
+ u32 fragment_size;
+ u32 fragments;
+};
+
+struct snd_compr_params32 {
+ struct snd_compressed_buffer32 buffer;
+ struct snd_codec32 codec;
+ u8 no_wake_mode;
+};
+
+struct snd_compr_caps32 {
+ u32 num_codecs;
+ u32 direction;
+ u32 min_fragment_size;
+ u32 max_fragment_size;
+ u32 min_fragments;
+ u32 max_fragments;
+ u32 codecs[MAX_NUM_CODECS];
+ u32 reserved[11];
+};
+struct snd_compr_tstamp32 {
+ u32 byte_offset;
+ u32 copied_total;
+ compat_ulong_t pcm_frames;
+ compat_ulong_t pcm_io_frames;
+ u32 sampling_rate;
+ compat_u64 timestamp;
+};
+enum {
+ SNDRV_COMPRESS_TSTAMP32 = _IOR('C', 0x20, struct snd_compr_tstamp32),
+ SNDRV_COMPRESS_GET_CAPS32 = _IOWR('C', 0x10, struct snd_compr_caps32),
+ SNDRV_COMPRESS_SET_PARAMS32 =
+ _IOW('C', 0x12, struct snd_compr_params32),
+};
+static int msm_compr_compat_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ int err = 0;
+ switch (cmd) {
+ case SNDRV_COMPRESS_TSTAMP32: {
+ struct snd_compr_tstamp tstamp;
+ struct snd_compr_tstamp32 tstamp32;
+ memset(&tstamp, 0, sizeof(tstamp));
+ memset(&tstamp32, 0, sizeof(tstamp32));
+ cmd = SNDRV_COMPRESS_TSTAMP;
+ err = msm_compr_ioctl_shared(substream, cmd, &tstamp);
+ if (err) {
+ pr_err("%s: COMPRESS_TSTAMP failed rc %d\n",
+ __func__, err);
+ goto bail_out;
+ }
+ tstamp32.byte_offset = tstamp.byte_offset;
+ tstamp32.copied_total = tstamp.copied_total;
+ tstamp32.pcm_frames = tstamp.pcm_frames;
+ tstamp32.pcm_io_frames = tstamp.pcm_io_frames;
+ tstamp32.sampling_rate = tstamp.sampling_rate;
+ tstamp32.timestamp = tstamp.timestamp;
+ if (copy_to_user(arg, &tstamp32, sizeof(tstamp32))) {
+ pr_err("%s: copytouser failed COMPRESS_TSTAMP32\n",
+ __func__);
+ err = -EFAULT;
+ }
+ break;
+ }
+ case SNDRV_COMPRESS_GET_CAPS32: {
+ struct snd_compr_caps caps;
+ struct snd_compr_caps32 caps32;
+ u32 i;
+ memset(&caps, 0, sizeof(caps));
+ memset(&caps32, 0, sizeof(caps32));
+ cmd = SNDRV_COMPRESS_GET_CAPS;
+ err = msm_compr_ioctl_shared(substream, cmd, &caps);
+ if (err) {
+ pr_err("%s: GET_CAPS failed rc %d\n",
+ __func__, err);
+ goto bail_out;
+ }
+ pr_debug("SNDRV_COMPRESS_GET_CAPS_32\n");
+ if (!err && caps.num_codecs >= MAX_NUM_CODECS) {
+ pr_err("%s: Invalid number of codecs\n", __func__);
+ err = -EINVAL;
+ goto bail_out;
+ }
+ caps32.direction = caps.direction;
+ caps32.max_fragment_size = caps.max_fragment_size;
+ caps32.max_fragments = caps.max_fragments;
+ caps32.min_fragment_size = caps.min_fragment_size;
+ caps32.num_codecs = caps.num_codecs;
+ for (i = 0; i < caps.num_codecs; i++)
+ caps32.codecs[i] = caps.codecs[i];
+ if (copy_to_user(arg, &caps32, sizeof(caps32))) {
+ pr_err("%s: copytouser failed COMPRESS_GETCAPS32\n",
+ __func__);
+ err = -EFAULT;
+ }
+ break;
+ }
+ case SNDRV_COMPRESS_SET_PARAMS32: {
+ struct snd_compr_params32 params32;
+ struct snd_compr_params params;
+ memset(¶ms32, 0 , sizeof(params32));
+ memset(¶ms, 0 , sizeof(params));
+ cmd = SNDRV_COMPRESS_SET_PARAMS;
+ if (copy_from_user(¶ms32, arg, sizeof(params32))) {
+ pr_err("%s: copyfromuser failed SET_PARAMS32\n",
+ __func__);
+ err = -EFAULT;
+ goto bail_out;
+ }
+ params.no_wake_mode = params32.no_wake_mode;
+ params.codec.id = params32.codec.id;
+ params.codec.ch_in = params32.codec.ch_in;
+ params.codec.ch_out = params32.codec.ch_out;
+ params.codec.sample_rate = params32.codec.sample_rate;
+ params.codec.bit_rate = params32.codec.bit_rate;
+ params.codec.rate_control = params32.codec.rate_control;
+ params.codec.profile = params32.codec.profile;
+ params.codec.level = params32.codec.level;
+ params.codec.ch_mode = params32.codec.ch_mode;
+ params.codec.format = params32.codec.format;
+ params.codec.align = params32.codec.align;
+ params.codec.transcode_dts = params32.codec.transcode_dts;
+
+ switch (params.codec.id) {
+ case SND_AUDIOCODEC_WMA:
+ case SND_AUDIOCODEC_WMA_PRO:
+ params.codec.options.wma.encodeopt1 =
+ params32.codec.options.wma.encodeopt1;
+ params.codec.options.wma.encodeopt2 =
+ params32.codec.options.wma.encodeopt2;
+ params.codec.options.wma.super_block_align =
+ params32.codec.options.wma.super_block_align;
+ break;
+ case SND_AUDIOCODEC_VORBIS:
+ params.codec.options.vorbis.downmix =
+ params32.codec.options.vorbis.downmix;
+ params.codec.options.vorbis.managed =
+ params32.codec.options.vorbis.managed;
+ params.codec.options.vorbis.max_bit_rate =
+ params32.codec.options.vorbis.max_bit_rate;
+ params.codec.options.vorbis.min_bit_rate =
+ params32.codec.options.vorbis.min_bit_rate;
+ params.codec.options.vorbis.quality =
+ params32.codec.options.vorbis.quality;
+ break;
+ case SND_AUDIOCODEC_REAL:
+ params.codec.options.real.num_regions =
+ params32.codec.options.real.num_regions;
+ params.codec.options.real.quant_bits =
+ params32.codec.options.real.quant_bits;
+ params.codec.options.real.start_region =
+ params32.codec.options.real.start_region;
+ break;
+ case SND_AUDIOCODEC_FLAC:
+ params.codec.options.flac.gain =
+ params32.codec.options.flac.gain;
+ params.codec.options.flac.num =
+ params32.codec.options.flac.num;
+ break;
+ case SND_AUDIOCODEC_DTS:
+ case SND_AUDIOCODEC_DTS_PASS_THROUGH:
+ case SND_AUDIOCODEC_DTS_LBR:
+ case SND_AUDIOCODEC_DTS_LBR_PASS_THROUGH:
+ case SND_AUDIOCODEC_DTS_TRANSCODE_LOOPBACK:
+ params.codec.options.dts.modelIdLength =
+ params32.codec.options.dts.modelIdLength;
+ params.codec.options.dts.modelId =
+ compat_ptr(params32.codec.options.dts.modelId);
+ break;
+ case SND_AUDIOCODEC_AC3:
+ case SND_AUDIOCODEC_EAC3:
+ params.codec.options.ddp.params_length =
+ params32.codec.options.ddp.params_length;
+ params.codec.options.ddp.params =
+ compat_ptr(params32.codec.options.ddp.params);
+ memcpy(params.codec.options.ddp.params_value,
+ params32.codec.options.ddp.params_value,
+ sizeof(params32.codec.options.ddp.params_value));
+ memcpy(params.codec.options.ddp.params_id,
+ params32.codec.options.ddp.params_id,
+ sizeof(params32.codec.options.ddp.params_id));
+ break;
+ default:
+ params.codec.options.generic.bw =
+ params32.codec.options.generic.bw;
+ break;
+ }
+ if (!err)
+ err = msm_compr_ioctl_shared(substream, cmd, ¶ms);
+ break;
+ }
+ default:
+ err = msm_compr_ioctl_shared(substream, cmd, arg);
+ }
+bail_out:
+ return err;
+
+}
+#endif
+static int msm_compr_ioctl(struct snd_pcm_substream *substream,
+ unsigned int cmd, void *arg)
+{
+ int err = 0;
+ if (!substream) {
+ pr_err("%s: Invalid params\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("%s called with cmd = %d\n", __func__, cmd);
+ switch (cmd) {
+ case SNDRV_COMPRESS_TSTAMP: {
+ struct snd_compr_tstamp tstamp;
+ if (!arg) {
+ pr_err("%s: Invalid params Tstamp\n", __func__);
+ return -EINVAL;
+ }
+ err = msm_compr_ioctl_shared(substream, cmd, &tstamp);
+ if (err)
+ pr_err("%s: COMPRESS_TSTAMP failed rc %d\n",
+ __func__, err);
+ if (!err && copy_to_user(arg, &tstamp, sizeof(tstamp))) {
+ pr_err("%s: copytouser failed COMPRESS_TSTAMP\n",
+ __func__);
+ err = -EFAULT;
+ }
+ break;
+ }
+ case SNDRV_COMPRESS_GET_CAPS: {
+ struct snd_compr_caps cap;
+ if (!arg) {
+ pr_err("%s: Invalid params getcaps\n", __func__);
+ return -EINVAL;
+ }
+ pr_debug("SNDRV_COMPRESS_GET_CAPS\n");
+ err = msm_compr_ioctl_shared(substream, cmd, &cap);
+ if (err)
+ pr_err("%s: GET_CAPS failed rc %d\n",
+ __func__, err);
+ if (!err && copy_to_user(arg, &cap, sizeof(cap))) {
+ pr_err("%s: copytouser failed GET_CAPS\n",
+ __func__);
+ err = -EFAULT;
+ }
+ break;
+ }
+ case SNDRV_COMPRESS_SET_PARAMS: {
+ struct snd_compr_params params;
+ if (!arg) {
+ pr_err("%s: Invalid params setparam\n", __func__);
+ return -EINVAL;
+ }
+ if (copy_from_user(¶ms, arg,
+ sizeof(struct snd_compr_params))) {
+ pr_err("%s: SET_PARAMS\n", __func__);
+ return -EFAULT;
+ }
+ err = msm_compr_ioctl_shared(substream, cmd, ¶ms);
+ if (err)
+ pr_err("%s: SET_PARAMS failed rc %d\n",
+ __func__, err);
+ break;
+ }
+ default:
+ err = msm_compr_ioctl_shared(substream, cmd, arg);
+ }
+ return err;
+}
+
+static int msm_compr_restart(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+ struct compr_audio *compr = runtime->private_data;
+ struct msm_audio *prtd = &compr->prtd;
+ struct audio_aio_write_param param;
+ struct audio_buffer *buf = NULL;
+ struct output_meta_data_st output_meta_data;
+ int time_stamp_flag = 0;
+ int buffer_length = 0;
+
+ pr_debug("%s, trigger restart\n", __func__);
+
+ if (runtime->render_flag & SNDRV_RENDER_STOPPED) {
+ buf = prtd->audio_client->port[IN].buf;
+ pr_debug("%s:writing %d bytes of buffer[%d] to dsp 2\n",
+ __func__, prtd->pcm_count, prtd->out_head);
+ pr_debug("%s:writing buffer[%d] from 0x%08x\n",
+ __func__, prtd->out_head,
+ ((unsigned int)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count)));
+
+ if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_ENABLE)
+ time_stamp_flag = SET_TIMESTAMP;
+ else
+ time_stamp_flag = NO_TIMESTAMP;
+ memcpy(&output_meta_data, (char *)(buf->data +
+ prtd->out_head * prtd->pcm_count),
+ COMPRE_OUTPUT_METADATA_SIZE);
+
+ buffer_length = output_meta_data.frame_size;
+ pr_debug("meta_data_length: %d, frame_length: %d\n",
+ output_meta_data.meta_data_length,
+ output_meta_data.frame_size);
+ pr_debug("timestamp_msw: %d, timestamp_lsw: %d\n",
+ output_meta_data.timestamp_msw,
+ output_meta_data.timestamp_lsw);
+
+ param.paddr = (unsigned long)buf[0].phys
+ + (prtd->out_head * prtd->pcm_count)
+ + output_meta_data.meta_data_length;
+ param.len = buffer_length;
+ param.msw_ts = output_meta_data.timestamp_msw;
+ param.lsw_ts = output_meta_data.timestamp_lsw;
+ param.flags = time_stamp_flag;
+ param.uid = prtd->session_id;
+ if (q6asm_async_write(prtd->audio_client,
+ ¶m) < 0)
+ pr_err("%s:q6asm_async_write failed\n",
+ __func__);
+ else
+ prtd->out_head =
+ (prtd->out_head + 1) & (runtime->periods - 1);
+
+ runtime->render_flag &= ~SNDRV_RENDER_STOPPED;
+ return 0;
+ }
+ return 0;
+}
+
+static int msm_compr_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 : %x\n", __func__, volume);
+ if (!substream)
+ return -ENODEV;
+ if (!substream->runtime)
+ return 0;
+ prtd = substream->runtime->private_data;
+ if (prtd)
+ rc = compressed_set_volume(prtd, volume);
+
+ return rc;
+}
+
+static int msm_compr_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)
+ return -ENODEV;
+ if (!substream->runtime)
+ return 0;
+ prtd = substream->runtime->private_data;
+ if (prtd)
+ ucontrol->value.integer.value[0] = prtd->volume;
+ return 0;
+}
+
+static int msm_compr_add_controls(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 cntrl add\n", __func__);
+ ret = snd_pcm_add_volume_ctls(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ NULL, 1, rtd->dai_link->be_id,
+ &volume_info);
+ if (ret < 0)
+ return ret;
+ kctl = volume_info->kctl;
+ kctl->put = msm_compr_volume_ctl_put;
+ kctl->get = msm_compr_volume_ctl_get;
+ kctl->tlv.p = compr_rx_vol_gain;
+ return 0;
+}
+
+static struct snd_pcm_ops msm_compr_ops = {
+ .open = msm_compr_open,
+ .hw_params = msm_compr_hw_params,
+ .close = msm_compr_close,
+ .ioctl = msm_compr_ioctl,
+ .prepare = msm_compr_prepare,
+ .trigger = msm_compr_trigger,
+ .pointer = msm_compr_pointer,
+ .mmap = msm_compr_mmap,
+ .restart = msm_compr_restart,
+#ifdef CONFIG_COMPAT
+ .compat_ioctl = msm_compr_compat_ioctl,
+#endif
+};
+
+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_compr_add_controls(rtd);
+ if (ret)
+ pr_err("%s, kctl add failed\n", __func__);
+ return ret;
+}
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .ops = &msm_compr_ops,
+ .pcm_new = msm_asoc_pcm_new,
+};
+
+static int msm_compr_probe(struct platform_device *pdev)
+{
+ if (pdev->dev.of_node)
+ dev_set_name(&pdev->dev, "%s", "msm-compr-dsp");
+
+ dev_info(&pdev->dev, "%s: dev name %s\n",
+ __func__, dev_name(&pdev->dev));
+
+ atomic_set(&compressed_audio.audio_ocmem_req, 0);
+ return snd_soc_register_platform(&pdev->dev,
+ &msm_soc_platform);
+}
+
+static int msm_compr_remove(struct platform_device *pdev)
+{
+ snd_soc_unregister_platform(&pdev->dev);
+ return 0;
+}
+
+static const struct of_device_id msm_compr_dt_match[] = {
+ {.compatible = "qcom,msm-compr-dsp"},
+ {}
+};
+MODULE_DEVICE_TABLE(of, msm_compr_dt_match);
+
+static struct platform_driver msm_compr_driver = {
+ .driver = {
+ .name = "msm-compr-dsp",
+ .owner = THIS_MODULE,
+ .of_match_table = msm_compr_dt_match,
+ },
+ .probe = msm_compr_probe,
+ .remove = msm_compr_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);
+ init_waitqueue_head(&the_locks.flush_wait);
+
+ return platform_driver_register(&msm_compr_driver);
+}
+module_init(msm_soc_platform_init);
+
+static void __exit msm_soc_platform_exit(void)
+{
+ platform_driver_unregister(&msm_compr_driver);
+}
+module_exit(msm_soc_platform_exit);
+
+MODULE_DESCRIPTION("PCM module platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
new file mode 100644
index 0000000000000000000000000000000000000000..d6e3ec6956b1b7ca97d17f30ff0ddfb1815420b8
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-compr-q6-v2.h
@@ -0,0 +1,36 @@
+/*
+ * Copyright (c) 2012, 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.
+ */
+
+#ifndef _MSM_COMPR_H
+#define _MSM_COMPR_H
+#include <sound/apr_audio-v2.h>
+#include <sound/q6asm-v2.h>
+#include <sound/compress_params.h>
+#include <sound/compress_offload.h>
+#include <sound/compress_driver.h>
+
+#include "msm-pcm-q6-v2.h"
+
+struct compr_info {
+ struct snd_compr_caps compr_cap;
+ struct snd_compr_codec_caps codec_caps;
+ struct snd_compr_params codec_param;
+};
+
+struct compr_audio {
+ struct msm_audio prtd;
+ struct compr_info info;
+ uint32_t codec;
+};
+
+#endif /*_MSM_COMPR_H*/