Merge "ASoC: msm: qdsp6v2: Upgrade to RTAC version 1.5" (723013cd) · Commits · e / devices / android_kernel_xiaomi_markw

include/linux/qdsp6v2/rtac.h

+4 −0

Original line number	Diff line number	Diff line
		@@ -24,6 +24,8 @@
		#define RTAC_MAX_ACTIVE_DEVICES 4
		#define RTAC_MAX_ACTIVE_POPP 8

		#define DEFAULT_APP_TYPE 0x00011130

		enum {
		ADM_RTAC_CAL,
		ASM_RTAC_CAL,
		@@ -53,10 +55,12 @@ struct rtac_cal_block_data {
		struct rtac_popp_data {
		uint32_t popp;
		uint32_t popp_topology;
		uint32_t app_type;
		};

		struct rtac_adm_data {
		uint32_t topology_id;
		uint32_t afe_topology;
		uint32_t afe_port;
		uint32_t copp;
		uint32_t num_of_popp;

include/sound/q6afe-v2.h

+1 −0

Original line number	Diff line number	Diff line
		@@ -152,6 +152,7 @@ int afe_sidetone(u16 tx_port_id, u16 rx_port_id, u16 enable, uint16_t gain);
		int afe_loopback_gain(u16 port_id, u16 volume);
		int afe_validate_port(u16 port_id);
		int afe_get_port_index(u16 port_id);
		int afe_get_topology(int port_id);
		int afe_start_pseudo_port(u16 port_id);
		int afe_stop_pseudo_port(u16 port_id);
		uint32_t afe_req_mmap_handle(struct afe_audio_client *ac);

include/sound/q6asm-v2.h

+3 −2

Original line number	Diff line number	Diff line
		@@ -190,6 +190,7 @@ struct audio_client {
		int stream_id;
		struct device *dev;
		int topology;
		int app_type;
		/* audio cache operations fptr*/
		int (fptr_cache_ops)(struct audio_buffer abuff, int cache_op);
		atomic_t unmap_cb_success;
		@@ -471,8 +472,8 @@ int q6asm_send_meta_data(struct audio_client *ac, uint32_t initial_samples,
		int q6asm_stream_send_meta_data(struct audio_client *ac, uint32_t stream_id,
		uint32_t initial_samples, uint32_t trailing_samples);

		/* Get current ASM topology */
		int q6asm_get_asm_topology(void);
		int q6asm_get_asm_topology(int session_id);
		int q6asm_get_asm_app_type(int session_id);

		int q6asm_send_mtmx_strtr_window(struct audio_client *ac,
		struct asm_session_mtmx_strtr_param_window_v2_t *window_param,

sound/soc/msm/qdsp6v2/q6afe.c

+20 −0

Original line number	Diff line number	Diff line
		@@ -89,6 +89,7 @@ struct afe_ctl {
		void *rx_private_data;
		uint32_t mmap_handle;

		int topology[AFE_MAX_PORTS];
		struct cal_type_data *cal_data[MAX_AFE_CAL_TYPES];

		atomic_t mem_map_cal_handles[MAX_AFE_CAL_TYPES];
		@@ -127,6 +128,22 @@ static int afe_get_cal_hw_delay(int32_t path,
		struct audio_cal_hw_delay_entry *entry);
		static int remap_cal_data(struct cal_block_data *cal_block, int cal_index);

		int afe_get_topology(int port_id)
		{
		int topology;
		int port_index = afe_get_port_index(port_id);

		if ((port_index < 0) \|\| (port_index > AFE_MAX_PORTS)) {
		pr_err("%s: Invalid port index %d\n", __func__, port_index);
		topology = -EINVAL;
		goto done;
		}

		topology = this_afe.topology[port_index];
		done:
		return topology;
		}

		void afe_set_aanc_info(struct aanc_data *q6_aanc_info)
		{
		this_afe.aanc_info.aanc_active = q6_aanc_info->aanc_active;
		@@ -1067,6 +1084,8 @@ static int afe_send_port_topology_id(u16 port_id)
		__func__, port_id, ret);
		goto done;
		}

		this_afe.topology[index] = topology_id;
		done:
		pr_debug("%s: AFE set topology id 0x%x enable for port 0x%x ret %d\n",
		__func__, topology_id, port_id, ret);
		@@ -3895,6 +3914,7 @@ int afe_close(int port_id)
		port_index = afe_get_port_index(port_id);
		if ((port_index >= 0) && (port_index < AFE_MAX_PORTS)) {
		this_afe.afe_sample_rates[port_index] = 0;
		this_afe.topology[port_index] = 0;
		} else {
		pr_err("%s: port %d\n", __func__, port_index);
		ret = -EINVAL;

sound/soc/msm/qdsp6v2/q6asm.c

+69 −7

Original line number	Diff line number	Diff line
		@@ -90,6 +90,8 @@ void *q6asm_mmap_apr_reg(void);

		static int q6asm_is_valid_session(struct apr_client_data data, void priv);
		static int q6asm_send_asm_cal(struct audio_client *ac);
		static int q6asm_get_asm_topology_cal(void);
		static int q6asm_get_asm_app_type_cal(void);

		/* for ASM custom topology */
		static struct cal_type_data *cal_data[ASM_MAX_CAL_TYPES];
		@@ -2138,13 +2140,15 @@ static int __q6asm_open_read(struct audio_client *ac,
		/* Stream prio : High, provide meta info with encoded frames */
		open.src_endpointype = ASM_END_POINT_DEVICE_MATRIX;

		open.preprocopo_id = q6asm_get_asm_topology();
		open.preprocopo_id = q6asm_get_asm_topology_cal();
		if ((open.preprocopo_id == ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX) \|\|
		(open.preprocopo_id == ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS))
		open.preprocopo_id = ASM_STREAM_POSTPROCOPO_ID_NONE;
		open.bits_per_sample = bits_per_sample;
		open.mode_flags = 0x0;

		ac->topology = open.preprocopo_id;
		ac->app_type = q6asm_get_asm_app_type_cal();
		if (ac->perf_mode == LOW_LATENCY_PCM_MODE) {
		open.mode_flags \|= ASM_LOW_LATENCY_TX_STREAM_SESSION <<
		ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_READ;
		@@ -2364,7 +2368,7 @@ static int __q6asm_open_write(struct audio_client *ac, uint32_t format,
		open.sink_endpointype = ASM_END_POINT_DEVICE_MATRIX;
		open.bits_per_sample = bits_per_sample;

		open.postprocopo_id = q6asm_get_asm_topology();
		open.postprocopo_id = q6asm_get_asm_topology_cal();
		if ((ac->perf_mode != LEGACY_PCM_MODE) &&
		((open.postprocopo_id == ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX) \|\|
		(open.postprocopo_id == ASM_STREAM_POSTPROC_TOPO_ID_HPX_PLUS)))
		@@ -2382,8 +2386,10 @@ static int __q6asm_open_write(struct audio_client *ac, uint32_t format,
		* For Gapless playback it will use the same session for next stream,
		* So use the same topology
		*/
		if (!ac->topology)
		if (!ac->topology) {
		ac->topology = open.postprocopo_id;
		ac->app_type = q6asm_get_asm_app_type_cal();
		}
		switch (format) {
		case FORMAT_LINEAR_PCM:
		open.dec_fmt_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2;
		@@ -2510,11 +2516,12 @@ static int __q6asm_open_read_write(struct audio_client *ac, uint32_t rd_format,
		open.mode_flags = is_meta_data_mode ? BUFFER_META_ENABLE : 0;
		open.bits_per_sample = bits_per_sample;
		/* source endpoint : matrix */
		open.postprocopo_id = q6asm_get_asm_topology();
		open.postprocopo_id = q6asm_get_asm_topology_cal();

		open.postprocopo_id = overwrite_topology ?
		topology : open.postprocopo_id;
		ac->topology = open.postprocopo_id;
		ac->app_type = q6asm_get_asm_app_type_cal();

		/* For DTS EAGLE only, force 24 bit */
		if ((open.postprocopo_id == ASM_STREAM_POSTPROC_TOPO_ID_DTS_HPX) \|\|
		@@ -2683,8 +2690,9 @@ int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample)
		open.src_endpointype = 0;
		open.sink_endpointype = 0;
		/* source endpoint : matrix */
		open.postprocopo_id = q6asm_get_asm_topology();
		open.postprocopo_id = q6asm_get_asm_topology_cal();

		ac->app_type = q6asm_get_asm_app_type_cal();
		ac->topology = open.postprocopo_id;
		open.bits_per_sample = bits_per_sample;
		open.reserved = 0;
		@@ -6492,11 +6500,40 @@ int q6asm_get_apr_service_id(int session_id)
		return ((struct apr_svc *)session[session_id]->apr)->id;
		}

		int q6asm_get_asm_topology(void)
		int q6asm_get_asm_topology(int session_id)
		{
		int topology;

		if (session_id <= 0 \|\| session_id > SESSION_MAX) {
		pr_err("%s: invalid session_id = %d\n", __func__, session_id);
		topology = -EINVAL;
		goto done;
		}

		topology = session[session_id]->topology;
		done:
		return topology;
		}

		int q6asm_get_asm_app_type(int session_id)
		{
		int app_type;

		if (session_id <= 0 \|\| session_id > SESSION_MAX) {
		pr_err("%s: invalid session_id = %d\n", __func__, session_id);
		app_type = -EINVAL;
		goto done;
		}

		app_type = session[session_id]->app_type;
		done:
		return app_type;
		}

		static int q6asm_get_asm_topology_cal(void)
		{
		int topology = DEFAULT_POPP_TOPOLOGY;
		struct cal_block_data *cal_block = NULL;
		pr_debug("%s:\n", __func__);

		if (cal_data[ASM_TOPOLOGY_CAL] == NULL)
		goto done;
		@@ -6515,6 +6552,31 @@ done:
		return topology;
		}

		static int q6asm_get_asm_app_type_cal(void)
		{
		int app_type = DEFAULT_APP_TYPE;
		struct cal_block_data *cal_block = NULL;

		if (cal_data[ASM_TOPOLOGY_CAL] == NULL)
		goto done;

		mutex_lock(&cal_data[ASM_TOPOLOGY_CAL]->lock);
		cal_block = cal_utils_get_only_cal_block(cal_data[ASM_TOPOLOGY_CAL]);
		if (cal_block == NULL)
		goto unlock;

		app_type = ((struct audio_cal_info_asm_top *)
		cal_block->cal_info)->app_type;

		if (app_type == 0)
		app_type = DEFAULT_APP_TYPE;
		unlock:
		mutex_unlock(&cal_data[ASM_TOPOLOGY_CAL]->lock);
		done:
		pr_debug("%s: Using app_type %d\n", __func__, app_type);
		return app_type;
		}

		static int q6asm_send_asm_cal(struct audio_client *ac)
		{
		struct cal_block_data *cal_block = NULL;