audio-lnx: Initial change for techpack of audio drivers.
Add snapshot for audio drivers for SDM targets. The code is
migrated from msm-4.9 kernel at the below cutoff -
(74ff856e8d6: "net: ipc_router: Add dynamic enable/disable
wakeup source feature")
This changes are done for new techpack addition
for audio kernel. Migrate all audio kernel drivers
to this techpack.
Change-Id: I33d580af3ba86a5cb777583efc5d4cdaf2882d93
Signed-off-by: Asish Bhattacharya <asishb@codeaurora.org>
diff --git a/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
new file mode 100644
index 0000000..c885265
--- /dev/null
+++ b/sound/soc/msm/qdsp6v2/msm-compress-q6-v2.c
@@ -0,0 +1,4541 @@
+/* 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/math64.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 <sound/audio_effects.h>
+#include <asm/dma.h>
+#include <linux/dma-mapping.h>
+#include <linux/msm_audio_ion.h>
+#include <linux/msm_audio.h>
+
+#include <sound/timer.h>
+#include <sound/tlv.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 <sound/msm-audio-effects-q6-v2.h>
+#include "msm-pcm-routing-v2.h"
+#include "msm-qti-pp-config.h"
+
+#define DSP_PP_BUFFERING_IN_MSEC 25
+#define PARTIAL_DRAIN_ACK_EARLY_BY_MSEC 150
+#define MP3_OUTPUT_FRAME_SZ 1152
+#define AAC_OUTPUT_FRAME_SZ 1024
+#define AC3_OUTPUT_FRAME_SZ 1536
+#define EAC3_OUTPUT_FRAME_SZ 1536
+#define DSP_NUM_OUTPUT_FRAME_BUFFERED 2
+#define FLAC_BLK_SIZE_LIMIT 65535
+
+/* Timestamp mode payload offsets */
+#define CAPTURE_META_DATA_TS_OFFSET_LSW 6
+#define CAPTURE_META_DATA_TS_OFFSET_MSW 7
+
+/* decoder parameter length */
+#define DDP_DEC_MAX_NUM_PARAM 18
+
+/* Default values used if user space does not set */
+#define COMPR_PLAYBACK_MIN_FRAGMENT_SIZE (8 * 1024)
+#define COMPR_PLAYBACK_MAX_FRAGMENT_SIZE (128 * 1024)
+#define COMPR_PLAYBACK_MIN_NUM_FRAGMENTS (4)
+#define COMPR_PLAYBACK_MAX_NUM_FRAGMENTS (16 * 4)
+
+#define COMPRESSED_LR_VOL_MAX_STEPS 0x2000
+const DECLARE_TLV_DB_LINEAR(msm_compr_vol_gain, 0,
+ COMPRESSED_LR_VOL_MAX_STEPS);
+
+/* Stream id switches between 1 and 2 */
+#define NEXT_STREAM_ID(stream_id) ((stream_id & 1) + 1)
+
+#define STREAM_ARRAY_INDEX(stream_id) (stream_id - 1)
+
+#define MAX_NUMBER_OF_STREAMS 2
+
+struct msm_compr_gapless_state {
+ bool set_next_stream_id;
+ int32_t stream_opened[MAX_NUMBER_OF_STREAMS];
+ uint32_t initial_samples_drop;
+ uint32_t trailing_samples_drop;
+ uint32_t gapless_transition;
+ bool use_dsp_gapless_mode;
+ union snd_codec_options codec_options;
+};
+
+static unsigned int supported_sample_rates[] = {
+ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000, 64000,
+ 88200, 96000, 128000, 144000, 176400, 192000, 352800, 384000, 2822400,
+ 5644800
+};
+
+struct msm_compr_pdata {
+ struct snd_compr_stream *cstream[MSM_FRONTEND_DAI_MAX];
+ uint32_t volume[MSM_FRONTEND_DAI_MAX][2]; /* For both L & R */
+ struct msm_compr_audio_effects *audio_effects[MSM_FRONTEND_DAI_MAX];
+ bool use_dsp_gapless_mode;
+ bool use_legacy_api; /* indicates use older asm apis*/
+ struct msm_compr_dec_params *dec_params[MSM_FRONTEND_DAI_MAX];
+ struct msm_compr_ch_map *ch_map[MSM_FRONTEND_DAI_MAX];
+};
+
+struct msm_compr_audio {
+ struct snd_compr_stream *cstream;
+ struct snd_compr_caps compr_cap;
+ struct snd_compr_codec_caps codec_caps;
+ struct snd_compr_params codec_param;
+ struct audio_client *audio_client;
+
+ uint32_t codec;
+ uint32_t compr_passthr;
+ void *buffer; /* virtual address */
+ phys_addr_t buffer_paddr; /* physical address */
+ uint32_t app_pointer;
+ uint32_t buffer_size;
+ uint32_t byte_offset;
+ uint64_t copied_total; /* bytes consumed by DSP */
+ uint64_t bytes_received; /* from userspace */
+ uint64_t bytes_sent; /* to DSP */
+
+ uint64_t received_total; /* bytes received from DSP */
+ uint64_t bytes_copied; /* to userspace */
+ uint64_t bytes_read; /* from DSP */
+ uint32_t bytes_read_offset; /* bytes read offset */
+
+ uint32_t ts_header_offset; /* holds the timestamp header offset */
+
+ int32_t first_buffer;
+ int32_t last_buffer;
+ int32_t partial_drain_delay;
+
+ uint16_t session_id;
+
+ uint32_t sample_rate;
+ uint32_t num_channels;
+
+ /*
+ * convention - commands coming from the same thread
+ * can use the common cmd_ack var. Others (e.g drain/EOS)
+ * must use separate vars to track command status.
+ */
+ uint32_t cmd_ack;
+ uint32_t cmd_interrupt;
+ uint32_t drain_ready;
+ uint32_t eos_ack;
+
+ uint32_t stream_available;
+ uint32_t next_stream;
+
+ uint32_t run_mode;
+ uint32_t start_delay_lsw;
+ uint32_t start_delay_msw;
+
+ uint64_t marker_timestamp;
+
+ struct msm_compr_gapless_state gapless_state;
+
+ atomic_t start;
+ atomic_t eos;
+ atomic_t drain;
+ atomic_t xrun;
+ atomic_t close;
+ atomic_t wait_on_close;
+ atomic_t error;
+
+ wait_queue_head_t eos_wait;
+ wait_queue_head_t drain_wait;
+ wait_queue_head_t close_wait;
+ wait_queue_head_t wait_for_stream_avail;
+
+ spinlock_t lock;
+};
+
+const u32 compr_codecs[] = {
+ SND_AUDIOCODEC_AC3, SND_AUDIOCODEC_EAC3, SND_AUDIOCODEC_DTS,
+ SND_AUDIOCODEC_DSD, SND_AUDIOCODEC_TRUEHD, SND_AUDIOCODEC_IEC61937};
+
+struct query_audio_effect {
+ uint32_t mod_id;
+ uint32_t parm_id;
+ uint32_t size;
+ uint32_t offset;
+ uint32_t device;
+};
+
+struct msm_compr_audio_effects {
+ struct bass_boost_params bass_boost;
+ struct pbe_params pbe;
+ struct virtualizer_params virtualizer;
+ struct reverb_params reverb;
+ struct eq_params equalizer;
+ struct soft_volume_params volume;
+ struct query_audio_effect query;
+};
+
+struct msm_compr_dec_params {
+ struct snd_dec_ddp ddp_params;
+};
+
+struct msm_compr_ch_map {
+ bool set_ch_map;
+ char channel_map[PCM_FORMAT_MAX_NUM_CHANNEL];
+};
+
+static int msm_compr_send_dec_params(struct snd_compr_stream *cstream,
+ struct msm_compr_dec_params *dec_params,
+ int stream_id);
+
+static int msm_compr_set_render_mode(struct msm_compr_audio *prtd,
+ uint32_t render_mode) {
+ int ret = -EINVAL;
+ struct audio_client *ac = prtd->audio_client;
+
+ pr_debug("%s, got render mode %u\n", __func__, render_mode);
+
+ if (render_mode == SNDRV_COMPRESS_RENDER_MODE_AUDIO_MASTER) {
+ render_mode = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_DEFAULT;
+ } else if (render_mode == SNDRV_COMPRESS_RENDER_MODE_STC_MASTER) {
+ render_mode = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_LOCAL_STC;
+ prtd->run_mode = ASM_SESSION_CMD_RUN_STARTIME_RUN_WITH_DELAY;
+ } else {
+ pr_err("%s, Invalid render mode %u\n", __func__,
+ render_mode);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ ret = q6asm_send_mtmx_strtr_render_mode(ac, render_mode);
+ if (ret) {
+ pr_err("%s, Render mode can't be set error %d\n", __func__,
+ ret);
+ }
+exit:
+ return ret;
+}
+
+static int msm_compr_set_clk_rec_mode(struct audio_client *ac,
+ uint32_t clk_rec_mode) {
+ int ret = -EINVAL;
+
+ pr_debug("%s, got clk rec mode %u\n", __func__, clk_rec_mode);
+
+ if (clk_rec_mode == SNDRV_COMPRESS_CLK_REC_MODE_NONE) {
+ clk_rec_mode = ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_NONE;
+ } else if (clk_rec_mode == SNDRV_COMPRESS_CLK_REC_MODE_AUTO) {
+ clk_rec_mode = ASM_SESSION_MTMX_STRTR_PARAM_CLK_REC_AUTO;
+ } else {
+ pr_err("%s, Invalid clk rec_mode mode %u\n", __func__,
+ clk_rec_mode);
+ ret = -EINVAL;
+ goto exit;
+ }
+
+ ret = q6asm_send_mtmx_strtr_clk_rec_mode(ac, clk_rec_mode);
+ if (ret) {
+ pr_err("%s, clk rec mode can't be set, error %d\n", __func__,
+ ret);
+ }
+
+exit:
+ return ret;
+}
+
+static int msm_compr_set_render_window(struct audio_client *ac,
+ uint32_t ws_lsw, uint32_t ws_msw,
+ uint32_t we_lsw, uint32_t we_msw)
+{
+ int ret = -EINVAL;
+ struct asm_session_mtmx_strtr_param_window_v2_t asm_mtmx_strtr_window;
+ uint32_t param_id;
+
+ pr_debug("%s, ws_lsw 0x%x ws_msw 0x%x we_lsw 0x%x we_ms 0x%x\n",
+ __func__, ws_lsw, ws_msw, we_lsw, we_msw);
+
+ memset(&asm_mtmx_strtr_window, 0,
+ sizeof(struct asm_session_mtmx_strtr_param_window_v2_t));
+ asm_mtmx_strtr_window.window_lsw = ws_lsw;
+ asm_mtmx_strtr_window.window_msw = ws_msw;
+ param_id = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_START_V2;
+ ret = q6asm_send_mtmx_strtr_window(ac, &asm_mtmx_strtr_window,
+ param_id);
+ if (ret) {
+ pr_err("%s, start window can't be set error %d\n", __func__,
+ ret);
+ goto exit;
+ }
+
+ asm_mtmx_strtr_window.window_lsw = we_lsw;
+ asm_mtmx_strtr_window.window_msw = we_msw;
+ param_id = ASM_SESSION_MTMX_STRTR_PARAM_RENDER_WINDOW_END_V2;
+ ret = q6asm_send_mtmx_strtr_window(ac, &asm_mtmx_strtr_window,
+ param_id);
+ if (ret) {
+ pr_err("%s, end window can't be set error %d\n", __func__,
+ ret);
+ }
+
+exit:
+ return ret;
+}
+
+static int msm_compr_enable_adjust_session_clock(struct audio_client *ac,
+ bool enable)
+{
+ int ret;
+
+ pr_debug("%s, enable adjust_session %d\n", __func__, enable);
+
+ ret = q6asm_send_mtmx_strtr_enable_adjust_session_clock(ac, enable);
+ if (ret)
+ pr_err("%s, adjust session clock can't be set error %d\n",
+ __func__, ret);
+
+ return ret;
+}
+
+static int msm_compr_adjust_session_clock(struct audio_client *ac,
+ uint32_t adjust_session_lsw, uint32_t adjust_session_msw)
+{
+ int ret;
+
+ pr_debug("%s, adjust_session_time_msw 0x%x adjust_session_time_lsw 0x%x\n",
+ __func__, adjust_session_msw, adjust_session_lsw);
+
+ ret = q6asm_adjust_session_clock(ac,
+ adjust_session_lsw,
+ adjust_session_msw);
+ if (ret)
+ pr_err("%s, adjust session clock can't be set error %d\n",
+ __func__, ret);
+
+ return ret;
+}
+
+static int msm_compr_set_volume(struct snd_compr_stream *cstream,
+ uint32_t volume_l, uint32_t volume_r)
+{
+ struct msm_compr_audio *prtd;
+ int rc = 0;
+ uint32_t avg_vol, gain_list[VOLUME_CONTROL_MAX_CHANNELS];
+ uint32_t num_channels;
+ struct snd_soc_pcm_runtime *rtd;
+ struct msm_compr_pdata *pdata;
+ bool use_default = true;
+ u8 *chmap = NULL;
+
+ pr_debug("%s: volume_l %d volume_r %d\n",
+ __func__, volume_l, volume_r);
+ if (!cstream || !cstream->runtime) {
+ pr_err("%s: session not active\n", __func__);
+ return -EPERM;
+ }
+ rtd = cstream->private_data;
+ prtd = cstream->runtime->private_data;
+
+ if (!rtd || !rtd->platform || !prtd || !prtd->audio_client) {
+ pr_err("%s: invalid rtd, prtd or audio client", __func__);
+ return rc;
+ }
+ pdata = snd_soc_platform_get_drvdata(rtd->platform);
+
+ if (prtd->compr_passthr != LEGACY_PCM) {
+ pr_debug("%s: No volume config for passthrough %d\n",
+ __func__, prtd->compr_passthr);
+ return rc;
+ }
+
+ use_default = !(pdata->ch_map[rtd->dai_link->id]->set_ch_map);
+ chmap = pdata->ch_map[rtd->dai_link->id]->channel_map;
+ num_channels = prtd->num_channels;
+
+ if (prtd->num_channels > 2) {
+ /*
+ * Currently the left and right gains are averaged an applied
+ * to all channels. This might not be desirable. But currently,
+ * there exists no API in userspace to send a list of gains for
+ * each channel either. If such an API does become available,
+ * the mixer control must be updated to accept more than 2
+ * channel gains.
+ *
+ */
+ avg_vol = (volume_l + volume_r) / 2;
+ rc = q6asm_set_volume(prtd->audio_client, avg_vol);
+ } else {
+ gain_list[0] = volume_l;
+ gain_list[1] = volume_r;
+ /* force sending FR/FL/FC volume for mono */
+ if (prtd->num_channels == 1) {
+ gain_list[2] = volume_l;
+ num_channels = 3;
+ use_default = true;
+ }
+ rc = q6asm_set_multich_gain(prtd->audio_client, num_channels,
+ gain_list, chmap, use_default);
+ }
+
+ if (rc < 0)
+ pr_err("%s: Send vol gain command failed rc=%d\n",
+ __func__, rc);
+
+ return rc;
+}
+
+static int msm_compr_send_ddp_cfg(struct audio_client *ac,
+ struct snd_dec_ddp *ddp,
+ int stream_id)
+{
+ int i, rc;
+
+ pr_debug("%s\n", __func__);
+ for (i = 0; i < ddp->params_length; i++) {
+ rc = q6asm_ds1_set_stream_endp_params(ac, ddp->params_id[i],
+ ddp->params_value[i],
+ stream_id);
+ if (rc) {
+ pr_err("sending params_id: %d failed\n",
+ ddp->params_id[i]);
+ return rc;
+ }
+ }
+ return 0;
+}
+
+static int msm_compr_send_buffer(struct msm_compr_audio *prtd)
+{
+ int buffer_length;
+ uint64_t bytes_available;
+ struct audio_aio_write_param param;
+ struct snd_codec_metadata *buff_addr;
+
+ if (!atomic_read(&prtd->start)) {
+ pr_err("%s: stream is not in started state\n", __func__);
+ return -EINVAL;
+ }
+
+
+ if (atomic_read(&prtd->xrun)) {
+ WARN(1, "%s called while xrun is true", __func__);
+ return -EPERM;
+ }
+
+ pr_debug("%s: bytes_received = %llu copied_total = %llu\n",
+ __func__, prtd->bytes_received, prtd->copied_total);
+ if (prtd->first_buffer && prtd->gapless_state.use_dsp_gapless_mode &&
+ prtd->compr_passthr == LEGACY_PCM)
+ q6asm_stream_send_meta_data(prtd->audio_client,
+ prtd->audio_client->stream_id,
+ prtd->gapless_state.initial_samples_drop,
+ prtd->gapless_state.trailing_samples_drop);
+
+ buffer_length = prtd->codec_param.buffer.fragment_size;
+ bytes_available = prtd->bytes_received - prtd->copied_total;
+ if (bytes_available < prtd->codec_param.buffer.fragment_size)
+ buffer_length = bytes_available;
+
+ if (prtd->byte_offset + buffer_length > prtd->buffer_size) {
+ buffer_length = (prtd->buffer_size - prtd->byte_offset);
+ pr_debug("%s: wrap around situation, send partial data %d now",
+ __func__, buffer_length);
+ }
+
+ if (buffer_length) {
+ param.paddr = prtd->buffer_paddr + prtd->byte_offset;
+ WARN(prtd->byte_offset % 32 != 0, "offset %x not multiple of 32\n",
+ prtd->byte_offset);
+ } else {
+ param.paddr = prtd->buffer_paddr;
+ }
+ param.len = buffer_length;
+ if (prtd->ts_header_offset) {
+ buff_addr = (struct snd_codec_metadata *)
+ (prtd->buffer + prtd->byte_offset);
+ param.len = buff_addr->length;
+ param.msw_ts = (uint32_t)
+ ((buff_addr->timestamp & 0xFFFFFFFF00000000LL) >> 32);
+ param.lsw_ts = (uint32_t) (buff_addr->timestamp & 0xFFFFFFFFLL);
+ param.paddr += prtd->ts_header_offset;
+ param.flags = SET_TIMESTAMP;
+ param.metadata_len = prtd->ts_header_offset;
+ } else {
+ param.msw_ts = 0;
+ param.lsw_ts = 0;
+ param.flags = NO_TIMESTAMP;
+ param.metadata_len = 0;
+ }
+ param.uid = buffer_length;
+ param.last_buffer = prtd->last_buffer;
+
+ pr_debug("%s: sending %d bytes to DSP byte_offset = %d\n",
+ __func__, param.len, prtd->byte_offset);
+ if (q6asm_async_write(prtd->audio_client, ¶m) < 0) {
+ pr_err("%s:q6asm_async_write failed\n", __func__);
+ } else {
+ prtd->bytes_sent += buffer_length;
+ if (prtd->first_buffer)
+ prtd->first_buffer = 0;
+ }
+
+ return 0;
+}
+
+static int msm_compr_read_buffer(struct msm_compr_audio *prtd)
+{
+ int buffer_length;
+ uint64_t bytes_available;
+ uint64_t buffer_sent;
+ struct audio_aio_read_param param;
+ int ret;
+
+ if (!atomic_read(&prtd->start)) {
+ pr_err("%s: stream is not in started state\n", __func__);
+ return -EINVAL;
+ }
+
+ buffer_length = prtd->codec_param.buffer.fragment_size -
+ prtd->ts_header_offset;
+ bytes_available = prtd->received_total - prtd->bytes_copied;
+ buffer_sent = prtd->bytes_read - prtd->bytes_copied;
+ if (buffer_sent + buffer_length + prtd->ts_header_offset
+ > prtd->buffer_size) {
+ pr_debug(" %s : Buffer is Full bytes_available: %llu\n",
+ __func__, bytes_available);
+ return 0;
+ }
+
+ memset(¶m, 0x0, sizeof(struct audio_aio_read_param));
+ param.paddr = prtd->buffer_paddr + prtd->bytes_read_offset +
+ prtd->ts_header_offset;
+ param.len = buffer_length;
+ param.uid = buffer_length;
+ param.flags = prtd->codec_param.codec.flags;
+
+ pr_debug("%s: reading %d bytes from DSP byte_offset = %llu\n",
+ __func__, buffer_length, prtd->bytes_read);
+ ret = q6asm_async_read(prtd->audio_client, ¶m);
+ if (ret < 0) {
+ pr_err("%s: q6asm_async_read failed - %d\n",
+ __func__, ret);
+ return ret;
+ }
+ prtd->bytes_read += buffer_length;
+ prtd->bytes_read_offset += buffer_length;
+ if (prtd->bytes_read_offset >= prtd->buffer_size)
+ prtd->bytes_read_offset -= prtd->buffer_size;
+
+ return 0;
+}
+
+static void compr_event_handler(uint32_t opcode,
+ uint32_t token, uint32_t *payload, void *priv)
+{
+ struct msm_compr_audio *prtd = priv;
+ struct snd_compr_stream *cstream;
+ struct audio_client *ac;
+ uint32_t chan_mode = 0;
+ uint32_t sample_rate = 0;
+ uint64_t bytes_available;
+ int stream_id;
+ uint32_t stream_index;
+ unsigned long flags;
+ uint64_t read_size;
+ uint32_t *buff_addr;
+ struct snd_soc_pcm_runtime *rtd;
+ int ret = 0;
+
+ if (!prtd) {
+ pr_err("%s: prtd is NULL\n", __func__);
+ return;
+ }
+ cstream = prtd->cstream;
+ if (!cstream) {
+ pr_err("%s: cstream is NULL\n", __func__);
+ return;
+ }
+
+ ac = prtd->audio_client;
+
+ /*
+ * Token for rest of the compressed commands use to set
+ * session id, stream id, dir etc.
+ */
+ stream_id = q6asm_get_stream_id_from_token(token);
+
+ pr_debug("%s opcode =%08x\n", __func__, opcode);
+ switch (opcode) {
+ case ASM_DATA_EVENT_WRITE_DONE_V2:
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ if (payload[3]) {
+ pr_err("%s: WRITE FAILED w/ err 0x%x !, paddr 0x%x, byte_offset=%d,copied_total=%llu,token=%d\n",
+ __func__,
+ payload[3],
+ payload[0],
+ prtd->byte_offset,
+ prtd->copied_total, token);
+
+ if (atomic_cmpxchg(&prtd->drain, 1, 0) &&
+ prtd->last_buffer) {
+ pr_debug("%s: wake up on drain\n", __func__);
+ prtd->drain_ready = 1;
+ wake_up(&prtd->drain_wait);
+ prtd->last_buffer = 0;
+ } else {
+ atomic_set(&prtd->start, 0);
+ }
+ } else {
+ pr_debug("ASM_DATA_EVENT_WRITE_DONE_V2 offset %d, length %d\n",
+ prtd->byte_offset, token);
+ }
+
+ /*
+ * Token for WRITE command represents the amount of data
+ * written to ADSP in the last write, update offset and
+ * total copied data accordingly.
+ */
+ if (prtd->ts_header_offset) {
+ /* Always assume that the data will be sent to DSP on
+ * frame boundary.
+ * i.e, one frame of userspace write will result in
+ * one kernel write to DSP. This is needed as
+ * timestamp will be sent per frame.
+ */
+ prtd->byte_offset +=
+ prtd->codec_param.buffer.fragment_size;
+ prtd->copied_total +=
+ prtd->codec_param.buffer.fragment_size;
+ } else {
+ prtd->byte_offset += token;
+ prtd->copied_total += token;
+ }
+ if (prtd->byte_offset >= prtd->buffer_size)
+ prtd->byte_offset -= prtd->buffer_size;
+
+ snd_compr_fragment_elapsed(cstream);
+
+ if (!atomic_read(&prtd->start)) {
+ /* Writes must be restarted from _copy() */
+ pr_debug("write_done received while not started, treat as xrun");
+ atomic_set(&prtd->xrun, 1);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+
+ bytes_available = prtd->bytes_received - prtd->copied_total;
+ if (bytes_available < cstream->runtime->fragment_size) {
+ pr_debug("WRITE_DONE Insufficient data to send. break out\n");
+ atomic_set(&prtd->xrun, 1);
+
+ if (prtd->last_buffer)
+ prtd->last_buffer = 0;
+ if (atomic_read(&prtd->drain)) {
+ pr_debug("wake up on drain\n");
+ prtd->drain_ready = 1;
+ wake_up(&prtd->drain_wait);
+ atomic_set(&prtd->drain, 0);
+ }
+ } else if ((bytes_available == cstream->runtime->fragment_size)
+ && atomic_read(&prtd->drain)) {
+ prtd->last_buffer = 1;
+ msm_compr_send_buffer(prtd);
+ prtd->last_buffer = 0;
+ } else
+ msm_compr_send_buffer(prtd);
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ case ASM_DATA_EVENT_READ_DONE_V2:
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ pr_debug("ASM_DATA_EVENT_READ_DONE_V2 offset %d, length %d\n",
+ prtd->byte_offset, payload[4]);
+
+ if (prtd->ts_header_offset) {
+ /* Update the header for received buffer */
+ buff_addr = prtd->buffer + prtd->byte_offset;
+ /* Write the length of the buffer */
+ *buff_addr = prtd->codec_param.buffer.fragment_size
+ - prtd->ts_header_offset;
+ buff_addr++;
+ /* Write the offset */
+ *buff_addr = prtd->ts_header_offset;
+ buff_addr++;
+ /* Write the TS LSW */
+ *buff_addr = payload[CAPTURE_META_DATA_TS_OFFSET_LSW];
+ buff_addr++;
+ /* Write the TS MSW */
+ *buff_addr = payload[CAPTURE_META_DATA_TS_OFFSET_MSW];
+ }
+ /* Always assume read_size is same as fragment_size */
+ read_size = prtd->codec_param.buffer.fragment_size;
+ prtd->byte_offset += read_size;
+ prtd->received_total += read_size;
+ if (prtd->byte_offset >= prtd->buffer_size)
+ prtd->byte_offset -= prtd->buffer_size;
+
+ snd_compr_fragment_elapsed(cstream);
+
+ if (!atomic_read(&prtd->start)) {
+ pr_debug("read_done received while not started, treat as xrun");
+ atomic_set(&prtd->xrun, 1);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+ msm_compr_read_buffer(prtd);
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+
+ case ASM_DATA_EVENT_RENDERED_EOS:
+ spin_lock_irqsave(&prtd->lock, flags);
+ pr_debug("%s: ASM_DATA_CMDRSP_EOS token 0x%x,stream id %d\n",
+ __func__, token, stream_id);
+ if (atomic_read(&prtd->eos) &&
+ !prtd->gapless_state.set_next_stream_id) {
+ pr_debug("ASM_DATA_CMDRSP_EOS wake up\n");
+ prtd->eos_ack = 1;
+ wake_up(&prtd->eos_wait);
+ }
+ atomic_set(&prtd->eos, 0);
+ stream_index = STREAM_ARRAY_INDEX(stream_id);
+ if (stream_index >= MAX_NUMBER_OF_STREAMS ||
+ stream_index < 0) {
+ pr_err("%s: Invalid stream index %d", __func__,
+ stream_index);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+
+ if (prtd->gapless_state.set_next_stream_id &&
+ prtd->gapless_state.stream_opened[stream_index]) {
+ pr_debug("%s: CMD_CLOSE stream_id %d\n",
+ __func__, stream_id);
+ q6asm_stream_cmd_nowait(ac, CMD_CLOSE, stream_id);
+ atomic_set(&prtd->close, 1);
+ prtd->gapless_state.stream_opened[stream_index] = 0;
+ prtd->gapless_state.set_next_stream_id = false;
+ }
+ if (prtd->gapless_state.gapless_transition)
+ prtd->gapless_state.gapless_transition = 0;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ case ASM_STREAM_PP_EVENT:
+ case ASM_STREAM_CMD_ENCDEC_EVENTS:
+ pr_debug("%s: ASM_STREAM_EVENT(0x%x)\n", __func__, opcode);
+ rtd = cstream->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 ctrl. err = %d\n",
+ __func__, ret);
+ return;
+ }
+ break;
+ case ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY:
+ case ASM_DATA_EVENT_ENC_SR_CM_CHANGE_NOTIFY: {
+ pr_debug("ASM_DATA_EVENT_SR_CM_CHANGE_NOTIFY\n");
+ chan_mode = payload[1] >> 16;
+ sample_rate = payload[2] >> 16;
+ if (prtd && (chan_mode != prtd->num_channels ||
+ sample_rate != prtd->sample_rate)) {
+ prtd->num_channels = chan_mode;
+ prtd->sample_rate = sample_rate;
+ }
+ }
+ /* Fallthrough here */
+ case APR_BASIC_RSP_RESULT: {
+ switch (payload[0]) {
+ case ASM_SESSION_CMD_RUN_V2:
+ /* check if the first buffer need to be sent to DSP */
+ pr_debug("ASM_SESSION_CMD_RUN_V2\n");
+
+ /* FIXME: A state is a better way, dealing with this */
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ if (cstream->direction == SND_COMPRESS_CAPTURE) {
+ atomic_set(&prtd->start, 1);
+ msm_compr_read_buffer(prtd);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+
+ if (!prtd->bytes_sent) {
+ bytes_available = prtd->bytes_received -
+ prtd->copied_total;
+ if (bytes_available <
+ cstream->runtime->fragment_size) {
+ pr_debug("CMD_RUN_V2 Insufficient data to send. break out\n");
+ atomic_set(&prtd->xrun, 1);
+ } else {
+ msm_compr_send_buffer(prtd);
+ }
+ }
+
+ /*
+ * The condition below ensures playback finishes in the
+ * follow cornercase
+ * WRITE(last buffer)
+ * WAIT_FOR_DRAIN
+ * PAUSE
+ * WRITE_DONE(X)
+ * RESUME
+ */
+ if ((prtd->copied_total == prtd->bytes_sent) &&
+ atomic_read(&prtd->drain)) {
+ pr_debug("RUN ack, wake up & continue pending drain\n");
+
+ if (prtd->last_buffer)
+ prtd->last_buffer = 0;
+
+ prtd->drain_ready = 1;
+ wake_up(&prtd->drain_wait);
+ atomic_set(&prtd->drain, 0);
+ }
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ case ASM_STREAM_CMD_FLUSH:
+ pr_debug("%s: ASM_STREAM_CMD_FLUSH:", __func__);
+ pr_debug("token 0x%x, stream id %d\n", token,
+ stream_id);
+ prtd->cmd_ack = 1;
+ break;
+ case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:
+ pr_debug("%s: ASM_DATA_CMD_REMOVE_INITIAL_SILENCE:",
+ __func__);
+ pr_debug("token 0x%x, stream id = %d\n", token,
+ stream_id);
+ break;
+ case ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:
+ pr_debug("%s: ASM_DATA_CMD_REMOVE_TRAILING_SILENCE:",
+ __func__);
+ pr_debug("token = 0x%x, stream id = %d\n", token,
+ stream_id);
+ break;
+ case ASM_STREAM_CMD_CLOSE:
+ pr_debug("%s: ASM_DATA_CMD_CLOSE:", __func__);
+ pr_debug("token 0x%x, stream id %d\n", token,
+ stream_id);
+ /*
+ * wakeup wait for stream avail on stream 3
+ * after stream 1 ends.
+ */
+ if (prtd->next_stream) {
+ pr_debug("%s:CLOSE:wakeup wait for stream\n",
+ __func__);
+ prtd->stream_available = 1;
+ wake_up(&prtd->wait_for_stream_avail);
+ prtd->next_stream = 0;
+ }
+ if (atomic_read(&prtd->close) &&
+ atomic_read(&prtd->wait_on_close)) {
+ prtd->cmd_ack = 1;
+ wake_up(&prtd->close_wait);
+ }
+ atomic_set(&prtd->close, 0);
+ break;
+ case ASM_STREAM_CMD_REGISTER_PP_EVENTS:
+ pr_debug("%s: ASM_STREAM_CMD_REGISTER_PP_EVENTS:",
+ __func__);
+ break;
+ default:
+ break;
+ }
+ break;
+ }
+ case ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3:
+ pr_debug("%s: ASM_SESSION_CMDRSP_GET_SESSIONTIME_V3\n",
+ __func__);
+ break;
+ case RESET_EVENTS:
+ pr_err("%s: Received reset events CB, move to error state",
+ __func__);
+ spin_lock_irqsave(&prtd->lock, flags);
+ /*
+ * Since ADSP is down, let this driver pretend that it copied
+ * all the bytes received, so that next write will be triggered
+ */
+ prtd->copied_total = prtd->bytes_received;
+ snd_compr_fragment_elapsed(cstream);
+ atomic_set(&prtd->error, 1);
+ wake_up(&prtd->drain_wait);
+ if (atomic_cmpxchg(&prtd->eos, 1, 0)) {
+ pr_debug("%s:unblock eos wait queues", __func__);
+ wake_up(&prtd->eos_wait);
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ default:
+ pr_debug("%s: Not Supported Event opcode[0x%x]\n",
+ __func__, opcode);
+ break;
+ }
+}
+
+static int msm_compr_get_partial_drain_delay(int frame_sz, int sample_rate)
+{
+ int delay_time_ms = 0;
+
+ delay_time_ms = ((DSP_NUM_OUTPUT_FRAME_BUFFERED * frame_sz * 1000) /
+ sample_rate) + DSP_PP_BUFFERING_IN_MSEC;
+ delay_time_ms = delay_time_ms > PARTIAL_DRAIN_ACK_EARLY_BY_MSEC ?
+ delay_time_ms - PARTIAL_DRAIN_ACK_EARLY_BY_MSEC : 0;
+
+ pr_debug("%s: frame_sz %d, sample_rate %d, partial drain delay %d\n",
+ __func__, frame_sz, sample_rate, delay_time_ms);
+ return delay_time_ms;
+}
+
+static void populate_codec_list(struct msm_compr_audio *prtd)
+{
+ pr_debug("%s\n", __func__);
+ prtd->compr_cap.direction = SND_COMPRESS_PLAYBACK;
+ prtd->compr_cap.min_fragment_size =
+ COMPR_PLAYBACK_MIN_FRAGMENT_SIZE;
+ prtd->compr_cap.max_fragment_size =
+ COMPR_PLAYBACK_MAX_FRAGMENT_SIZE;
+ prtd->compr_cap.min_fragments =
+ COMPR_PLAYBACK_MIN_NUM_FRAGMENTS;
+ prtd->compr_cap.max_fragments =
+ COMPR_PLAYBACK_MAX_NUM_FRAGMENTS;
+ prtd->compr_cap.num_codecs = 17;
+ prtd->compr_cap.codecs[0] = SND_AUDIOCODEC_MP3;
+ prtd->compr_cap.codecs[1] = SND_AUDIOCODEC_AAC;
+ prtd->compr_cap.codecs[2] = SND_AUDIOCODEC_AC3;
+ prtd->compr_cap.codecs[3] = SND_AUDIOCODEC_EAC3;
+ prtd->compr_cap.codecs[4] = SND_AUDIOCODEC_MP2;
+ prtd->compr_cap.codecs[5] = SND_AUDIOCODEC_PCM;
+ prtd->compr_cap.codecs[6] = SND_AUDIOCODEC_WMA;
+ prtd->compr_cap.codecs[7] = SND_AUDIOCODEC_WMA_PRO;
+ prtd->compr_cap.codecs[8] = SND_AUDIOCODEC_FLAC;
+ prtd->compr_cap.codecs[9] = SND_AUDIOCODEC_VORBIS;
+ prtd->compr_cap.codecs[10] = SND_AUDIOCODEC_ALAC;
+ prtd->compr_cap.codecs[11] = SND_AUDIOCODEC_APE;
+ prtd->compr_cap.codecs[12] = SND_AUDIOCODEC_DTS;
+ prtd->compr_cap.codecs[13] = SND_AUDIOCODEC_DSD;
+ prtd->compr_cap.codecs[14] = SND_AUDIOCODEC_APTX;
+ prtd->compr_cap.codecs[15] = SND_AUDIOCODEC_TRUEHD;
+ prtd->compr_cap.codecs[16] = SND_AUDIOCODEC_IEC61937;
+}
+
+static int msm_compr_send_media_format_block(struct snd_compr_stream *cstream,
+ int stream_id,
+ bool use_gapless_codec_options)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct msm_compr_pdata *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ struct asm_aac_cfg aac_cfg;
+ struct asm_wma_cfg wma_cfg;
+ struct asm_wmapro_cfg wma_pro_cfg;
+ struct asm_flac_cfg flac_cfg;
+ struct asm_vorbis_cfg vorbis_cfg;
+ struct asm_alac_cfg alac_cfg;
+ struct asm_ape_cfg ape_cfg;
+ struct asm_dsd_cfg dsd_cfg;
+ struct aptx_dec_bt_addr_cfg aptx_cfg;
+ union snd_codec_options *codec_options;
+
+ int ret = 0;
+ uint16_t bit_width;
+ bool use_default_chmap = true;
+ char *chmap = NULL;
+ uint16_t sample_word_size;
+
+ pr_debug("%s: use_gapless_codec_options %d\n",
+ __func__, use_gapless_codec_options);
+
+ if (use_gapless_codec_options)
+ codec_options = &(prtd->gapless_state.codec_options);
+ else
+ codec_options = &(prtd->codec_param.codec.options);
+
+ if (!codec_options) {
+ pr_err("%s: codec_options is NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ switch (prtd->codec) {
+ case FORMAT_LINEAR_PCM:
+ pr_debug("SND_AUDIOCODEC_PCM\n");
+ if (pdata->ch_map[rtd->dai_link->id]) {
+ use_default_chmap =
+ !(pdata->ch_map[rtd->dai_link->id]->set_ch_map);
+ chmap =
+ pdata->ch_map[rtd->dai_link->id]->channel_map;
+ }
+
+ switch (prtd->codec_param.codec.format) {
+ case SNDRV_PCM_FORMAT_S32_LE:
+ bit_width = 32;
+ sample_word_size = 32;
+ break;
+ case SNDRV_PCM_FORMAT_S24_LE:
+ bit_width = 24;
+ sample_word_size = 32;
+ break;
+ case SNDRV_PCM_FORMAT_S24_3LE:
+ bit_width = 24;
+ sample_word_size = 24;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ default:
+ bit_width = 16;
+ sample_word_size = 16;
+ break;
+ }
+ ret = q6asm_media_format_block_pcm_format_support_v4(
+ prtd->audio_client,
+ prtd->sample_rate,
+ prtd->num_channels,
+ bit_width, stream_id,
+ use_default_chmap,
+ chmap,
+ sample_word_size,
+ ASM_LITTLE_ENDIAN,
+ DEFAULT_QF);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+
+ break;
+ case FORMAT_MP3:
+ pr_debug("SND_AUDIOCODEC_MP3\n");
+ /* no media format block needed */
+ break;
+ case FORMAT_MPEG4_AAC:
+ pr_debug("SND_AUDIOCODEC_AAC\n");
+ memset(&aac_cfg, 0x0, sizeof(struct asm_aac_cfg));
+ aac_cfg.aot = AAC_ENC_MODE_EAAC_P;
+ if (prtd->codec_param.codec.format ==
+ SND_AUDIOSTREAMFORMAT_MP4ADTS)
+ aac_cfg.format = 0x0;
+ else if (prtd->codec_param.codec.format ==
+ SND_AUDIOSTREAMFORMAT_MP4LATM)
+ aac_cfg.format = 0x04;
+ else
+ aac_cfg.format = 0x03;
+ aac_cfg.ch_cfg = prtd->num_channels;
+ aac_cfg.sample_rate = prtd->sample_rate;
+ ret = q6asm_stream_media_format_block_aac(prtd->audio_client,
+ &aac_cfg, stream_id);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+ break;
+ case FORMAT_AC3:
+ pr_debug("SND_AUDIOCODEC_AC3\n");
+ break;
+ case FORMAT_EAC3:
+ pr_debug("SND_AUDIOCODEC_EAC3\n");
+ break;
+ case FORMAT_WMA_V9:
+ pr_debug("SND_AUDIOCODEC_WMA\n");
+ memset(&wma_cfg, 0x0, sizeof(struct asm_wma_cfg));
+ wma_cfg.format_tag = prtd->codec_param.codec.format;
+ wma_cfg.ch_cfg = prtd->codec_param.codec.ch_in;
+ wma_cfg.sample_rate = prtd->sample_rate;
+ wma_cfg.avg_bytes_per_sec = codec_options->wma.avg_bit_rate/8;
+ wma_cfg.block_align = codec_options->wma.super_block_align;
+ wma_cfg.valid_bits_per_sample =
+ codec_options->wma.bits_per_sample;
+ wma_cfg.ch_mask = codec_options->wma.channelmask;
+ wma_cfg.encode_opt = codec_options->wma.encodeopt;
+ ret = q6asm_media_format_block_wma(prtd->audio_client,
+ &wma_cfg, stream_id);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+ break;
+ case FORMAT_WMA_V10PRO:
+ pr_debug("SND_AUDIOCODEC_WMA_PRO\n");
+ memset(&wma_pro_cfg, 0x0, sizeof(struct asm_wmapro_cfg));
+ wma_pro_cfg.format_tag = prtd->codec_param.codec.format;
+ wma_pro_cfg.ch_cfg = prtd->codec_param.codec.ch_in;
+ wma_pro_cfg.sample_rate = prtd->sample_rate;
+ wma_cfg.avg_bytes_per_sec = codec_options->wma.avg_bit_rate/8;
+ wma_pro_cfg.block_align = codec_options->wma.super_block_align;
+ wma_pro_cfg.valid_bits_per_sample =
+ codec_options->wma.bits_per_sample;
+ wma_pro_cfg.ch_mask = codec_options->wma.channelmask;
+ wma_pro_cfg.encode_opt = codec_options->wma.encodeopt;
+ wma_pro_cfg.adv_encode_opt = codec_options->wma.encodeopt1;
+ wma_pro_cfg.adv_encode_opt2 = codec_options->wma.encodeopt2;
+ ret = q6asm_media_format_block_wmapro(prtd->audio_client,
+ &wma_pro_cfg, stream_id);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed\n", __func__);
+ break;
+ case FORMAT_MP2:
+ pr_debug("%s: SND_AUDIOCODEC_MP2\n", __func__);
+ break;
+ case FORMAT_FLAC:
+ pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__);
+ memset(&flac_cfg, 0x0, sizeof(struct asm_flac_cfg));
+ flac_cfg.ch_cfg = prtd->num_channels;
+ flac_cfg.sample_rate = prtd->sample_rate;
+ flac_cfg.stream_info_present = 1;
+ flac_cfg.sample_size = codec_options->flac_dec.sample_size;
+ flac_cfg.min_blk_size = codec_options->flac_dec.min_blk_size;
+ flac_cfg.max_blk_size = codec_options->flac_dec.max_blk_size;
+ flac_cfg.max_frame_size =
+ codec_options->flac_dec.max_frame_size;
+ flac_cfg.min_frame_size =
+ codec_options->flac_dec.min_frame_size;
+
+ ret = q6asm_stream_media_format_block_flac(prtd->audio_client,
+ &flac_cfg, stream_id);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed ret %d\n",
+ __func__, ret);
+
+ break;
+ case FORMAT_VORBIS:
+ pr_debug("%s: SND_AUDIOCODEC_VORBIS\n", __func__);
+ memset(&vorbis_cfg, 0x0, sizeof(struct asm_vorbis_cfg));
+ vorbis_cfg.bit_stream_fmt =
+ codec_options->vorbis_dec.bit_stream_fmt;
+
+ ret = q6asm_stream_media_format_block_vorbis(
+ prtd->audio_client, &vorbis_cfg,
+ stream_id);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed ret %d\n",
+ __func__, ret);
+
+ break;
+ case FORMAT_ALAC:
+ pr_debug("%s: SND_AUDIOCODEC_ALAC\n", __func__);
+ memset(&alac_cfg, 0x0, sizeof(struct asm_alac_cfg));
+ alac_cfg.num_channels = prtd->num_channels;
+ alac_cfg.sample_rate = prtd->sample_rate;
+ alac_cfg.frame_length = codec_options->alac.frame_length;
+ alac_cfg.compatible_version =
+ codec_options->alac.compatible_version;
+ alac_cfg.bit_depth = codec_options->alac.bit_depth;
+ alac_cfg.pb = codec_options->alac.pb;
+ alac_cfg.mb = codec_options->alac.mb;
+ alac_cfg.kb = codec_options->alac.kb;
+ alac_cfg.max_run = codec_options->alac.max_run;
+ alac_cfg.max_frame_bytes = codec_options->alac.max_frame_bytes;
+ alac_cfg.avg_bit_rate = codec_options->alac.avg_bit_rate;
+ alac_cfg.channel_layout_tag =
+ codec_options->alac.channel_layout_tag;
+
+ ret = q6asm_media_format_block_alac(prtd->audio_client,
+ &alac_cfg, stream_id);
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed ret %d\n",
+ __func__, ret);
+ break;
+ case FORMAT_APE:
+ pr_debug("%s: SND_AUDIOCODEC_APE\n", __func__);
+ memset(&ape_cfg, 0x0, sizeof(struct asm_ape_cfg));
+ ape_cfg.num_channels = prtd->num_channels;
+ ape_cfg.sample_rate = prtd->sample_rate;
+ ape_cfg.compatible_version =
+ codec_options->ape.compatible_version;
+ ape_cfg.compression_level =
+ codec_options->ape.compression_level;
+ ape_cfg.format_flags = codec_options->ape.format_flags;
+ ape_cfg.blocks_per_frame = codec_options->ape.blocks_per_frame;
+ ape_cfg.final_frame_blocks =
+ codec_options->ape.final_frame_blocks;
+ ape_cfg.total_frames = codec_options->ape.total_frames;
+ ape_cfg.bits_per_sample = codec_options->ape.bits_per_sample;
+ ape_cfg.seek_table_present =
+ codec_options->ape.seek_table_present;
+
+ ret = q6asm_media_format_block_ape(prtd->audio_client,
+ &ape_cfg, stream_id);
+
+ if (ret < 0)
+ pr_err("%s: CMD Format block failed ret %d\n",
+ __func__, ret);
+ break;
+ case FORMAT_DTS:
+ pr_debug("SND_AUDIOCODEC_DTS\n");
+ /* no media format block needed */
+ break;
+ case FORMAT_DSD:
+ pr_debug("%s: SND_AUDIOCODEC_DSD\n", __func__);
+ memset(&dsd_cfg, 0x0, sizeof(struct asm_dsd_cfg));
+ dsd_cfg.num_channels = prtd->num_channels;
+ dsd_cfg.dsd_data_rate = prtd->sample_rate;
+ dsd_cfg.num_version = 0;
+ dsd_cfg.is_bitwise_big_endian = 1;
+ dsd_cfg.dsd_channel_block_size = 1;
+ ret = q6asm_media_format_block_dsd(prtd->audio_client,
+ &dsd_cfg, stream_id);
+ if (ret < 0)
+ pr_err("%s: CMD DSD Format block failed ret %d\n",
+ __func__, ret);
+ break;
+ case FORMAT_TRUEHD:
+ pr_debug("SND_AUDIOCODEC_TRUEHD\n");
+ /* no media format block needed */
+ break;
+ case FORMAT_IEC61937:
+ pr_debug("SND_AUDIOCODEC_IEC61937\n");
+ ret = q6asm_media_format_block_iec(prtd->audio_client,
+ prtd->sample_rate,
+ prtd->num_channels);
+ if (ret < 0)
+ pr_err("%s: CMD IEC61937 Format block failed ret %d\n",
+ __func__, ret);
+ break;
+ case FORMAT_APTX:
+ pr_debug("SND_AUDIOCODEC_APTX\n");
+ memset(&aptx_cfg, 0x0, sizeof(struct aptx_dec_bt_addr_cfg));
+ ret = q6asm_stream_media_format_block_aptx_dec(
+ prtd->audio_client,
+ prtd->sample_rate,
+ stream_id);
+ if (ret >= 0) {
+ aptx_cfg.nap = codec_options->aptx_dec.nap;
+ aptx_cfg.uap = codec_options->aptx_dec.uap;
+ aptx_cfg.lap = codec_options->aptx_dec.lap;
+ q6asm_set_aptx_dec_bt_addr(prtd->audio_client,
+ &aptx_cfg);
+ } else {
+ pr_err("%s: CMD Format block failed ret %d\n",
+ __func__, ret);
+ }
+ break;
+ default:
+ pr_debug("%s, unsupported format, skip", __func__);
+ break;
+ }
+ return ret;
+}
+
+static int msm_compr_init_pp_params(struct snd_compr_stream *cstream,
+ struct audio_client *ac)
+{
+ int ret = 0;
+ struct asm_softvolume_params softvol = {
+ .period = SOFT_VOLUME_PERIOD,
+ .step = SOFT_VOLUME_STEP,
+ .rampingcurve = SOFT_VOLUME_CURVE_LINEAR,
+ };
+
+ switch (ac->topology) {
+ default:
+ ret = q6asm_set_softvolume_v2(ac, &softvol,
+ SOFT_VOLUME_INSTANCE_1);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+
+ break;
+ }
+ return ret;
+}
+
+static int msm_compr_configure_dsp_for_playback
+ (struct snd_compr_stream *cstream)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data;
+ uint16_t bits_per_sample = 16;
+ int dir = IN, ret = 0;
+ struct audio_client *ac = prtd->audio_client;
+ uint32_t stream_index;
+ 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: stream_id %d\n", __func__, ac->stream_id);
+ stream_index = STREAM_ARRAY_INDEX(ac->stream_id);
+ if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) {
+ pr_err("%s: Invalid stream index:%d", __func__, stream_index);
+ return -EINVAL;
+ }
+
+ if ((prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_LE) ||
+ (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_3LE))
+ bits_per_sample = 24;
+ else if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S32_LE)
+ bits_per_sample = 32;
+
+ if (prtd->compr_passthr != LEGACY_PCM) {
+ ret = q6asm_open_write_compressed(ac, prtd->codec,
+ prtd->compr_passthr);
+ if (ret < 0) {
+ pr_err("%s:ASM open write err[%d] for compr_type[%d]\n",
+ __func__, ret, prtd->compr_passthr);
+ return ret;
+ }
+ prtd->gapless_state.stream_opened[stream_index] = 1;
+
+ ret = msm_pcm_routing_reg_phy_compr_stream(
+ soc_prtd->dai_link->id,
+ ac->perf_mode,
+ prtd->session_id,
+ SNDRV_PCM_STREAM_PLAYBACK,
+ prtd->compr_passthr);
+ if (ret) {
+ pr_err("%s: compr stream reg failed:%d\n", __func__,
+ ret);
+ return ret;
+ }
+ } else {
+ pr_debug("%s: stream_id %d bits_per_sample %d\n",
+ __func__, ac->stream_id, bits_per_sample);
+ ret = q6asm_stream_open_write_v4(ac,
+ prtd->codec, bits_per_sample,
+ ac->stream_id,
+ prtd->gapless_state.use_dsp_gapless_mode);
+ if (ret < 0) {
+ pr_err("%s:ASM open write err[%d] for compr type[%d]\n",
+ __func__, ret, prtd->compr_passthr);
+ return -ENOMEM;
+ }
+ prtd->gapless_state.stream_opened[stream_index] = 1;
+
+ pr_debug("%s: BE id %d\n", __func__, soc_prtd->dai_link->id);
+ ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->id,
+ ac->perf_mode,
+ prtd->session_id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ if (ret) {
+ pr_err("%s: stream reg failed:%d\n", __func__, ret);
+ return ret;
+ }
+ }
+
+ ret = msm_compr_set_volume(cstream, 0, 0);
+ if (ret < 0)
+ pr_err("%s : Set Volume failed : %d", __func__, ret);
+
+ if (prtd->compr_passthr != LEGACY_PCM) {
+ pr_debug("%s : Don't send cal and PP params for compress path",
+ __func__);
+ } else {
+ ret = q6asm_send_cal(ac);
+ if (ret < 0)
+ pr_debug("%s : Send cal failed : %d", __func__, ret);
+
+ ret = q6asm_set_softpause(ac, &softpause);
+ if (ret < 0)
+ pr_err("%s: Send SoftPause Param failed ret=%d\n",
+ __func__, ret);
+
+ ret = q6asm_set_softvolume(ac, &softvol);
+ if (ret < 0)
+ pr_err("%s: Send SoftVolume Param failed ret=%d\n",
+ __func__, ret);
+ }
+ ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE));
+ if (ret < 0) {
+ pr_err("%s: Set IO mode failed\n", __func__);
+ return -EINVAL;
+ }
+
+ runtime->fragments = prtd->codec_param.buffer.fragments;
+ runtime->fragment_size = prtd->codec_param.buffer.fragment_size;
+ pr_debug("allocate %d buffers each of size %d\n",
+ runtime->fragments,
+ runtime->fragment_size);
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir, ac,
+ runtime->fragment_size,
+ runtime->fragments);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret);
+ return -ENOMEM;
+ }
+
+ prtd->byte_offset = 0;
+ prtd->copied_total = 0;
+ prtd->app_pointer = 0;
+ prtd->bytes_received = 0;
+ prtd->bytes_sent = 0;
+ prtd->buffer = ac->port[dir].buf[0].data;
+ prtd->buffer_paddr = ac->port[dir].buf[0].phys;
+ prtd->buffer_size = runtime->fragments * runtime->fragment_size;
+
+ /* Bit-0 of flags represent timestamp mode */
+ if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG)
+ prtd->ts_header_offset = sizeof(struct snd_codec_metadata);
+ else
+ prtd->ts_header_offset = 0;
+
+ ret = msm_compr_send_media_format_block(cstream, ac->stream_id, false);
+ if (ret < 0)
+ pr_err("%s, failed to send media format block\n", __func__);
+
+ return ret;
+}
+
+static int msm_compr_configure_dsp_for_capture(struct snd_compr_stream *cstream)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *soc_prtd = cstream->private_data;
+ uint16_t bits_per_sample;
+ uint16_t sample_word_size;
+ int dir = OUT, ret = 0;
+ struct audio_client *ac = prtd->audio_client;
+ uint32_t stream_index;
+
+ switch (prtd->codec_param.codec.format) {
+ 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_S32_LE:
+ bits_per_sample = 32;
+ sample_word_size = 32;
+ break;
+ case SNDRV_PCM_FORMAT_S16_LE:
+ default:
+ bits_per_sample = 16;
+ sample_word_size = 16;
+ break;
+ }
+
+ pr_debug("%s: stream_id %d bits_per_sample %d\n",
+ __func__, ac->stream_id, bits_per_sample);
+
+ if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) {
+ ret = q6asm_open_read_v4(prtd->audio_client, FORMAT_LINEAR_PCM,
+ bits_per_sample, true);
+ } else {
+ 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:%d\n", __func__, ret);
+ return ret;
+ }
+
+ ret = msm_pcm_routing_reg_phy_stream(soc_prtd->dai_link->id,
+ ac->perf_mode,
+ prtd->session_id,
+ SNDRV_PCM_STREAM_CAPTURE);
+ if (ret) {
+ pr_err("%s: stream reg failed:%d\n", __func__, ret);
+ return ret;
+ }
+
+ ret = q6asm_set_io_mode(ac, (COMPRESSED_STREAM_IO | ASYNC_IO_MODE));
+ if (ret < 0) {
+ pr_err("%s: Set IO mode failed\n", __func__);
+ return -EINVAL;
+ }
+
+ stream_index = STREAM_ARRAY_INDEX(ac->stream_id);
+ if (stream_index >= MAX_NUMBER_OF_STREAMS || stream_index < 0) {
+ pr_err("%s: Invalid stream index:%d", __func__, stream_index);
+ return -EINVAL;
+ }
+
+ runtime->fragments = prtd->codec_param.buffer.fragments;
+ runtime->fragment_size = prtd->codec_param.buffer.fragment_size;
+ pr_debug("%s: allocate %d buffers each of size %d\n",
+ __func__, runtime->fragments,
+ runtime->fragment_size);
+ ret = q6asm_audio_client_buf_alloc_contiguous(dir, ac,
+ runtime->fragment_size,
+ runtime->fragments);
+ if (ret < 0) {
+ pr_err("Audio Start: Buffer Allocation failed rc = %d\n", ret);
+ return -ENOMEM;
+ }
+
+ prtd->byte_offset = 0;
+ prtd->received_total = 0;
+ prtd->app_pointer = 0;
+ prtd->bytes_copied = 0;
+ prtd->bytes_read = 0;
+ prtd->bytes_read_offset = 0;
+ prtd->buffer = ac->port[dir].buf[0].data;
+ prtd->buffer_paddr = ac->port[dir].buf[0].phys;
+ prtd->buffer_size = runtime->fragments * runtime->fragment_size;
+
+ /* Bit-0 of flags represent timestamp mode */
+ if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG)
+ prtd->ts_header_offset = sizeof(struct snd_codec_metadata);
+ else
+ prtd->ts_header_offset = 0;
+
+ pr_debug("%s: sample_rate = %d channels = %d bps = %d sample_word_size = %d\n",
+ __func__, prtd->sample_rate, prtd->num_channels,
+ bits_per_sample, sample_word_size);
+ ret = q6asm_enc_cfg_blk_pcm_format_support_v3(prtd->audio_client,
+ prtd->sample_rate, prtd->num_channels,
+ bits_per_sample, sample_word_size);
+
+ return ret;
+}
+
+static int msm_compr_playback_open(struct snd_compr_stream *cstream)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct msm_compr_audio *prtd;
+ struct msm_compr_pdata *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_compr_audio\n");
+ return -ENOMEM;
+ }
+
+ runtime->private_data = NULL;
+ prtd->cstream = cstream;
+ pdata->cstream[rtd->dai_link->id] = cstream;
+ pdata->audio_effects[rtd->dai_link->id] =
+ kzalloc(sizeof(struct msm_compr_audio_effects), GFP_KERNEL);
+ if (!pdata->audio_effects[rtd->dai_link->id]) {
+ pr_err("%s: Could not allocate memory for effects\n", __func__);
+ pdata->cstream[rtd->dai_link->id] = NULL;
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ pdata->dec_params[rtd->dai_link->id] =
+ kzalloc(sizeof(struct msm_compr_dec_params), GFP_KERNEL);
+ if (!pdata->dec_params[rtd->dai_link->id]) {
+ pr_err("%s: Could not allocate memory for dec params\n",
+ __func__);
+ kfree(pdata->audio_effects[rtd->dai_link->id]);
+ pdata->cstream[rtd->dai_link->id] = NULL;
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ prtd->codec = FORMAT_MP3;
+ prtd->bytes_received = 0;
+ prtd->bytes_sent = 0;
+ prtd->copied_total = 0;
+ prtd->byte_offset = 0;
+ prtd->sample_rate = 44100;
+ prtd->num_channels = 2;
+ prtd->drain_ready = 0;
+ prtd->last_buffer = 0;
+ prtd->first_buffer = 1;
+ prtd->partial_drain_delay = 0;
+ prtd->next_stream = 0;
+ memset(&prtd->gapless_state, 0, sizeof(struct msm_compr_gapless_state));
+ /*
+ * Update the use_dsp_gapless_mode from gapless struture with the value
+ * part of platform data.
+ */
+ prtd->gapless_state.use_dsp_gapless_mode = pdata->use_dsp_gapless_mode;
+
+ pr_debug("%s: gapless mode %d", __func__, pdata->use_dsp_gapless_mode);
+
+ spin_lock_init(&prtd->lock);
+
+ atomic_set(&prtd->eos, 0);
+ atomic_set(&prtd->start, 0);
+ atomic_set(&prtd->drain, 0);
+ atomic_set(&prtd->xrun, 0);
+ atomic_set(&prtd->close, 0);
+ atomic_set(&prtd->wait_on_close, 0);
+ atomic_set(&prtd->error, 0);
+
+ init_waitqueue_head(&prtd->eos_wait);
+ init_waitqueue_head(&prtd->drain_wait);
+ init_waitqueue_head(&prtd->close_wait);
+ init_waitqueue_head(&prtd->wait_for_stream_avail);
+
+ runtime->private_data = prtd;
+ populate_codec_list(prtd);
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)compr_event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_err("%s: Could not allocate memory for client\n", __func__);
+ kfree(pdata->audio_effects[rtd->dai_link->id]);
+ kfree(pdata->dec_params[rtd->dai_link->id]);
+ pdata->cstream[rtd->dai_link->id] = NULL;
+ runtime->private_data = NULL;
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+ prtd->audio_client->perf_mode = false;
+ prtd->session_id = prtd->audio_client->session;
+ msm_adsp_init_mixer_ctl_pp_event_queue(rtd);
+
+ return 0;
+}
+
+static int msm_compr_capture_open(struct snd_compr_stream *cstream)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct msm_compr_audio *prtd;
+ struct msm_compr_pdata *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+
+ pr_debug("%s\n", __func__);
+ prtd = kzalloc(sizeof(struct msm_compr_audio), GFP_KERNEL);
+ if (prtd == NULL) {
+ pr_err("Failed to allocate memory for msm_compr_audio\n");
+ return -ENOMEM;
+ }
+
+ runtime->private_data = NULL;
+ prtd->cstream = cstream;
+ pdata->cstream[rtd->dai_link->id] = cstream;
+
+ prtd->audio_client = q6asm_audio_client_alloc(
+ (app_cb)compr_event_handler, prtd);
+ if (!prtd->audio_client) {
+ pr_err("%s: Could not allocate memory for client\n", __func__);
+ pdata->cstream[rtd->dai_link->id] = NULL;
+ kfree(prtd);
+ return -ENOMEM;
+ }
+ pr_debug("%s: session ID %d\n", __func__, prtd->audio_client->session);
+ prtd->audio_client->perf_mode = false;
+ prtd->session_id = prtd->audio_client->session;
+ prtd->codec = FORMAT_LINEAR_PCM;
+ prtd->bytes_copied = 0;
+ prtd->bytes_read = 0;
+ prtd->bytes_read_offset = 0;
+ prtd->received_total = 0;
+ prtd->byte_offset = 0;
+ prtd->sample_rate = 48000;
+ prtd->num_channels = 2;
+ prtd->first_buffer = 0;
+
+ spin_lock_init(&prtd->lock);
+
+ atomic_set(&prtd->eos, 0);
+ atomic_set(&prtd->start, 0);
+ atomic_set(&prtd->drain, 0);
+ atomic_set(&prtd->xrun, 0);
+ atomic_set(&prtd->close, 0);
+ atomic_set(&prtd->wait_on_close, 0);
+ atomic_set(&prtd->error, 0);
+
+ runtime->private_data = prtd;
+
+ return 0;
+}
+
+static int msm_compr_open(struct snd_compr_stream *cstream)
+{
+ int ret = 0;
+
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ ret = msm_compr_playback_open(cstream);
+ else if (cstream->direction == SND_COMPRESS_CAPTURE)
+ ret = msm_compr_capture_open(cstream);
+ return ret;
+}
+
+static int msm_compr_playback_free(struct snd_compr_stream *cstream)
+{
+ struct snd_compr_runtime *runtime;
+ struct msm_compr_audio *prtd;
+ struct snd_soc_pcm_runtime *soc_prtd;
+ struct msm_compr_pdata *pdata;
+ struct audio_client *ac;
+ int dir = IN, ret = 0, stream_id;
+ unsigned long flags;
+ uint32_t stream_index;
+
+ pr_debug("%s\n", __func__);
+
+ if (!cstream) {
+ pr_err("%s cstream is null\n", __func__);
+ return 0;
+ }
+ runtime = cstream->runtime;
+ soc_prtd = cstream->private_data;
+ if (!runtime || !soc_prtd || !(soc_prtd->platform)) {
+ pr_err("%s runtime or soc_prtd or platform is null\n",
+ __func__);
+ return 0;
+ }
+ prtd = runtime->private_data;
+ if (!prtd) {
+ pr_err("%s prtd is null\n", __func__);
+ return 0;
+ }
+ prtd->cmd_interrupt = 1;
+ wake_up(&prtd->drain_wait);
+ pdata = snd_soc_platform_get_drvdata(soc_prtd->platform);
+ ac = prtd->audio_client;
+ if (!pdata || !ac) {
+ pr_err("%s pdata or ac is null\n", __func__);
+ return 0;
+ }
+ if (atomic_read(&prtd->eos)) {
+ ret = wait_event_timeout(prtd->eos_wait,
+ prtd->eos_ack, 5 * HZ);
+ if (!ret)
+ pr_err("%s: CMD_EOS failed\n", __func__);
+ }
+ if (atomic_read(&prtd->close)) {
+ prtd->cmd_ack = 0;
+ atomic_set(&prtd->wait_on_close, 1);
+ ret = wait_event_timeout(prtd->close_wait,
+ prtd->cmd_ack, 5 * HZ);
+ if (!ret)
+ pr_err("%s: CMD_CLOSE failed\n", __func__);
+ }
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ stream_id = ac->stream_id;
+ stream_index = STREAM_ARRAY_INDEX(NEXT_STREAM_ID(stream_id));
+
+ if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0) &&
+ (prtd->gapless_state.stream_opened[stream_index])) {
+ prtd->gapless_state.stream_opened[stream_index] = 0;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ pr_debug(" close stream %d", NEXT_STREAM_ID(stream_id));
+ q6asm_stream_cmd(ac, CMD_CLOSE, NEXT_STREAM_ID(stream_id));
+ spin_lock_irqsave(&prtd->lock, flags);
+ }
+
+ stream_index = STREAM_ARRAY_INDEX(stream_id);
+ if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0) &&
+ (prtd->gapless_state.stream_opened[stream_index])) {
+ prtd->gapless_state.stream_opened[stream_index] = 0;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ pr_debug("close stream %d", stream_id);
+ q6asm_stream_cmd(ac, CMD_CLOSE, stream_id);
+ spin_lock_irqsave(&prtd->lock, flags);
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ pdata->cstream[soc_prtd->dai_link->id] = NULL;
+ if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->id,
+ SNDRV_PCM_STREAM_PLAYBACK);
+ }
+
+ q6asm_audio_client_buf_free_contiguous(dir, ac);
+
+ q6asm_audio_client_free(ac);
+ msm_adsp_clean_mixer_ctl_pp_event_queue(soc_prtd);
+ kfree(pdata->audio_effects[soc_prtd->dai_link->id]);
+ pdata->audio_effects[soc_prtd->dai_link->id] = NULL;
+ kfree(pdata->dec_params[soc_prtd->dai_link->id]);
+ pdata->dec_params[soc_prtd->dai_link->id] = NULL;
+ kfree(prtd);
+ runtime->private_data = NULL;
+
+ return 0;
+}
+
+static int msm_compr_capture_free(struct snd_compr_stream *cstream)
+{
+ struct snd_compr_runtime *runtime;
+ struct msm_compr_audio *prtd;
+ struct snd_soc_pcm_runtime *soc_prtd;
+ struct msm_compr_pdata *pdata;
+ struct audio_client *ac;
+ int dir = OUT, stream_id;
+ unsigned long flags;
+ uint32_t stream_index;
+
+ if (!cstream) {
+ pr_err("%s cstream is null\n", __func__);
+ return 0;
+ }
+ runtime = cstream->runtime;
+ soc_prtd = cstream->private_data;
+ if (!runtime || !soc_prtd || !(soc_prtd->platform)) {
+ pr_err("%s runtime or soc_prtd or platform is null\n",
+ __func__);
+ return 0;
+ }
+ prtd = runtime->private_data;
+ if (!prtd) {
+ pr_err("%s prtd is null\n", __func__);
+ return 0;
+ }
+ pdata = snd_soc_platform_get_drvdata(soc_prtd->platform);
+ ac = prtd->audio_client;
+ if (!pdata || !ac) {
+ pr_err("%s pdata or ac is null\n", __func__);
+ return 0;
+ }
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ stream_id = ac->stream_id;
+
+ stream_index = STREAM_ARRAY_INDEX(stream_id);
+ if ((stream_index < MAX_NUMBER_OF_STREAMS && stream_index >= 0)) {
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ pr_debug("close stream %d", stream_id);
+ q6asm_stream_cmd(ac, CMD_CLOSE, stream_id);
+ spin_lock_irqsave(&prtd->lock, flags);
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ pdata->cstream[soc_prtd->dai_link->id] = NULL;
+ msm_pcm_routing_dereg_phy_stream(soc_prtd->dai_link->id,
+ SNDRV_PCM_STREAM_CAPTURE);
+
+ q6asm_audio_client_buf_free_contiguous(dir, ac);
+
+ q6asm_audio_client_free(ac);
+
+ kfree(prtd);
+ runtime->private_data = NULL;
+
+ return 0;
+}
+
+static int msm_compr_free(struct snd_compr_stream *cstream)
+{
+ int ret = 0;
+
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ ret = msm_compr_playback_free(cstream);
+ else if (cstream->direction == SND_COMPRESS_CAPTURE)
+ ret = msm_compr_capture_free(cstream);
+ return ret;
+}
+
+static bool msm_compr_validate_codec_compr(__u32 codec_id)
+{
+ int32_t i;
+
+ for (i = 0; i < ARRAY_SIZE(compr_codecs); i++) {
+ if (compr_codecs[i] == codec_id)
+ return true;
+ }
+ return false;
+}
+
+/* compress stream operations */
+static int msm_compr_set_params(struct snd_compr_stream *cstream,
+ struct snd_compr_params *params)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ int ret = 0, frame_sz = 0;
+ int i, num_rates;
+ bool is_format_gapless = false;
+
+ pr_debug("%s\n", __func__);
+
+ num_rates = sizeof(supported_sample_rates)/sizeof(unsigned int);
+ for (i = 0; i < num_rates; i++)
+ if (params->codec.sample_rate == supported_sample_rates[i])
+ break;
+ if (i == num_rates)
+ return -EINVAL;
+
+ memcpy(&prtd->codec_param, params, sizeof(struct snd_compr_params));
+ /* ToDo: remove duplicates */
+ prtd->num_channels = prtd->codec_param.codec.ch_in;
+ prtd->sample_rate = prtd->codec_param.codec.sample_rate;
+ pr_debug("%s: sample_rate %d\n", __func__, prtd->sample_rate);
+
+ if ((prtd->codec_param.codec.compr_passthr >= LEGACY_PCM &&
+ prtd->codec_param.
+ codec.compr_passthr <= COMPRESSED_PASSTHROUGH_DSD) ||
+ (prtd->codec_param.
+ codec.compr_passthr == COMPRESSED_PASSTHROUGH_IEC61937))
+ prtd->compr_passthr = prtd->codec_param.codec.compr_passthr;
+ else
+ prtd->compr_passthr = LEGACY_PCM;
+ pr_debug("%s: compr_passthr = %d", __func__, prtd->compr_passthr);
+ if (prtd->compr_passthr != LEGACY_PCM) {
+ pr_debug("%s: Reset gapless mode playback for compr_type[%d]\n",
+ __func__, prtd->compr_passthr);
+ prtd->gapless_state.use_dsp_gapless_mode = 0;
+ if (!msm_compr_validate_codec_compr(params->codec.id)) {
+ pr_err("%s codec not supported in passthrough,id =%d\n",
+ __func__, params->codec.id);
+ return -EINVAL;
+ }
+ }
+
+ switch (params->codec.id) {
+ case SND_AUDIOCODEC_PCM: {
+ pr_debug("SND_AUDIOCODEC_PCM\n");
+ prtd->codec = FORMAT_LINEAR_PCM;
+ is_format_gapless = true;
+ break;
+ }
+
+ case SND_AUDIOCODEC_MP3: {
+ pr_debug("SND_AUDIOCODEC_MP3\n");
+ prtd->codec = FORMAT_MP3;
+ frame_sz = MP3_OUTPUT_FRAME_SZ;
+ is_format_gapless = true;
+ break;
+ }
+
+ case SND_AUDIOCODEC_AAC: {
+ pr_debug("SND_AUDIOCODEC_AAC\n");
+ prtd->codec = FORMAT_MPEG4_AAC;
+ frame_sz = AAC_OUTPUT_FRAME_SZ;
+ is_format_gapless = true;
+ break;
+ }
+
+ case SND_AUDIOCODEC_AC3: {
+ pr_debug("SND_AUDIOCODEC_AC3\n");
+ prtd->codec = FORMAT_AC3;
+ frame_sz = AC3_OUTPUT_FRAME_SZ;
+ is_format_gapless = true;
+ break;
+ }
+
+ case SND_AUDIOCODEC_EAC3: {
+ pr_debug("SND_AUDIOCODEC_EAC3\n");
+ prtd->codec = FORMAT_EAC3;
+ frame_sz = EAC3_OUTPUT_FRAME_SZ;
+ is_format_gapless = true;
+ break;
+ }
+
+ case SND_AUDIOCODEC_MP2: {
+ pr_debug("SND_AUDIOCODEC_MP2\n");
+ prtd->codec = FORMAT_MP2;
+ break;
+ }
+
+ case SND_AUDIOCODEC_WMA: {
+ pr_debug("SND_AUDIOCODEC_WMA\n");
+ prtd->codec = FORMAT_WMA_V9;
+ break;
+ }
+
+ case SND_AUDIOCODEC_WMA_PRO: {
+ pr_debug("SND_AUDIOCODEC_WMA_PRO\n");
+ prtd->codec = FORMAT_WMA_V10PRO;
+ break;
+ }
+
+ case SND_AUDIOCODEC_FLAC: {
+ pr_debug("%s: SND_AUDIOCODEC_FLAC\n", __func__);
+ prtd->codec = FORMAT_FLAC;
+ /*
+ * DSP bufferring is based on blk size,
+ * consider mininum buffering to rule out any false wait
+ */
+ frame_sz =
+ prtd->codec_param.codec.options.flac_dec.min_blk_size;
+ is_format_gapless = true;
+ break;
+ }
+
+ case SND_AUDIOCODEC_VORBIS: {
+ pr_debug("%s: SND_AUDIOCODEC_VORBIS\n", __func__);
+ prtd->codec = FORMAT_VORBIS;
+ break;
+ }
+
+ case SND_AUDIOCODEC_ALAC: {
+ pr_debug("%s: SND_AUDIOCODEC_ALAC\n", __func__);
+ prtd->codec = FORMAT_ALAC;
+ break;
+ }
+
+ case SND_AUDIOCODEC_APE: {
+ pr_debug("%s: SND_AUDIOCODEC_APE\n", __func__);
+ prtd->codec = FORMAT_APE;
+ break;
+ }
+
+ case SND_AUDIOCODEC_DTS: {
+ pr_debug("%s: SND_AUDIOCODEC_DTS\n", __func__);
+ prtd->codec = FORMAT_DTS;
+ break;
+ }
+
+ case SND_AUDIOCODEC_DSD: {
+ pr_debug("%s: SND_AUDIOCODEC_DSD\n", __func__);
+ prtd->codec = FORMAT_DSD;
+ break;
+ }
+
+ case SND_AUDIOCODEC_TRUEHD: {
+ pr_debug("%s: SND_AUDIOCODEC_TRUEHD\n", __func__);
+ prtd->codec = FORMAT_TRUEHD;
+ break;
+ }
+
+ case SND_AUDIOCODEC_IEC61937: {
+ pr_debug("%s: SND_AUDIOCODEC_IEC61937\n", __func__);
+ prtd->codec = FORMAT_IEC61937;
+ break;
+ }
+
+ case SND_AUDIOCODEC_APTX: {
+ pr_debug("%s: SND_AUDIOCODEC_APTX\n", __func__);
+ prtd->codec = FORMAT_APTX;
+ break;
+ }
+
+ default:
+ pr_err("codec not supported, id =%d\n", params->codec.id);
+ return -EINVAL;
+ }
+
+ if (!is_format_gapless)
+ prtd->gapless_state.use_dsp_gapless_mode = false;
+
+ prtd->partial_drain_delay =
+ msm_compr_get_partial_drain_delay(frame_sz, prtd->sample_rate);
+
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ ret = msm_compr_configure_dsp_for_playback(cstream);
+ else if (cstream->direction == SND_COMPRESS_CAPTURE)
+ ret = msm_compr_configure_dsp_for_capture(cstream);
+
+ return ret;
+}
+
+static int msm_compr_drain_buffer(struct msm_compr_audio *prtd,
+ unsigned long *flags)
+{
+ int rc = 0;
+
+ atomic_set(&prtd->drain, 1);
+ prtd->drain_ready = 0;
+ spin_unlock_irqrestore(&prtd->lock, *flags);
+ pr_debug("%s: wait for buffer to be drained\n", __func__);
+ rc = wait_event_interruptible(prtd->drain_wait,
+ prtd->drain_ready ||
+ prtd->cmd_interrupt ||
+ atomic_read(&prtd->xrun) ||
+ atomic_read(&prtd->error));
+ pr_debug("%s: out of buffer drain wait with ret %d\n", __func__, rc);
+ spin_lock_irqsave(&prtd->lock, *flags);
+ if (prtd->cmd_interrupt) {
+ pr_debug("%s: buffer drain interrupted by flush)\n", __func__);
+ rc = -EINTR;
+ prtd->cmd_interrupt = 0;
+ }
+ if (atomic_read(&prtd->error)) {
+ pr_err("%s: Got RESET EVENTS notification, return\n",
+ __func__);
+ rc = -ENETRESET;
+ }
+ return rc;
+}
+
+static int msm_compr_wait_for_stream_avail(struct msm_compr_audio *prtd,
+ unsigned long *flags)
+{
+ int rc = 0;
+
+ pr_debug("next session is already in opened state\n");
+ prtd->next_stream = 1;
+ prtd->cmd_interrupt = 0;
+ spin_unlock_irqrestore(&prtd->lock, *flags);
+ /*
+ * Wait for stream to be available, or the wait to be interrupted by
+ * commands like flush or till a timeout of one second.
+ */
+ rc = wait_event_timeout(prtd->wait_for_stream_avail,
+ prtd->stream_available || prtd->cmd_interrupt, 1 * HZ);
+ pr_err("%s:prtd->stream_available %d, prtd->cmd_interrupt %d rc %d\n",
+ __func__, prtd->stream_available, prtd->cmd_interrupt, rc);
+
+ spin_lock_irqsave(&prtd->lock, *flags);
+ if (rc == 0) {
+ pr_err("%s: wait_for_stream_avail timed out\n",
+ __func__);
+ rc = -ETIMEDOUT;
+ } else if (prtd->cmd_interrupt == 1) {
+ /*
+ * This scenario might not happen as we do not allow
+ * flush in transition state.
+ */
+ pr_debug("%s: wait_for_stream_avail interrupted\n", __func__);
+ prtd->cmd_interrupt = 0;
+ prtd->stream_available = 0;
+ rc = -EINTR;
+ } else {
+ prtd->stream_available = 0;
+ rc = 0;
+ }
+ pr_debug("%s : rc = %d", __func__, rc);
+ return rc;
+}
+
+static int msm_compr_trigger(struct snd_compr_stream *cstream, int cmd)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct msm_compr_pdata *pdata =
+ snd_soc_platform_get_drvdata(rtd->platform);
+ uint32_t *volume = pdata->volume[rtd->dai_link->id];
+ struct audio_client *ac = prtd->audio_client;
+ unsigned long fe_id = rtd->dai_link->id;
+ int rc = 0;
+ int bytes_to_write;
+ unsigned long flags;
+ int stream_id;
+ uint32_t stream_index;
+ uint16_t bits_per_sample = 16;
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (atomic_read(&prtd->error)) {
+ pr_err("%s Got RESET EVENTS notification, return immediately",
+ __func__);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ return 0;
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ pr_debug("%s: SNDRV_PCM_TRIGGER_START\n", __func__);
+ atomic_set(&prtd->start, 1);
+
+ /*
+ * compr_set_volume and compr_init_pp_params
+ * are used to configure ASM volume hence not
+ * needed for compress passthrough playback.
+ *
+ * compress passthrough volume is controlled in
+ * ADM by adm_send_compressed_device_mute()
+ */
+ if (prtd->compr_passthr == LEGACY_PCM &&
+ cstream->direction == SND_COMPRESS_PLAYBACK) {
+ /* set volume for the stream before RUN */
+ rc = msm_compr_set_volume(cstream,
+ volume[0], volume[1]);
+ if (rc)
+ pr_err("%s : Set Volume failed : %d\n",
+ __func__, rc);
+
+ rc = msm_compr_init_pp_params(cstream, ac);
+ if (rc)
+ pr_err("%s : init PP params failed : %d\n",
+ __func__, rc);
+ } else {
+ msm_compr_read_buffer(prtd);
+ }
+ /* issue RUN command for the stream */
+ q6asm_run_nowait(prtd->audio_client, prtd->run_mode,
+ prtd->start_delay_msw, prtd->start_delay_lsw);
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ spin_lock_irqsave(&prtd->lock, flags);
+ pr_debug("%s: SNDRV_PCM_TRIGGER_STOP transition %d\n", __func__,
+ prtd->gapless_state.gapless_transition);
+ stream_id = ac->stream_id;
+ atomic_set(&prtd->start, 0);
+ if (cstream->direction == SND_COMPRESS_CAPTURE) {
+ q6asm_cmd_nowait(prtd->audio_client, CMD_PAUSE);
+ atomic_set(&prtd->xrun, 0);
+ prtd->received_total = 0;
+ prtd->bytes_copied = 0;
+ prtd->bytes_read = 0;
+ prtd->bytes_read_offset = 0;
+ prtd->byte_offset = 0;
+ prtd->app_pointer = 0;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+ if (prtd->next_stream) {
+ pr_debug("%s: interrupt next track wait queues\n",
+ __func__);
+ prtd->cmd_interrupt = 1;
+ wake_up(&prtd->wait_for_stream_avail);
+ prtd->next_stream = 0;
+ }
+ if (atomic_read(&prtd->eos)) {
+ pr_debug("%s: interrupt eos wait queues", __func__);
+ /*
+ * Gapless playback does not wait for eos, do not set
+ * cmd_int and do not wake up eos_wait during gapless
+ * transition
+ */
+ if (!prtd->gapless_state.gapless_transition) {
+ prtd->cmd_interrupt = 1;
+ wake_up(&prtd->eos_wait);
+ }
+ atomic_set(&prtd->eos, 0);
+ }
+ if (atomic_read(&prtd->drain)) {
+ pr_debug("%s: interrupt drain wait queues", __func__);
+ prtd->cmd_interrupt = 1;
+ prtd->drain_ready = 1;
+ wake_up(&prtd->drain_wait);
+ atomic_set(&prtd->drain, 0);
+ }
+ prtd->last_buffer = 0;
+ prtd->cmd_ack = 0;
+ if (!prtd->gapless_state.gapless_transition) {
+ pr_debug("issue CMD_FLUSH stream_id %d\n", stream_id);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ q6asm_stream_cmd(
+ prtd->audio_client, CMD_FLUSH, stream_id);
+ spin_lock_irqsave(&prtd->lock, flags);
+ } else {
+ prtd->first_buffer = 0;
+ }
+ /* FIXME. only reset if flush was successful */
+ prtd->byte_offset = 0;
+ prtd->copied_total = 0;
+ prtd->app_pointer = 0;
+ prtd->bytes_received = 0;
+ prtd->bytes_sent = 0;
+ prtd->marker_timestamp = 0;
+
+ atomic_set(&prtd->xrun, 0);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE_PUSH transition %d\n",
+ prtd->gapless_state.gapless_transition);
+ if (!prtd->gapless_state.gapless_transition) {
+ pr_debug("issue CMD_PAUSE stream_id %d\n",
+ ac->stream_id);
+ q6asm_stream_cmd_nowait(ac, CMD_PAUSE, ac->stream_id);
+ atomic_set(&prtd->start, 0);
+ }
+ break;
+ case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+ pr_debug("SNDRV_PCM_TRIGGER_PAUSE_RELEASE transition %d\n",
+ prtd->gapless_state.gapless_transition);
+ if (!prtd->gapless_state.gapless_transition) {
+ atomic_set(&prtd->start, 1);
+ q6asm_run_nowait(prtd->audio_client, prtd->run_mode,
+ 0, 0);
+ }
+ break;
+ case SND_COMPR_TRIGGER_PARTIAL_DRAIN:
+ pr_debug("%s: SND_COMPR_TRIGGER_PARTIAL_DRAIN\n", __func__);
+ if (!prtd->gapless_state.use_dsp_gapless_mode) {
+ pr_debug("%s: set partial drain as drain\n", __func__);
+ cmd = SND_COMPR_TRIGGER_DRAIN;
+ }
+ case SND_COMPR_TRIGGER_DRAIN:
+ pr_debug("%s: SNDRV_COMPRESS_DRAIN\n", __func__);
+ /* Make sure all the data is sent to DSP before sending EOS */
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ if (!atomic_read(&prtd->start)) {
+ pr_err("%s: stream is not in started state\n",
+ __func__);
+ rc = -EPERM;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+ if (prtd->bytes_received > prtd->copied_total) {
+ pr_debug("%s: wait till all the data is sent to dsp\n",
+ __func__);
+ rc = msm_compr_drain_buffer(prtd, &flags);
+ if (rc || !atomic_read(&prtd->start)) {
+ if (rc != -ENETRESET)
+ rc = -EINTR;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+ /*
+ * FIXME: Bug.
+ * Write(32767)
+ * Start
+ * Drain <- Indefinite wait
+ * sol1 : if (prtd->copied_total) then wait?
+ * sol2 : (prtd->cmd_interrupt || prtd->drain_ready ||
+ * atomic_read(xrun)
+ */
+ bytes_to_write = prtd->bytes_received
+ - prtd->copied_total;
+ WARN(bytes_to_write > runtime->fragment_size,
+ "last write %d cannot be > than fragment_size",
+ bytes_to_write);
+
+ if (bytes_to_write > 0) {
+ pr_debug("%s: send %d partial bytes at the end",
+ __func__, bytes_to_write);
+ atomic_set(&prtd->xrun, 0);
+ prtd->last_buffer = 1;
+ msm_compr_send_buffer(prtd);
+ }
+ }
+
+ if ((cmd == SND_COMPR_TRIGGER_PARTIAL_DRAIN) &&
+ (prtd->gapless_state.set_next_stream_id)) {
+ /* wait for the last buffer to be returned */
+
+ if (prtd->last_buffer) {
+ pr_debug("%s: last buffer drain\n", __func__);
+ rc = msm_compr_drain_buffer(prtd, &flags);
+ if (rc || !atomic_read(&prtd->start)) {
+ spin_unlock_irqrestore(&prtd->lock,
+ flags);
+ break;
+ }
+ }
+ /* send EOS */
+ prtd->eos_ack = 0;
+ atomic_set(&prtd->eos, 1);
+ pr_debug("issue CMD_EOS stream_id %d\n", ac->stream_id);
+ q6asm_stream_cmd_nowait(ac, CMD_EOS, ac->stream_id);
+ pr_info("PARTIAL DRAIN, do not wait for EOS ack\n");
+
+ /* send a zero length buffer */
+ atomic_set(&prtd->xrun, 0);
+ msm_compr_send_buffer(prtd);
+
+ /* wait for the zero length buffer to be returned */
+ pr_debug("%s: zero length buffer drain\n", __func__);
+ rc = msm_compr_drain_buffer(prtd, &flags);
+ if (rc || !atomic_read(&prtd->start)) {
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+
+ /* sleep for additional duration partial drain */
+ atomic_set(&prtd->drain, 1);
+ prtd->drain_ready = 0;
+ pr_debug("%s, additional sleep: %d\n", __func__,
+ prtd->partial_drain_delay);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ rc = wait_event_timeout(prtd->drain_wait,
+ prtd->drain_ready || prtd->cmd_interrupt,
+ msecs_to_jiffies(prtd->partial_drain_delay));
+ pr_debug("%s: out of additional wait for low sample rate\n",
+ __func__);
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (prtd->cmd_interrupt) {
+ pr_debug("%s: additional wait interrupted by flush)\n",
+ __func__);
+ rc = -EINTR;
+ prtd->cmd_interrupt = 0;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+
+ /* move to next stream and reset vars */
+ pr_debug("%s: Moving to next stream in gapless\n",
+ __func__);
+ ac->stream_id = NEXT_STREAM_ID(ac->stream_id);
+ prtd->byte_offset = 0;
+ prtd->app_pointer = 0;
+ prtd->first_buffer = 1;
+ prtd->last_buffer = 0;
+ /*
+ * Set gapless transition flag only if EOS hasn't been
+ * acknowledged already.
+ */
+ if (atomic_read(&prtd->eos))
+ prtd->gapless_state.gapless_transition = 1;
+ prtd->marker_timestamp = 0;
+
+ /*
+ * Don't reset these as these vars map to
+ * total_bytes_transferred and total_bytes_available
+ * directly, only total_bytes_transferred will be
+ * updated in the next avail() ioctl
+ * prtd->copied_total = 0;
+ * prtd->bytes_received = 0;
+ */
+ atomic_set(&prtd->drain, 0);
+ atomic_set(&prtd->xrun, 1);
+ pr_debug("%s: issue CMD_RUN", __func__);
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+ /*
+ * moving to next stream failed, so reset the gapless state
+ * set next stream id for the same session so that the same
+ * stream can be used for gapless playback
+ */
+ prtd->gapless_state.set_next_stream_id = false;
+ prtd->gapless_state.gapless_transition = 0;
+ pr_debug("%s:CMD_EOS stream_id %d\n", __func__, ac->stream_id);
+
+ prtd->eos_ack = 0;
+ atomic_set(&prtd->eos, 1);
+ q6asm_stream_cmd_nowait(ac, CMD_EOS, ac->stream_id);
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+
+ /* Wait indefinitely for DRAIN. Flush can also signal this*/
+ rc = wait_event_interruptible(prtd->eos_wait,
+ (prtd->eos_ack ||
+ prtd->cmd_interrupt ||
+ atomic_read(&prtd->error)));
+
+ if (rc < 0)
+ pr_err("%s: EOS wait failed\n", __func__);
+
+ pr_debug("%s: SNDRV_COMPRESS_DRAIN out of wait for EOS\n",
+ __func__);
+
+ if (prtd->cmd_interrupt)
+ rc = -EINTR;
+
+ if (atomic_read(&prtd->error)) {
+ pr_err("%s: Got RESET EVENTS notification, return\n",
+ __func__);
+ rc = -ENETRESET;
+ }
+
+ /*FIXME : what if a flush comes while PC is here */
+ if (rc == 0) {
+ /*
+ * Failed to open second stream in DSP for gapless
+ * so prepare the current stream in session
+ * for gapless playback
+ */
+ spin_lock_irqsave(&prtd->lock, flags);
+ pr_debug("%s:issue CMD_PAUSE stream_id %d",
+ __func__, ac->stream_id);
+ q6asm_stream_cmd_nowait(ac, CMD_PAUSE, ac->stream_id);
+ prtd->cmd_ack = 0;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ /*
+ * Cache this time as last known time
+ */
+ if (pdata->use_legacy_api)
+ q6asm_get_session_time_legacy(
+ prtd->audio_client,
+ &prtd->marker_timestamp);
+ else
+ q6asm_get_session_time(prtd->audio_client,
+ &prtd->marker_timestamp);
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ /*
+ * Don't reset these as these vars map to
+ * total_bytes_transferred and total_bytes_available.
+ * Just total_bytes_transferred will be updated
+ * in the next avail() ioctl.
+ * prtd->copied_total = 0;
+ * prtd->bytes_received = 0;
+ * do not reset prtd->bytes_sent as well as the same
+ * session is used for gapless playback
+ */
+ prtd->byte_offset = 0;
+
+ prtd->app_pointer = 0;
+ prtd->first_buffer = 1;
+ prtd->last_buffer = 0;
+ atomic_set(&prtd->drain, 0);
+ atomic_set(&prtd->xrun, 1);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ pr_debug("%s:issue CMD_FLUSH ac->stream_id %d",
+ __func__, ac->stream_id);
+ q6asm_stream_cmd(ac, CMD_FLUSH, ac->stream_id);
+
+ q6asm_run_nowait(prtd->audio_client, 0, 0, 0);
+ }
+ prtd->cmd_interrupt = 0;
+ break;
+ case SND_COMPR_TRIGGER_NEXT_TRACK:
+ if (!prtd->gapless_state.use_dsp_gapless_mode) {
+ pr_debug("%s: ignore trigger next track\n", __func__);
+ rc = 0;
+ break;
+ }
+ pr_debug("%s: SND_COMPR_TRIGGER_NEXT_TRACK\n", __func__);
+ spin_lock_irqsave(&prtd->lock, flags);
+ rc = 0;
+ /* next stream in gapless */
+ stream_id = NEXT_STREAM_ID(ac->stream_id);
+ /*
+ * Wait if stream 1 has not completed before honoring next
+ * track for stream 3. Scenario happens if second clip is
+ * small and fills in one buffer so next track will be
+ * called immediately.
+ */
+ stream_index = STREAM_ARRAY_INDEX(stream_id);
+ if (stream_index >= MAX_NUMBER_OF_STREAMS ||
+ stream_index < 0) {
+ pr_err("%s: Invalid stream index: %d", __func__,
+ stream_index);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ rc = -EINVAL;
+ break;
+ }
+
+ if (prtd->gapless_state.stream_opened[stream_index]) {
+ if (prtd->gapless_state.gapless_transition) {
+ rc = msm_compr_wait_for_stream_avail(prtd,
+ &flags);
+ } else {
+ /*
+ * If session is already opened break out if
+ * the state is not gapless transition. This
+ * is when seek happens after the last buffer
+ * is sent to the driver. Next track would be
+ * called again after last buffer is sent.
+ */
+ pr_debug("next session is in opened state\n");
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ break;
+ }
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ if (rc < 0) {
+ /*
+ * if return type EINTR then reset to zero. Tiny
+ * compress treats EINTR as error and prevents PARTIAL
+ * DRAIN. EINTR is not an error. wait for stream avail
+ * is interrupted by some other command like FLUSH.
+ */
+ if (rc == -EINTR) {
+ pr_debug("%s: EINTR reset rc to 0\n", __func__);
+ rc = 0;
+ }
+ break;
+ }
+
+ if (prtd->codec_param.codec.format == SNDRV_PCM_FORMAT_S24_LE)
+ bits_per_sample = 24;
+ else if (prtd->codec_param.codec.format ==
+ SNDRV_PCM_FORMAT_S32_LE)
+ bits_per_sample = 32;
+
+ pr_debug("%s: open_write stream_id %d bits_per_sample %d",
+ __func__, stream_id, bits_per_sample);
+ rc = q6asm_stream_open_write_v4(prtd->audio_client,
+ prtd->codec, bits_per_sample,
+ stream_id,
+ prtd->gapless_state.use_dsp_gapless_mode);
+ if (rc < 0) {
+ pr_err("%s: Session out open failed for gapless\n",
+ __func__);
+ break;
+ }
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ prtd->gapless_state.stream_opened[stream_index] = 1;
+ prtd->gapless_state.set_next_stream_id = true;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ rc = msm_compr_send_media_format_block(cstream,
+ stream_id, false);
+ if (rc < 0) {
+ pr_err("%s, failed to send media format block\n",
+ __func__);
+ break;
+ }
+ msm_compr_send_dec_params(cstream, pdata->dec_params[fe_id],
+ stream_id);
+ break;
+ }
+
+ return rc;
+}
+
+static int msm_compr_pointer(struct snd_compr_stream *cstream,
+ struct snd_compr_tstamp *arg)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct snd_soc_pcm_runtime *rtd = cstream->private_data;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ struct msm_compr_pdata *pdata = NULL;
+ struct snd_compr_tstamp tstamp;
+ uint64_t timestamp = 0;
+ int rc = 0, first_buffer;
+ unsigned long flags;
+ uint32_t gapless_transition;
+
+ pdata = snd_soc_platform_get_drvdata(rtd->platform);
+ pr_debug("%s\n", __func__);
+ memset(&tstamp, 0x0, sizeof(struct snd_compr_tstamp));
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ tstamp.sampling_rate = prtd->sample_rate;
+ tstamp.byte_offset = prtd->byte_offset;
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ tstamp.copied_total = prtd->copied_total;
+ else if (cstream->direction == SND_COMPRESS_CAPTURE)
+ tstamp.copied_total = prtd->received_total;
+ first_buffer = prtd->first_buffer;
+ if (atomic_read(&prtd->error)) {
+ pr_err("%s Got RESET EVENTS notification, return error\n",
+ __func__);
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ runtime->total_bytes_transferred = tstamp.copied_total;
+ else
+ runtime->total_bytes_available = tstamp.copied_total;
+ tstamp.pcm_io_frames = 0;
+ memcpy(arg, &tstamp, sizeof(struct snd_compr_tstamp));
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ return -ENETRESET;
+ }
+ if (cstream->direction == SND_COMPRESS_PLAYBACK) {
+
+ gapless_transition = prtd->gapless_state.gapless_transition;
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ if (gapless_transition)
+ pr_debug("%s session time in gapless transition",
+ __func__);
+ /*
+ *- Do not query if no buffer has been given.
+ *- Do not query on a gapless transition.
+ * Playback for the 2nd stream can start (thus returning time
+ * starting from 0) before the driver knows about EOS of first
+ * stream.
+ */
+ if (!first_buffer || gapless_transition) {
+
+ if (pdata->use_legacy_api)
+ rc = q6asm_get_session_time_legacy(
+ prtd->audio_client, &prtd->marker_timestamp);
+ else
+ rc = q6asm_get_session_time(
+ prtd->audio_client, &prtd->marker_timestamp);
+ if (rc < 0) {
+ pr_err("%s: Get Session Time return =%lld\n",
+ __func__, timestamp);
+ if (atomic_read(&prtd->error))
+ return -ENETRESET;
+ else
+ return -EAGAIN;
+ }
+ }
+ } else {
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ }
+ timestamp = prtd->marker_timestamp;
+
+ /* DSP returns timestamp in usec */
+ pr_debug("%s: timestamp = %lld usec\n", __func__, timestamp);
+ timestamp *= prtd->sample_rate;
+ tstamp.pcm_io_frames = (snd_pcm_uframes_t)div64_u64(timestamp, 1000000);
+ memcpy(arg, &tstamp, sizeof(struct snd_compr_tstamp));
+
+ return 0;
+}
+
+static int msm_compr_ack(struct snd_compr_stream *cstream,
+ size_t count)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ void *src, *dstn;
+ size_t copy;
+ unsigned long flags;
+
+ WARN(1, "This path is untested");
+ return -EINVAL;
+
+ pr_debug("%s: count = %zd\n", __func__, count);
+ if (!prtd->buffer) {
+ pr_err("%s: Buffer is not allocated yet ??\n", __func__);
+ return -EINVAL;
+ }
+ src = runtime->buffer + prtd->app_pointer;
+ dstn = prtd->buffer + prtd->app_pointer;
+ if (count < prtd->buffer_size - prtd->app_pointer) {
+ memcpy(dstn, src, count);
+ prtd->app_pointer += count;
+ } else {
+ copy = prtd->buffer_size - prtd->app_pointer;
+ memcpy(dstn, src, copy);
+ memcpy(prtd->buffer, runtime->buffer, count - copy);
+ prtd->app_pointer = count - copy;
+ }
+
+ /*
+ * If the stream is started and all the bytes received were
+ * copied to DSP, the newly received bytes should be
+ * sent right away
+ */
+ spin_lock_irqsave(&prtd->lock, flags);
+
+ if (atomic_read(&prtd->start) &&
+ prtd->bytes_received == prtd->copied_total) {
+ prtd->bytes_received += count;
+ msm_compr_send_buffer(prtd);
+ } else
+ prtd->bytes_received += count;
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return 0;
+}
+
+static int msm_compr_playback_copy(struct snd_compr_stream *cstream,
+ char __user *buf, size_t count)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ void *dstn;
+ size_t copy;
+ uint64_t bytes_available = 0;
+ unsigned long flags;
+
+ pr_debug("%s: count = %zd\n", __func__, count);
+ if (!prtd->buffer) {
+ pr_err("%s: Buffer is not allocated yet ??", __func__);
+ return 0;
+ }
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (atomic_read(&prtd->error)) {
+ pr_err("%s Got RESET EVENTS notification", __func__);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ return -ENETRESET;
+ }
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ dstn = prtd->buffer + prtd->app_pointer;
+ if (count < prtd->buffer_size - prtd->app_pointer) {
+ if (copy_from_user(dstn, buf, count))
+ return -EFAULT;
+ prtd->app_pointer += count;
+ } else {
+ copy = prtd->buffer_size - prtd->app_pointer;
+ if (copy_from_user(dstn, buf, copy))
+ return -EFAULT;
+ if (copy_from_user(prtd->buffer, buf + copy, count - copy))
+ return -EFAULT;
+ prtd->app_pointer = count - copy;
+ }
+
+ /*
+ * If stream is started and there has been an xrun,
+ * since the available bytes fits fragment_size, copy the data
+ * right away.
+ */
+ spin_lock_irqsave(&prtd->lock, flags);
+ prtd->bytes_received += count;
+ if (atomic_read(&prtd->start)) {
+ if (atomic_read(&prtd->xrun)) {
+ pr_debug("%s: in xrun, count = %zd\n", __func__, count);
+ bytes_available = prtd->bytes_received -
+ prtd->copied_total;
+ if (bytes_available >= runtime->fragment_size) {
+ pr_debug("%s: handle xrun, bytes_to_write = %llu\n",
+ __func__, bytes_available);
+ atomic_set(&prtd->xrun, 0);
+ msm_compr_send_buffer(prtd);
+ } /* else not sufficient data */
+ } /* writes will continue on the next write_done */
+ }
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+
+ return count;
+}
+
+static int msm_compr_capture_copy(struct snd_compr_stream *cstream,
+ char __user *buf, size_t count)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ void *source;
+ unsigned long flags;
+
+ pr_debug("%s: count = %zd\n", __func__, count);
+ if (!prtd->buffer) {
+ pr_err("%s: Buffer is not allocated yet ??", __func__);
+ return 0;
+ }
+
+ spin_lock_irqsave(&prtd->lock, flags);
+ if (atomic_read(&prtd->error)) {
+ pr_err("%s Got RESET EVENTS notification", __func__);
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ return -ENETRESET;
+ }
+
+ source = prtd->buffer + prtd->app_pointer;
+ /* check if we have requested amount of data to copy to user*/
+ if (count <= prtd->received_total - prtd->bytes_copied) {
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ if (copy_to_user(buf, source, count)) {
+ pr_err("copy_to_user failed");
+ return -EFAULT;
+ }
+ spin_lock_irqsave(&prtd->lock, flags);
+ prtd->app_pointer += count;
+ if (prtd->app_pointer >= prtd->buffer_size)
+ prtd->app_pointer -= prtd->buffer_size;
+ prtd->bytes_copied += count;
+ }
+ msm_compr_read_buffer(prtd);
+
+ spin_unlock_irqrestore(&prtd->lock, flags);
+ return count;
+}
+
+static int msm_compr_copy(struct snd_compr_stream *cstream,
+ char __user *buf, size_t count)
+{
+ int ret = 0;
+
+ pr_debug(" In %s\n", __func__);
+ if (cstream->direction == SND_COMPRESS_PLAYBACK)
+ ret = msm_compr_playback_copy(cstream, buf, count);
+ else if (cstream->direction == SND_COMPRESS_CAPTURE)
+ ret = msm_compr_capture_copy(cstream, buf, count);
+ return ret;
+}
+
+static int msm_compr_get_caps(struct snd_compr_stream *cstream,
+ struct snd_compr_caps *arg)
+{
+ struct snd_compr_runtime *runtime = cstream->runtime;
+ struct msm_compr_audio *prtd = runtime->private_data;
+ int ret = 0;
+
+ pr_debug("%s\n", __func__);
+ if ((arg != NULL) && (prtd != NULL)) {
+ memcpy(arg, &prtd->compr_cap, sizeof(struct snd_compr_caps));
+ } else {
+ ret = -EINVAL;
+ pr_err("%s: arg (0x%pK), prtd (0x%pK)\n", __func__, arg, prtd);
+ }
+
+ return ret;
+}
+
+static int msm_compr_get_codec_caps(struct snd_compr_stream *cstream,
+ struct snd_compr_codec_caps *codec)
+{
+ pr_debug("%s\n", __func__);
+
+ switch (codec->codec) {
+ case SND_AUDIOCODEC_MP3:
+ codec->num_descriptors = 2;
+ codec->descriptor[0].max_ch = 2;
+ memcpy(codec->descriptor[0].sample_rates,
+ supported_sample_rates,
+ sizeof(supported_sample_rates));
+ codec->descriptor[0].num_sample_rates =
+ sizeof(supported_sample_rates)/sizeof(unsigned int);
+ codec->descriptor[0].bit_rate[0] = 320; /* 320kbps */
+ codec->descriptor[0].bit_rate[1] = 128;
+ codec->descriptor[0].num_bitrates = 2;
+ codec->descriptor[0].profiles = 0;
+ codec->descriptor[0].modes = SND_AUDIOCHANMODE_MP3_STEREO;
+ codec->descriptor[0].formats = 0;
+ break;
+ case SND_AUDIOCODEC_AAC:
+ codec->num_descriptors = 2;
+ codec->descriptor[1].max_ch = 2;
+ memcpy(codec->descriptor[1].sample_rates,
+ supported_sample_rates,
+ sizeof(supported_sample_rates));
+ codec->descriptor[1].num_sample_rates =
+ sizeof(supported_sample_rates)/sizeof(unsigned int);
+ codec->descriptor[1].bit_rate[0] = 320; /* 320kbps */
+ codec->descriptor[1].bit_rate[1] = 128;
+ codec->descriptor[1].num_bitrates = 2;
+ codec->descriptor[1].profiles = 0;
+ codec->descriptor[1].modes = 0;
+ codec->descriptor[1].formats =
+ (SND_AUDIOSTREAMFORMAT_MP4ADTS |
+ SND_AUDIOSTREAMFORMAT_RAW);
+ break;
+ case SND_AUDIOCODEC_AC3:
+ case SND_AUDIOCODEC_EAC3:
+ case SND_AUDIOCODEC_FLAC:
+ case SND_AUDIOCODEC_VORBIS:
+ case SND_AUDIOCODEC_ALAC:
+ case SND_AUDIOCODEC_APE:
+ case SND_AUDIOCODEC_DTS:
+ case SND_AUDIOCODEC_DSD:
+ case SND_AUDIOCODEC_TRUEHD:
+ case SND_AUDIOCODEC_IEC61937:
+ case SND_AUDIOCODEC_APTX:
+ break;
+ default:
+ pr_err("%s: Unsupported audio codec %d\n",
+ __func__, codec->codec);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int msm_compr_set_metadata(struct snd_compr_stream *cstream,
+ struct snd_compr_metadata *metadata)
+{
+ struct msm_compr_audio *prtd;
+ struct audio_client *ac;
+ pr_debug("%s\n", __func__);
+
+ if (!metadata || !cstream)
+ return -EINVAL;
+
+ prtd = cstream->runtime->private_data;
+ if (!prtd || !prtd->audio_client) {
+ pr_err("%s: prtd or audio client is NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ if (((metadata->key == SNDRV_COMPRESS_ENCODER_PADDING) ||
+ (metadata->key == SNDRV_COMPRESS_ENCODER_DELAY)) &&
+ (prtd->compr_passthr != LEGACY_PCM)) {
+ pr_debug("%s: No trailing silence for compress_type[%d]\n",
+ __func__, prtd->compr_passthr);
+ return 0;
+ }
+
+ ac = prtd->audio_client;
+ if (metadata->key == SNDRV_COMPRESS_ENCODER_PADDING) {
+ pr_debug("%s, got encoder padding %u",
+ __func__, metadata->value[0]);
+ prtd->gapless_state.trailing_samples_drop = metadata->value[0];
+ } else if (metadata->key == SNDRV_COMPRESS_ENCODER_DELAY) {
+ pr_debug("%s, got encoder delay %u",
+ __func__, metadata->value[0]);
+ prtd->gapless_state.initial_samples_drop = metadata->value[0];
+ } else if (metadata->key == SNDRV_COMPRESS_RENDER_MODE) {
+ return msm_compr_set_render_mode(prtd, metadata->value[0]);
+ } else if (metadata->key == SNDRV_COMPRESS_CLK_REC_MODE) {
+ return msm_compr_set_clk_rec_mode(ac, metadata->value[0]);
+ } else if (metadata->key == SNDRV_COMPRESS_RENDER_WINDOW) {
+ return msm_compr_set_render_window(
+ ac,
+ metadata->value[0],
+ metadata->value[1],
+ metadata->value[2],
+ metadata->value[3]);
+ } else if (metadata->key == SNDRV_COMPRESS_START_DELAY) {
+ prtd->start_delay_lsw = metadata->value[0];
+ prtd->start_delay_msw = metadata->value[1];
+ } else if (metadata->key ==
+ SNDRV_COMPRESS_ENABLE_ADJUST_SESSION_CLOCK) {
+ return msm_compr_enable_adjust_session_clock(ac,
+ metadata->value[0]);
+ } else if (metadata->key == SNDRV_COMPRESS_ADJUST_SESSION_CLOCK) {
+ return msm_compr_adjust_session_clock(ac,
+ metadata->value[0],
+ metadata->value[1]);
+ }
+
+ return 0;
+}
+
+static int msm_compr_get_metadata(struct snd_compr_stream *cstream,
+ struct snd_compr_metadata *metadata)
+{
+ struct msm_compr_audio *prtd;
+ struct audio_client *ac;
+ int ret = -EINVAL;
+
+ pr_debug("%s\n", __func__);
+
+ if (!metadata || !cstream || !cstream->runtime)
+ return ret;
+
+ if (metadata->key != SNDRV_COMPRESS_PATH_DELAY) {
+ pr_err("%s, unsupported key %d\n", __func__, metadata->key);
+ return ret;
+ }
+
+ prtd = cstream->runtime->private_data;
+ if (!prtd || !prtd->audio_client) {
+ pr_err("%s: prtd or audio client is NULL\n", __func__);
+ return ret;
+ }
+
+ ac = prtd->audio_client;
+ ret = q6asm_get_path_delay(prtd->audio_client);
+ if (ret) {
+ pr_err("%s: get_path_delay failed, ret=%d\n", __func__, ret);
+ return ret;
+ }
+
+ pr_debug("%s, path delay(in us) %u\n", __func__, ac->path_delay);
+
+ metadata->value[0] = ac->path_delay;
+
+ return ret;
+}
+
+
+static int msm_compr_set_next_track_param(struct snd_compr_stream *cstream,
+ union snd_codec_options *codec_options)
+{
+ struct msm_compr_audio *prtd;
+ struct audio_client *ac;
+ int ret = 0;
+
+ if (!codec_options || !cstream)
+ return -EINVAL;
+
+ prtd = cstream->runtime->private_data;
+ if (!prtd || !prtd->audio_client) {
+ pr_err("%s: prtd or audio client is NULL\n", __func__);
+ return -EINVAL;
+ }
+
+ ac = prtd->audio_client;
+
+ pr_debug("%s: got codec options for codec type %u",
+ __func__, prtd->codec);
+ switch (prtd->codec) {
+ case FORMAT_WMA_V9:
+ case FORMAT_WMA_V10PRO:
+ case FORMAT_FLAC:
+ case FORMAT_VORBIS:
+ case FORMAT_ALAC:
+ case FORMAT_APE:
+ memcpy(&(prtd->gapless_state.codec_options),
+ codec_options,
+ sizeof(union snd_codec_options));
+ ret = msm_compr_send_media_format_block(cstream,
+ ac->stream_id, true);
+ if (ret < 0) {
+ pr_err("%s: failed to send media format block\n",
+ __func__);
+ }
+ break;
+
+ default:
+ pr_debug("%s: Ignore sending CMD Format block\n",
+ __func__);
+ break;
+ }
+
+ return ret;
+}
+
+static int msm_compr_volume_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct snd_compr_stream *cstream = NULL;
+ uint32_t *volume = NULL;
+
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bounds fe_id %lu\n",
+ __func__, fe_id);
+ return -EINVAL;
+ }
+
+ cstream = pdata->cstream[fe_id];
+ volume = pdata->volume[fe_id];
+
+ volume[0] = ucontrol->value.integer.value[0];
+ volume[1] = ucontrol->value.integer.value[1];
+ pr_debug("%s: fe_id %lu left_vol %d right_vol %d\n",
+ __func__, fe_id, volume[0], volume[1]);
+ if (cstream)
+ msm_compr_set_volume(cstream, volume[0], volume[1]);
+ return 0;
+}
+
+static int msm_compr_volume_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+
+ struct msm_compr_pdata *pdata =
+ snd_soc_component_get_drvdata(comp);
+ uint32_t *volume = NULL;
+
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bound fe_id %lu\n", __func__, fe_id);
+ return -EINVAL;
+ }
+
+ volume = pdata->volume[fe_id];
+ pr_debug("%s: fe_id %lu\n", __func__, fe_id);
+ ucontrol->value.integer.value[0] = volume[0];
+ ucontrol->value.integer.value[1] = volume[1];
+
+ return 0;
+}
+
+static int msm_compr_audio_effects_config_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct msm_compr_audio_effects *audio_effects = NULL;
+ struct snd_compr_stream *cstream = NULL;
+ struct msm_compr_audio *prtd = NULL;
+ long *values = &(ucontrol->value.integer.value[0]);
+ int effects_module;
+
+ pr_debug("%s\n", __func__);
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bounds fe_id %lu\n",
+ __func__, fe_id);
+ return -EINVAL;
+ }
+ cstream = pdata->cstream[fe_id];
+ audio_effects = pdata->audio_effects[fe_id];
+ if (!cstream || !audio_effects) {
+ pr_err("%s: stream or effects inactive\n", __func__);
+ return -EINVAL;
+ }
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: cannot set audio effects\n", __func__);
+ return -EINVAL;
+ }
+ if (prtd->compr_passthr != LEGACY_PCM) {
+ pr_debug("%s: No effects for compr_type[%d]\n",
+ __func__, prtd->compr_passthr);
+ return 0;
+ }
+ pr_debug("%s: Effects supported for compr_type[%d]\n",
+ __func__, prtd->compr_passthr);
+
+ effects_module = *values++;
+ switch (effects_module) {
+ case VIRTUALIZER_MODULE:
+ pr_debug("%s: VIRTUALIZER_MODULE\n", __func__);
+ if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
+ prtd->audio_client->topology))
+ msm_audio_effects_virtualizer_handler(
+ prtd->audio_client,
+ &(audio_effects->virtualizer),
+ values);
+ break;
+ case REVERB_MODULE:
+ pr_debug("%s: REVERB_MODULE\n", __func__);
+ if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
+ prtd->audio_client->topology))
+ msm_audio_effects_reverb_handler(prtd->audio_client,
+ &(audio_effects->reverb),
+ values);
+ break;
+ case BASS_BOOST_MODULE:
+ pr_debug("%s: BASS_BOOST_MODULE\n", __func__);
+ if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
+ prtd->audio_client->topology))
+ msm_audio_effects_bass_boost_handler(prtd->audio_client,
+ &(audio_effects->bass_boost),
+ values);
+ break;
+ case PBE_MODULE:
+ pr_debug("%s: PBE_MODULE\n", __func__);
+ if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
+ prtd->audio_client->topology))
+ msm_audio_effects_pbe_handler(prtd->audio_client,
+ &(audio_effects->pbe),
+ values);
+ break;
+ case EQ_MODULE:
+ pr_debug("%s: EQ_MODULE\n", __func__);
+ if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
+ prtd->audio_client->topology))
+ msm_audio_effects_popless_eq_handler(prtd->audio_client,
+ &(audio_effects->equalizer),
+ values);
+ break;
+ case SOFT_VOLUME_MODULE:
+ pr_debug("%s: SOFT_VOLUME_MODULE\n", __func__);
+ break;
+ case SOFT_VOLUME2_MODULE:
+ pr_debug("%s: SOFT_VOLUME2_MODULE\n", __func__);
+ if (msm_audio_effects_is_effmodule_supp_in_top(effects_module,
+ prtd->audio_client->topology))
+ msm_audio_effects_volume_handler_v2(prtd->audio_client,
+ &(audio_effects->volume),
+ values, SOFT_VOLUME_INSTANCE_2);
+ break;
+ default:
+ pr_err("%s Invalid effects config module\n", __func__);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static int msm_compr_audio_effects_config_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct msm_compr_audio_effects *audio_effects = NULL;
+ struct snd_compr_stream *cstream = NULL;
+ struct msm_compr_audio *prtd = NULL;
+
+ pr_debug("%s\n", __func__);
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bounds fe_id %lu\n",
+ __func__, fe_id);
+ return -EINVAL;
+ }
+ cstream = pdata->cstream[fe_id];
+ audio_effects = pdata->audio_effects[fe_id];
+ if (!cstream || !audio_effects) {
+ pr_err("%s: stream or effects inactive\n", __func__);
+ return -EINVAL;
+ }
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: cannot set audio effects\n", __func__);
+ return -EINVAL;
+ }
+
+ return 0;
+}
+
+static int msm_compr_query_audio_effect_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct msm_compr_audio_effects *audio_effects = NULL;
+ struct snd_compr_stream *cstream = NULL;
+ struct msm_compr_audio *prtd = NULL;
+ long *values = &(ucontrol->value.integer.value[0]);
+
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bounds fe_id %lu\n",
+ __func__, fe_id);
+ return -EINVAL;
+ }
+ cstream = pdata->cstream[fe_id];
+ audio_effects = pdata->audio_effects[fe_id];
+ if (!cstream || !audio_effects) {
+ pr_err("%s: stream or effects inactive\n", __func__);
+ return -EINVAL;
+ }
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: cannot set audio effects\n", __func__);
+ return -EINVAL;
+ }
+ if (prtd->compr_passthr != LEGACY_PCM) {
+ pr_err("%s: No effects for compr_type[%d]\n",
+ __func__, prtd->compr_passthr);
+ return -EPERM;
+ }
+ audio_effects->query.mod_id = (u32)*values++;
+ audio_effects->query.parm_id = (u32)*values++;
+ audio_effects->query.size = (u32)*values++;
+ audio_effects->query.offset = (u32)*values++;
+ audio_effects->query.device = (u32)*values++;
+ return 0;
+}
+
+static int msm_compr_query_audio_effect_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct msm_compr_audio_effects *audio_effects = NULL;
+ struct snd_compr_stream *cstream = NULL;
+ struct msm_compr_audio *prtd = NULL;
+ long *values = &(ucontrol->value.integer.value[0]);
+
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bounds fe_id %lu\n",
+ __func__, fe_id);
+ return -EINVAL;
+ }
+ cstream = pdata->cstream[fe_id];
+ audio_effects = pdata->audio_effects[fe_id];
+ if (!cstream || !audio_effects) {
+ pr_debug("%s: stream or effects inactive\n", __func__);
+ return -EINVAL;
+ }
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: cannot set audio effects\n", __func__);
+ return -EINVAL;
+ }
+ values[0] = (long)audio_effects->query.mod_id;
+ values[1] = (long)audio_effects->query.parm_id;
+ values[2] = (long)audio_effects->query.size;
+ values[3] = (long)audio_effects->query.offset;
+ values[4] = (long)audio_effects->query.device;
+ return 0;
+}
+
+static int msm_compr_send_dec_params(struct snd_compr_stream *cstream,
+ struct msm_compr_dec_params *dec_params,
+ int stream_id)
+{
+
+ int rc = 0;
+ struct msm_compr_audio *prtd = NULL;
+ struct snd_dec_ddp *ddp = &dec_params->ddp_params;
+
+ if (!cstream || !dec_params) {
+ pr_err("%s: stream or dec_params inactive\n", __func__);
+ rc = -EINVAL;
+ goto end;
+ }
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: cannot set dec_params\n", __func__);
+ rc = -EINVAL;
+ goto end;
+ }
+ switch (prtd->codec) {
+ case FORMAT_MP3:
+ case FORMAT_MPEG4_AAC:
+ case FORMAT_TRUEHD:
+ case FORMAT_IEC61937:
+ case FORMAT_APTX:
+ pr_debug("%s: no runtime parameters for codec: %d\n", __func__,
+ prtd->codec);
+ break;
+ case FORMAT_AC3:
+ case FORMAT_EAC3:
+ if (prtd->compr_passthr != LEGACY_PCM) {
+ pr_debug("%s: No DDP param for compr_type[%d]\n",
+ __func__, prtd->compr_passthr);
+ break;
+ }
+ rc = msm_compr_send_ddp_cfg(prtd->audio_client, ddp, stream_id);
+ if (rc < 0)
+ pr_err("%s: DDP CMD CFG failed %d\n", __func__, rc);
+ break;
+ default:
+ break;
+ }
+end:
+ return rc;
+
+}
+static int msm_compr_dec_params_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct msm_compr_dec_params *dec_params = NULL;
+ struct snd_compr_stream *cstream = NULL;
+ struct msm_compr_audio *prtd = NULL;
+ long *values = &(ucontrol->value.integer.value[0]);
+ int rc = 0;
+
+ pr_debug("%s\n", __func__);
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bounds fe_id %lu\n",
+ __func__, fe_id);
+ rc = -EINVAL;
+ goto end;
+ }
+
+ cstream = pdata->cstream[fe_id];
+ dec_params = pdata->dec_params[fe_id];
+
+ if (!cstream || !dec_params) {
+ pr_err("%s: stream or dec_params inactive\n", __func__);
+ rc = -EINVAL;
+ goto end;
+ }
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: cannot set dec_params\n", __func__);
+ rc = -EINVAL;
+ goto end;
+ }
+
+ switch (prtd->codec) {
+ case FORMAT_MP3:
+ case FORMAT_MPEG4_AAC:
+ case FORMAT_FLAC:
+ case FORMAT_VORBIS:
+ case FORMAT_ALAC:
+ case FORMAT_APE:
+ case FORMAT_DTS:
+ case FORMAT_DSD:
+ case FORMAT_TRUEHD:
+ case FORMAT_IEC61937:
+ case FORMAT_APTX:
+ pr_debug("%s: no runtime parameters for codec: %d\n", __func__,
+ prtd->codec);
+ break;
+ case FORMAT_AC3:
+ case FORMAT_EAC3: {
+ struct snd_dec_ddp *ddp = &dec_params->ddp_params;
+ int cnt;
+
+ if (prtd->compr_passthr != LEGACY_PCM) {
+ pr_debug("%s: No DDP param for compr_type[%d]\n",
+ __func__, prtd->compr_passthr);
+ break;
+ }
+
+ ddp->params_length = (*values++);
+ if (ddp->params_length > DDP_DEC_MAX_NUM_PARAM) {
+ pr_err("%s: invalid num of params:: %d\n", __func__,
+ ddp->params_length);
+ rc = -EINVAL;
+ goto end;
+ }
+ for (cnt = 0; cnt < ddp->params_length; cnt++) {
+ ddp->params_id[cnt] = *values++;
+ ddp->params_value[cnt] = *values++;
+ }
+ prtd = cstream->runtime->private_data;
+ if (prtd && prtd->audio_client)
+ rc = msm_compr_send_dec_params(cstream, dec_params,
+ prtd->audio_client->stream_id);
+ break;
+ }
+ default:
+ break;
+ }
+end:
+ pr_debug("%s: ret %d\n", __func__, rc);
+ return rc;
+}
+
+static int msm_compr_dec_params_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ /* dummy function */
+ return 0;
+}
+
+static int msm_compr_playback_app_type_cfg_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_compr_playback_app_type_cfg_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_compr_capture_app_type_cfg_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_compr_capture_app_type_cfg_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_compr_channel_map_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ u64 fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ int rc = 0, i;
+
+ pr_debug("%s: fe_id- %llu\n", __func__, fe_id);
+
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bounds fe_id %llu\n",
+ __func__, fe_id);
+ rc = -EINVAL;
+ goto end;
+ }
+
+ if (pdata->ch_map[fe_id]) {
+ pdata->ch_map[fe_id]->set_ch_map = true;
+ for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++)
+ pdata->ch_map[fe_id]->channel_map[i] =
+ (char)(ucontrol->value.integer.value[i]);
+ } else {
+ pr_debug("%s: no memory for ch_map, default will be set\n",
+ __func__);
+ }
+end:
+ pr_debug("%s: ret %d\n", __func__, rc);
+ return rc;
+}
+
+static int msm_compr_channel_map_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ u64 fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ int rc = 0, i;
+
+ pr_debug("%s: fe_id- %llu\n", __func__, fe_id);
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s: Received out of bounds fe_id %llu\n",
+ __func__, fe_id);
+ rc = -EINVAL;
+ goto end;
+ }
+ if (pdata->ch_map[fe_id]) {
+ for (i = 0; i < PCM_FORMAT_MAX_NUM_CHANNEL; i++)
+ ucontrol->value.integer.value[i] =
+ pdata->ch_map[fe_id]->channel_map[i];
+ }
+end:
+ pr_debug("%s: ret %d\n", __func__, rc);
+ return rc;
+}
+
+static int msm_compr_adsp_stream_cmd_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct snd_compr_stream *cstream = NULL;
+ struct msm_compr_audio *prtd;
+ int ret = 0;
+ struct msm_adsp_event_data *event_data = NULL;
+
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received invalid fe_id %lu\n",
+ __func__, fe_id);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ cstream = pdata->cstream[fe_id];
+ if (cstream == NULL) {
+ pr_err("%s cstream is null\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: prtd is null\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if (prtd->audio_client == NULL) {
+ pr_err("%s: audio_client 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_compr_ion_fd_map_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct snd_compr_stream *cstream = NULL;
+ struct msm_compr_audio *prtd;
+ int fd;
+ int ret = 0;
+
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received out of bounds invalid fe_id %lu\n",
+ __func__, fe_id);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ cstream = pdata->cstream[fe_id];
+ if (cstream == NULL) {
+ pr_err("%s cstream is null\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: prtd is null\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if (prtd->audio_client == NULL) {
+ pr_err("%s: audio_client is null\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ memcpy(&fd, ucontrol->value.bytes.data, sizeof(fd));
+ ret = q6asm_send_ion_fd(prtd->audio_client, fd);
+ if (ret < 0)
+ pr_err("%s: failed to register ion fd\n", __func__);
+done:
+ return ret;
+}
+
+static int msm_compr_rtic_event_ack_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ unsigned long fe_id = kcontrol->private_value;
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ struct snd_compr_stream *cstream = NULL;
+ struct msm_compr_audio *prtd;
+ int ret = 0;
+ int param_length = 0;
+
+ if (fe_id >= MSM_FRONTEND_DAI_MAX) {
+ pr_err("%s Received invalid fe_id %lu\n",
+ __func__, fe_id);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ cstream = pdata->cstream[fe_id];
+ if (cstream == NULL) {
+ pr_err("%s cstream is null\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ prtd = cstream->runtime->private_data;
+ if (!prtd) {
+ pr_err("%s: prtd is null\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ if (prtd->audio_client == NULL) {
+ pr_err("%s: audio_client is null\n", __func__);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ memcpy(¶m_length, ucontrol->value.bytes.data,
+ sizeof(param_length));
+ if ((param_length + sizeof(param_length))
+ >= sizeof(ucontrol->value.bytes.data)) {
+ pr_err("%s param length=%d exceeds limit",
+ __func__, param_length);
+ ret = -EINVAL;
+ goto done;
+ }
+
+ ret = q6asm_send_rtic_event_ack(prtd->audio_client,
+ ucontrol->value.bytes.data + sizeof(param_length),
+ param_length);
+ if (ret < 0)
+ pr_err("%s: failed to send rtic event ack, err = %d\n",
+ __func__, ret);
+done:
+ return ret;
+}
+
+static int msm_compr_gapless_put(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ struct msm_compr_pdata *pdata = (struct msm_compr_pdata *)
+ snd_soc_component_get_drvdata(comp);
+ pdata->use_dsp_gapless_mode = ucontrol->value.integer.value[0];
+ pr_debug("%s: value: %ld\n", __func__,
+ ucontrol->value.integer.value[0]);
+
+ return 0;
+}
+
+static int msm_compr_gapless_get(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_value *ucontrol)
+{
+ struct snd_soc_component *comp = snd_kcontrol_chip(kcontrol);
+ struct msm_compr_pdata *pdata =
+ snd_soc_component_get_drvdata(comp);
+ pr_debug("%s:gapless mode %d\n", __func__, pdata->use_dsp_gapless_mode);
+ ucontrol->value.integer.value[0] = pdata->use_dsp_gapless_mode;
+
+ return 0;
+}
+
+static const struct snd_kcontrol_new msm_compr_gapless_controls[] = {
+ SOC_SINGLE_EXT("Compress Gapless Playback",
+ 0, 0, 1, 0,
+ msm_compr_gapless_get,
+ msm_compr_gapless_put),
+};
+
+static int msm_compr_probe(struct snd_soc_platform *platform)
+{
+ struct msm_compr_pdata *pdata;
+ int i;
+ int rc;
+ const char *qdsp_version;
+
+ pr_debug("%s\n", __func__);
+ pdata = (struct msm_compr_pdata *)
+ kzalloc(sizeof(*pdata), GFP_KERNEL);
+ if (!pdata)
+ return -ENOMEM;
+
+ snd_soc_platform_set_drvdata(platform, pdata);
+
+ for (i = 0; i < MSM_FRONTEND_DAI_MAX; i++) {
+ pdata->volume[i][0] = COMPRESSED_LR_VOL_MAX_STEPS;
+ pdata->volume[i][1] = COMPRESSED_LR_VOL_MAX_STEPS;
+ pdata->audio_effects[i] = NULL;
+ pdata->dec_params[i] = NULL;
+ pdata->cstream[i] = NULL;
+ pdata->ch_map[i] = NULL;
+ }
+
+ snd_soc_add_platform_controls(platform, msm_compr_gapless_controls,
+ ARRAY_SIZE(msm_compr_gapless_controls));
+
+ rc = of_property_read_string(platform->dev->of_node,
+ "qcom,adsp-version", &qdsp_version);
+ if (!rc) {
+ if (!strcmp(qdsp_version, "MDSP 1.2"))
+ pdata->use_legacy_api = true;
+ else
+ pdata->use_legacy_api = false;
+ } else
+ pdata->use_legacy_api = false;
+
+ pr_debug("%s: use legacy api %d\n", __func__, pdata->use_legacy_api);
+ /*
+ * use_dsp_gapless_mode part of platform data(pdata) is updated from HAL
+ * through a mixer control before compress driver is opened. The mixer
+ * control is used to decide if dsp gapless mode needs to be enabled.
+ * Gapless is disabled by default.
+ */
+ pdata->use_dsp_gapless_mode = false;
+ return 0;
+}
+
+static int msm_compr_volume_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 2;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = COMPRESSED_LR_VOL_MAX_STEPS;
+ return 0;
+}
+
+static int msm_compr_audio_effects_config_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = MAX_PP_PARAMS_SZ;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xFFFFFFFF;
+ return 0;
+}
+
+static int msm_compr_query_audio_effect_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 128;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xFFFFFFFF;
+ return 0;
+}
+
+static int msm_compr_dec_params_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 128;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xFFFFFFFF;
+ return 0;
+}
+
+static int msm_compr_app_type_cfg_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 5;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xFFFFFFFF;
+ return 0;
+}
+
+static int msm_compr_channel_map_info(struct snd_kcontrol *kcontrol,
+ struct snd_ctl_elem_info *uinfo)
+{
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+ uinfo->count = 8;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 0xFFFFFFFF;
+ return 0;
+}
+
+static int msm_compr_add_volume_control(struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = "Compress Playback";
+ const char *deviceNo = "NN";
+ const char *suffix = "Volume";
+ char *mixer_str = NULL;
+ int ctl_len;
+ struct snd_kcontrol_new fe_volume_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_TLV_READ |
+ SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_compr_volume_info,
+ .tlv.p = msm_compr_vol_gain,
+ .get = msm_compr_volume_get,
+ .put = msm_compr_volume_put,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s NULL rtd\n", __func__);
+ return 0;
+ }
+ pr_debug("%s: added new compr 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 +
+ strlen(suffix) + 1;
+ mixer_str = kzalloc(ctl_len, GFP_KERNEL);
+ if (!mixer_str) {
+ pr_err("failed to allocate mixer ctrl str of len %d", ctl_len);
+ return 0;
+ }
+ snprintf(mixer_str, ctl_len, "%s %d %s", mixer_ctl_name,
+ rtd->pcm->device, suffix);
+ fe_volume_control[0].name = mixer_str;
+ fe_volume_control[0].private_value = rtd->dai_link->id;
+ pr_debug("Registering new mixer ctl %s", mixer_str);
+ snd_soc_add_platform_controls(rtd->platform, fe_volume_control,
+ ARRAY_SIZE(fe_volume_control));
+ kfree(mixer_str);
+ return 0;
+}
+
+static int msm_compr_add_audio_effects_control(struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = "Audio Effects Config";
+ const char *deviceNo = "NN";
+ char *mixer_str = NULL;
+ int ctl_len;
+ struct snd_kcontrol_new fe_audio_effects_config_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_compr_audio_effects_config_info,
+ .get = msm_compr_audio_effects_config_get,
+ .put = msm_compr_audio_effects_config_put,
+ .private_value = 0,
+ }
+ };
+
+
+ if (!rtd) {
+ pr_err("%s NULL rtd\n", __func__);
+ return 0;
+ }
+
+ pr_debug("%s: added new compr 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)
+ return 0;
+
+ snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device);
+
+ fe_audio_effects_config_control[0].name = mixer_str;
+ fe_audio_effects_config_control[0].private_value = rtd->dai_link->id;
+ pr_debug("Registering new mixer ctl %s\n", mixer_str);
+ snd_soc_add_platform_controls(rtd->platform,
+ fe_audio_effects_config_control,
+ ARRAY_SIZE(fe_audio_effects_config_control));
+ kfree(mixer_str);
+ return 0;
+}
+
+static int msm_compr_add_query_audio_effect_control(
+ struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = "Query Audio Effect Param";
+ const char *deviceNo = "NN";
+ char *mixer_str = NULL;
+ int ctl_len;
+ struct snd_kcontrol_new fe_query_audio_effect_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_compr_query_audio_effect_info,
+ .get = msm_compr_query_audio_effect_get,
+ .put = msm_compr_query_audio_effect_put,
+ .private_value = 0,
+ }
+ };
+ if (!rtd) {
+ pr_err("%s NULL rtd\n", __func__);
+ return 0;
+ }
+ pr_debug("%s: added new compr 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) {
+ pr_err("failed to allocate mixer ctrl str of len %d", ctl_len);
+ return 0;
+ }
+ snprintf(mixer_str, ctl_len, "%s %d", mixer_ctl_name, rtd->pcm->device);
+ fe_query_audio_effect_control[0].name = mixer_str;
+ fe_query_audio_effect_control[0].private_value = rtd->dai_link->id;
+ pr_debug("%s: registering new mixer ctl %s\n", __func__, mixer_str);
+ snd_soc_add_platform_controls(rtd->platform,
+ fe_query_audio_effect_control,
+ ARRAY_SIZE(fe_query_audio_effect_control));
+ kfree(mixer_str);
+ return 0;
+}
+
+static int msm_compr_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_compr_adsp_stream_cmd_put,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s NULL rtd\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("%s: Registering new mixer ctl %s\n", __func__, 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 to add ctl %s. err = %d\n",
+ __func__, mixer_str, ret);
+
+ kfree(mixer_str);
+done:
+ return ret;
+}
+
+static int msm_compr_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: 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_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_compr_add_dec_runtime_params_control(
+ struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = "Audio Stream";
+ const char *deviceNo = "NN";
+ const char *suffix = "Dec Params";
+ char *mixer_str = NULL;
+ int ctl_len;
+ struct snd_kcontrol_new fe_dec_params_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_compr_dec_params_info,
+ .get = msm_compr_dec_params_get,
+ .put = msm_compr_dec_params_put,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s NULL rtd\n", __func__);
+ return 0;
+ }
+
+ pr_debug("%s: added new compr 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 +
+ strlen(suffix) + 1;
+ mixer_str = kzalloc(ctl_len, GFP_KERNEL);
+
+ if (!mixer_str)
+ return 0;
+
+ snprintf(mixer_str, ctl_len, "%s %d %s", mixer_ctl_name,
+ rtd->pcm->device, suffix);
+
+ fe_dec_params_control[0].name = mixer_str;
+ fe_dec_params_control[0].private_value = rtd->dai_link->id;
+ pr_debug("Registering new mixer ctl %s", mixer_str);
+ snd_soc_add_platform_controls(rtd->platform,
+ fe_dec_params_control,
+ ARRAY_SIZE(fe_dec_params_control));
+ kfree(mixer_str);
+ return 0;
+}
+
+static int msm_compr_add_app_type_cfg_control(struct snd_soc_pcm_runtime *rtd)
+{
+ 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";
+ char *mixer_str = NULL;
+ int ctl_len;
+ struct snd_kcontrol_new fe_app_type_cfg_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_compr_app_type_cfg_info,
+ .put = msm_compr_playback_app_type_cfg_put,
+ .get = msm_compr_playback_app_type_cfg_get,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s NULL rtd\n", __func__);
+ return 0;
+ }
+
+ pr_debug("%s: added new compr FE ctl 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);
+ if (rtd->compr->direction == SND_COMPRESS_PLAYBACK)
+ ctl_len = strlen(playback_mixer_ctl_name) + 1 + strlen(deviceNo)
+ + 1 + strlen(suffix) + 1;
+ else
+ ctl_len = strlen(capture_mixer_ctl_name) + 1 + strlen(deviceNo)
+ + 1 + strlen(suffix) + 1;
+
+ mixer_str = kzalloc(ctl_len, GFP_KERNEL);
+
+ if (!mixer_str)
+ return 0;
+
+ if (rtd->compr->direction == SND_COMPRESS_PLAYBACK)
+ snprintf(mixer_str, ctl_len, "%s %d %s",
+ playback_mixer_ctl_name, rtd->pcm->device, suffix);
+ else
+ snprintf(mixer_str, ctl_len, "%s %d %s",
+ capture_mixer_ctl_name, rtd->pcm->device, suffix);
+
+ fe_app_type_cfg_control[0].name = mixer_str;
+ fe_app_type_cfg_control[0].private_value = rtd->dai_link->id;
+
+ if (rtd->compr->direction == SND_COMPRESS_PLAYBACK) {
+ fe_app_type_cfg_control[0].put =
+ msm_compr_playback_app_type_cfg_put;
+ fe_app_type_cfg_control[0].get =
+ msm_compr_playback_app_type_cfg_get;
+ } else {
+ fe_app_type_cfg_control[0].put =
+ msm_compr_capture_app_type_cfg_put;
+ fe_app_type_cfg_control[0].get =
+ msm_compr_capture_app_type_cfg_get;
+ }
+ pr_debug("Registering new mixer ctl %s", mixer_str);
+ snd_soc_add_platform_controls(rtd->platform,
+ fe_app_type_cfg_control,
+ ARRAY_SIZE(fe_app_type_cfg_control));
+ kfree(mixer_str);
+ return 0;
+}
+
+static int msm_compr_add_channel_map_control(struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = "Playback Channel Map";
+ const char *deviceNo = "NN";
+ char *mixer_str = NULL;
+ struct msm_compr_pdata *pdata = NULL;
+ int ctl_len;
+ struct snd_kcontrol_new fe_channel_map_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_compr_channel_map_info,
+ .get = msm_compr_channel_map_get,
+ .put = msm_compr_channel_map_put,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s: NULL rtd\n", __func__);
+ return -EINVAL;
+ }
+
+ pr_debug("%s: added new compr 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) + strlen(deviceNo) + 1;
+ mixer_str = kzalloc(ctl_len, GFP_KERNEL);
+
+ if (!mixer_str)
+ return -ENOMEM;
+
+ snprintf(mixer_str, ctl_len, "%s%d", mixer_ctl_name, rtd->pcm->device);
+
+ fe_channel_map_control[0].name = mixer_str;
+ fe_channel_map_control[0].private_value = rtd->dai_link->id;
+ pr_debug("%s: Registering new mixer ctl %s\n", __func__, mixer_str);
+ snd_soc_add_platform_controls(rtd->platform,
+ fe_channel_map_control,
+ ARRAY_SIZE(fe_channel_map_control));
+
+ pdata = snd_soc_platform_get_drvdata(rtd->platform);
+ pdata->ch_map[rtd->dai_link->id] =
+ kzalloc(sizeof(struct msm_compr_ch_map), GFP_KERNEL);
+ if (!pdata->ch_map[rtd->dai_link->id]) {
+ pr_err("%s: Could not allocate memory for channel map\n",
+ __func__);
+ kfree(mixer_str);
+ return -ENOMEM;
+ }
+ kfree(mixer_str);
+ return 0;
+}
+
+static int msm_compr_add_io_fd_cmd_control(struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = "Playback ION FD";
+ const char *deviceNo = "NN";
+ char *mixer_str = NULL;
+ int ctl_len = 0, ret = 0;
+ struct snd_kcontrol_new fe_ion_fd_config_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_adsp_stream_cmd_info,
+ .put = msm_compr_ion_fd_map_put,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s NULL rtd\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_ion_fd_config_control[0].name = mixer_str;
+ fe_ion_fd_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_ion_fd_config_control,
+ ARRAY_SIZE(fe_ion_fd_config_control));
+ if (ret < 0)
+ pr_err("%s: failed to add ctl %s\n", __func__, mixer_str);
+
+ kfree(mixer_str);
+done:
+ return ret;
+}
+
+static int msm_compr_add_event_ack_cmd_control(struct snd_soc_pcm_runtime *rtd)
+{
+ const char *mixer_ctl_name = "Playback Event Ack";
+ const char *deviceNo = "NN";
+ char *mixer_str = NULL;
+ int ctl_len = 0, ret = 0;
+ struct snd_kcontrol_new fe_event_ack_config_control[1] = {
+ {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "?",
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .info = msm_adsp_stream_cmd_info,
+ .put = msm_compr_rtic_event_ack_put,
+ .private_value = 0,
+ }
+ };
+
+ if (!rtd) {
+ pr_err("%s NULL rtd\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_event_ack_config_control[0].name = mixer_str;
+ fe_event_ack_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_event_ack_config_control,
+ ARRAY_SIZE(fe_event_ack_config_control));
+ if (ret < 0)
+ pr_err("%s: failed to add ctl %s\n", __func__, mixer_str);
+
+ kfree(mixer_str);
+done:
+ return ret;
+}
+
+static int msm_compr_new(struct snd_soc_pcm_runtime *rtd)
+{
+ int rc;
+
+ rc = msm_compr_add_volume_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr Volume Control\n", __func__);
+
+ rc = msm_compr_add_audio_effects_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr Audio Effects Control\n",
+ __func__);
+
+ rc = msm_compr_add_audio_adsp_stream_cmd_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr ADSP Stream Cmd Control\n",
+ __func__);
+
+ rc = msm_compr_add_audio_adsp_stream_callback_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr ADSP Stream Callback Control\n",
+ __func__);
+
+ rc = msm_compr_add_io_fd_cmd_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr ion fd Control\n",
+ __func__);
+
+ rc = msm_compr_add_event_ack_cmd_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr event ack Control\n",
+ __func__);
+
+ rc = msm_compr_add_query_audio_effect_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr Query Audio Effect Control\n",
+ __func__);
+
+ rc = msm_compr_add_dec_runtime_params_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr Dec runtime params Control\n",
+ __func__);
+ rc = msm_compr_add_app_type_cfg_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr App Type Cfg Control\n",
+ __func__);
+ rc = msm_compr_add_channel_map_control(rtd);
+ if (rc)
+ pr_err("%s: Could not add Compr Channel Map Control\n",
+ __func__);
+ return 0;
+}
+
+static struct snd_compr_ops msm_compr_ops = {
+ .open = msm_compr_open,
+ .free = msm_compr_free,
+ .trigger = msm_compr_trigger,
+ .pointer = msm_compr_pointer,
+ .set_params = msm_compr_set_params,
+ .set_metadata = msm_compr_set_metadata,
+ .get_metadata = msm_compr_get_metadata,
+ .set_next_track_param = msm_compr_set_next_track_param,
+ .ack = msm_compr_ack,
+ .copy = msm_compr_copy,
+ .get_caps = msm_compr_get_caps,
+ .get_codec_caps = msm_compr_get_codec_caps,
+};
+
+static struct snd_soc_platform_driver msm_soc_platform = {
+ .probe = msm_compr_probe,
+ .compr_ops = &msm_compr_ops,
+ .pcm_new = msm_compr_new,
+};
+
+static int msm_compr_dev_probe(struct platform_device *pdev)
+{
+
+ pr_debug("%s: dev name %s\n", __func__, dev_name(&pdev->dev));
+ 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-compress-dsp"},
+ {}
+};
+MODULE_DEVICE_TABLE(of, msm_compr_dt_match);
+
+static struct platform_driver msm_compr_driver = {
+ .driver = {
+ .name = "msm-compress-dsp",
+ .owner = THIS_MODULE,
+ .of_match_table = msm_compr_dt_match,
+ },
+ .probe = msm_compr_dev_probe,
+ .remove = msm_compr_remove,
+};
+
+static int __init msm_soc_platform_init(void)
+{
+ 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("Compress Offload platform driver");
+MODULE_LICENSE("GPL v2");