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kernel-fxtec-pro1x/asoc/msm-compress-q6-v2.c
Ralf Herz 2257355fe4 asoc: Support compressed capture from AFE interfaces 
Support compressed IEC61937 audio capture from AFE interfaces
with variable data packet lengths.

Change-Id: Iffd5a06ed5fe1a96f34e225d54fab660b21d2a48
Signed-off-by: Ralf Herz <rherz@codeaurora.org>
2018-07-25 21:35:45 +05:30

4616 lines
132 KiB
C
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/* Copyright (c) 2012-2018, 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/pcm_params.h>
#include <sound/audio_effects.h>
#include <asm/dma.h>
#include <linux/dma-mapping.h>
#include <linux/msm_audio.h>
#include <sound/timer.h>
#include <sound/tlv.h>
#include <sound/compress_params.h>
#include <sound/compress_offload.h>
#include <sound/compress_driver.h>
#include <dsp/msm_audio_ion.h>
#include <dsp/apr_audio-v2.h>
#include <dsp/q6asm-v2.h>
#include <dsp/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];
bool is_in_use[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;
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, &param) < 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(&param, 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, &param);
if (ret < 0) {
pr_err("%s: q6asm_async_read failed - %d\n",
__func__, ret);
return ret;
}
prtd->bytes_read += buffer_length + prtd->ts_header_offset;
prtd->bytes_read_offset += buffer_length + prtd->ts_header_offset;
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 actual length of the received buffer */
*buff_addr = payload[4];
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,
};
struct snd_kcontrol *kctl;
struct snd_ctl_elem_value kctl_elem_value;
uint16_t target_asm_bit_width = 0;
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;
}
kctl = snd_soc_card_get_kcontrol(soc_prtd->card,
DSP_BIT_WIDTH_MIXER_CTL);
if (kctl) {
kctl->get(kctl, &kctl_elem_value);
target_asm_bit_width = kctl_elem_value.value.integer.value[0];
if (target_asm_bit_width > 0) {
pr_debug("%s enforce ASM bitwidth to %d from %d\n",
__func__,
target_asm_bit_width,
bits_per_sample);
bits_per_sample = target_asm_bit_width;
}
} else {
pr_info("%s: failed to get mixer ctl for %s.\n",
__func__, DSP_BIT_WIDTH_MIXER_CTL);
}
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;
uint32_t enc_cfg_id = ENC_CFG_ID_NONE;
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;
if (prtd->codec == FORMAT_BESPOKE)
enc_cfg_id =
prtd->codec_param.codec.options.generic.reserved[0];
break;
}
pr_debug("%s: stream_id %d bits_per_sample %d compr_passthr %d\n",
__func__, ac->stream_id, bits_per_sample,
prtd->compr_passthr);
if (prtd->compr_passthr != LEGACY_PCM) {
ret = q6asm_open_read_compressed(prtd->audio_client,
prtd->codec, prtd->compr_passthr);
if (ret < 0) {
pr_err("%s:ASM open read err[%d] for compr_type[%d]\n",
__func__, ret, prtd->compr_passthr);
return ret;
}
ret = msm_pcm_routing_reg_phy_compr_stream(
soc_prtd->dai_link->id,
ac->perf_mode,
prtd->session_id,
SNDRV_PCM_STREAM_CAPTURE,
prtd->compr_passthr);
if (ret) {
pr_err("%s: compr stream reg failed:%d\n",
__func__, ret);
return ret;
}
} else {
if (prtd->codec_param.codec.flags & COMPRESSED_TIMESTAMP_FLAG) {
ret = q6asm_open_read_v4(prtd->audio_client,
prtd->codec,
bits_per_sample, true, enc_cfg_id);
} else {
ret = q6asm_open_read_v4(prtd->audio_client,
prtd->codec,
bits_per_sample, false, enc_cfg_id);
}
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);
if (prtd->codec == FORMAT_BESPOKE) {
/*
* For BESPOKE codec, encoder specific config params are
* included as part of generic.
*/
ret = q6asm_enc_cfg_blk_custom(prtd->audio_client, prtd->sample_rate,
prtd->num_channels, prtd->codec,
(void *)&prtd->codec_param.codec.options.generic);
} else if (prtd->compr_passthr == LEGACY_PCM) {
ret = q6asm_enc_cfg_blk_pcm_format_support_v4(prtd->audio_client,
prtd->sample_rate, prtd->num_channels,
bits_per_sample, sample_word_size,
ASM_LITTLE_ENDIAN, DEFAULT_QF);
}
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 = NULL;
struct msm_compr_pdata *pdata =
snd_soc_platform_get_drvdata(rtd->platform);
pr_debug("%s\n", __func__);
if (pdata->is_in_use[rtd->dai_link->id] == true) {
pr_err("%s: %s is already in use, err: %d\n",
__func__, rtd->dai_link->cpu_dai_name, -EBUSY);
return -EBUSY;
}
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] == NULL) {
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] == NULL) {
pr_err("%s: Could not allocate memory for dec params\n",
__func__);
kfree(pdata->audio_effects[rtd->dai_link->id]);
pdata->audio_effects[rtd->dai_link->id] = NULL;
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 == NULL) {
pr_err("%s: Could not allocate memory for client\n", __func__);
kfree(pdata->audio_effects[rtd->dai_link->id]);
pdata->audio_effects[rtd->dai_link->id] = NULL;
kfree(pdata->dec_params[rtd->dai_link->id]);
pdata->dec_params[rtd->dai_link->id] = NULL;
pdata->cstream[rtd->dai_link->id] = NULL;
kfree(prtd);
runtime->private_data = NULL;
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);
pdata->is_in_use[rtd->dai_link->id] = true;
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);
if (pdata->audio_effects[soc_prtd->dai_link->id] != NULL) {
kfree(pdata->audio_effects[soc_prtd->dai_link->id]);
pdata->audio_effects[soc_prtd->dai_link->id] = NULL;
}
if (pdata->dec_params[soc_prtd->dai_link->id] != NULL) {
kfree(pdata->dec_params[soc_prtd->dai_link->id]);
pdata->dec_params[soc_prtd->dai_link->id] = NULL;
}
pdata->is_in_use[soc_prtd->dai_link->id] = false;
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;
}
case SND_AUDIOCODEC_BESPOKE: {
pr_debug("%s: SND_AUDIOCODEC_BESPOKE\n", __func__);
prtd->codec = FORMAT_BESPOKE;
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_ratelimited("%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 (event_data->payload_len > sizeof(ucontrol->value.bytes.data)
- sizeof(struct msm_adsp_event_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(&param_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;
pdata->is_in_use[i] = false;
}
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,
};
int __init msm_compress_dsp_init(void)
{
return platform_driver_register(&msm_compr_driver);
}
void msm_compress_dsp_exit(void)
{
platform_driver_unregister(&msm_compr_driver);
}
MODULE_DESCRIPTION("Compress Offload platform driver");
MODULE_LICENSE("GPL v2");
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