evie/android_kernel_oneplus_msm8998
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ASoC: msm: add q6asm api for transcode loopback support

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Transcode loopback api can be used by client to configure
compress loopback.

CRs-fixed: 2049714
Change-Id: I7286fbdfa89375d517f3dad7d04e1d2a360d6f7e
Signed-off-by: Siddartha Shaik <sshaik@codeaurora.org>
This commit is contained in:
Siddartha Shaik 2017-05-19 12:37:00 +05:30 • committed by Gerrit - the friendly Code Review server
parent 60be71604a
commit 19fc81edfe
2 changed files with 75 additions and 0 deletions
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4
include/sound/q6asm-v2.h
View file

@ -318,6 +318,10 @@ int q6asm_open_read_write_v2(struct audio_client *ac, uint32_t rd_format,
int q6asm_open_loopback_v2(struct audio_client *ac, int q6asm_open_loopback_v2(struct audio_client *ac,
uint16_t bits_per_sample); uint16_t bits_per_sample);
int q6asm_open_transcode_loopback(struct audio_client *ac,
uint16_t bits_per_sample, uint32_t source_format,
uint32_t sink_format);
int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts, int q6asm_write(struct audio_client *ac, uint32_t len, uint32_t msw_ts,
uint32_t lsw_ts, uint32_t flags); uint32_t lsw_ts, uint32_t flags);
int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts, int q6asm_write_nolock(struct audio_client *ac, uint32_t len, uint32_t msw_ts,

71
sound/soc/msm/qdsp6v2/q6asm.c
View file

@ -3225,6 +3225,77 @@ fail_cmd:
return rc; return rc;
} }
int q6asm_open_transcode_loopback(struct audio_client *ac,
uint16_t bits_per_sample,
uint32_t source_format, uint32_t sink_format)
{
int rc = 0x00;
struct asm_stream_cmd_open_transcode_loopback_t open;
if (ac == NULL) {
pr_err("%s: APR handle NULL\n", __func__);
return -EINVAL;
}
if (ac->apr == NULL) {
pr_err("%s: AC APR handle NULL\n", __func__);
return -EINVAL;
}
pr_debug("%s: session[%d]\n", __func__, ac->session);
q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE);
atomic_set(&ac->cmd_state, -1);
open.hdr.opcode = ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK;
open.mode_flags = 0;
open.src_endpoint_type = 0;
open.sink_endpoint_type = 0;
open.src_format_id = source_format;
open.sink_format_id = sink_format;
/* source endpoint : matrix */
open.audproc_topo_id = q6asm_get_asm_topology_cal();
ac->app_type = q6asm_get_asm_app_type_cal();
if (ac->perf_mode == LOW_LATENCY_PCM_MODE)
open.mode_flags |= ASM_LOW_LATENCY_STREAM_SESSION;
else
open.mode_flags |= ASM_LEGACY_STREAM_SESSION;
ac->topology = open.audproc_topo_id;
open.bits_per_sample = bits_per_sample;
open.reserved = 0;
pr_debug("%s: opening a transcode_loopback with mode_flags =[%d] session[%d]\n",
__func__, open.mode_flags, ac->session);
rc = apr_send_pkt(ac->apr, (uint32_t *) &open);
if (rc < 0) {
pr_err("%s: open failed op[0x%x]rc[%d]\n",
__func__, open.hdr.opcode, rc);
rc = -EINVAL;
goto fail_cmd;
}
rc = wait_event_timeout(ac->cmd_wait,
(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
if (!rc) {
pr_err("%s: timeout. waited for open_transcode_loopback\n",
__func__);
rc = -ETIMEDOUT;
goto fail_cmd;
}
if (atomic_read(&ac->cmd_state) > 0) {
pr_err("%s: DSP returned error[%s]\n",
__func__, adsp_err_get_err_str(
atomic_read(&ac->cmd_state)));
rc = adsp_err_get_lnx_err_code(
atomic_read(&ac->cmd_state));
goto fail_cmd;
}
return 0;
fail_cmd:
return rc;
}
static static
int q6asm_set_shared_circ_buff(struct audio_client *ac, int q6asm_set_shared_circ_buff(struct audio_client *ac,
struct asm_stream_cmd_open_shared_io *open, struct asm_stream_cmd_open_shared_io *open,