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Commit a633dbc5 authored by Siddartha Shaik's avatar Siddartha Shaik Committed by Banajit Goswami
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ASoC: msm: add q6asm api for transcode loopback support 



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

CRs-fixed: 2049714
Change-Id: I7286fbdfa89375d517f3dad7d04e1d2a360d6f7e
Signed-off-by: default avatarSiddartha Shaik <sshaik@codeaurora.org>
parent 9f49c9c2

include/sound/q6asm-v2.h

+4 −0
Original line number Original line Diff line number Diff line
@@ -319,6 +319,10 @@ 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,

sound/soc/msm/qdsp6v2/q6asm.c

+71 −0
Original line number Original line Diff line number Diff line
@@ -3222,6 +3222,77 @@ int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample) 
	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,