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Commit db567c19 authored by Josh Kirsch's avatar Josh Kirsch Committed by Banajit Goswami
Browse files

ASoC: msm: qdsp6v2: Add TDM support for PCM compressed mode 



Add support for 352800 sampling rates and 32 bps for the TDM.
Add support for PCM compress passthough mode with mixer controls.

CRs-fixed: 1116515
Change-Id: Iab059a5a6b6ce8f57717023467677a399a60032e
Signed-off-by: default avatarJosh Kirsch <jkirsch@codeaurora.org>
Signed-off-by: default avatarKarthikeyan Mani <kmani@codeaurora.org>
parent 84557094

include/sound/apr_audio-v2.h

+47 −4
Original line number Diff line number Diff line
@@ -1815,11 +1815,14 @@ struct afe_port_data_cmd_rt_proxy_port_read_v2 { 
#define AFE_PORT_SAMPLE_RATE_16K          16000

#define AFE_PORT_SAMPLE_RATE_48K          48000

#define AFE_PORT_SAMPLE_RATE_96K          96000

#define AFE_PORT_SAMPLE_RATE_176P4K       176400

#define AFE_PORT_SAMPLE_RATE_192K         192000

#define AFE_PORT_SAMPLE_RATE_352P8K       352800

#define AFE_LINEAR_PCM_DATA				0x0

#define AFE_NON_LINEAR_DATA				0x1

#define AFE_LINEAR_PCM_DATA_PACKED_60958 0x2

#define AFE_NON_LINEAR_DATA_PACKED_60958 0x3

#define AFE_GENERIC_COMPRESSED           0x8



/* This param id is used to configure I2S interface */

#define AFE_PARAM_ID_I2S_CONFIG	0x0001020D
@@ -2745,7 +2748,9 @@ struct afe_param_id_tdm_cfg { 
	 * - #AFE_PORT_SAMPLE_RATE_16K

	 * - #AFE_PORT_SAMPLE_RATE_24K

	 * - #AFE_PORT_SAMPLE_RATE_32K

	 * - #AFE_PORT_SAMPLE_RATE_48K @tablebulletend

	 * - #AFE_PORT_SAMPLE_RATE_48K

	 * - #AFE_PORT_SAMPLE_RATE_176P4K

	 * - #AFE_PORT_SAMPLE_RATE_352P8K @tablebulletend

	 */



	u32	bit_width;
@@ -2754,10 +2759,11 @@ struct afe_param_id_tdm_cfg { 
	 */



	u16	data_format;

	/* < Data format: linear and compressed

	/* < Data format: linear ,compressed, generic compresssed

	 * @values

	 * - #AFE_LINEAR_PCM_DATA

	 * - #AFE_NON_LINEAR_DATA @tablebulletend

	 * - #AFE_NON_LINEAR_DATA

	 * - #AFE_GENERIC_COMPRESSED

	 */



	u16	sync_mode;
@@ -3636,7 +3642,7 @@ struct afe_lpass_core_shared_clk_config_command { 
#define DEFAULT_COPP_TOPOLOGY				0x00010314

#define DEFAULT_POPP_TOPOLOGY				0x00010BE4

#define COMPRESSED_PASSTHROUGH_DEFAULT_TOPOLOGY         0x0001076B

#define COMPRESS_PASSTHROUGH_NONE_TOPOLOGY      0x00010774

#define COMPRESSED_PASSTHROUGH_NONE_TOPOLOGY            0x00010774

#define VPM_TX_SM_ECNS_COPP_TOPOLOGY			0x00010F71

#define VPM_TX_DM_FLUENCE_COPP_TOPOLOGY			0x00010F72

#define VPM_TX_QMIC_FLUENCE_COPP_TOPOLOGY		0x00010F75
@@ -3942,6 +3948,8 @@ struct asm_softvolume_params { 


#define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0



#define ASM_MEDIA_FMT_GENERIC_COMPRESSED  0x00013212



#define ASM_MAX_EQ_BANDS 12



#define ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2 0x00010D98
@@ -3951,6 +3959,40 @@ u32 fmt_blk_size; 
	/* Media format block size in bytes.*/

}  __packed;



struct asm_generic_compressed_fmt_blk_t {

	struct apr_hdr hdr;

	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;



	/*

	 * Channel mapping array of bitstream output.

	 * Channel[i] mapping describes channel i inside the buffer, where

	 * i < num_channels. All valid used channels must be

	 * present at the beginning of the array.

	 */

	uint8_t channel_mapping[8];



	/*

	 * Number of channels of the incoming bitstream.

	 * Supported values: 1,2,3,4,5,6,7,8

	 */

	uint16_t num_channels;



	/*

	 * Nominal bits per sample value of the incoming bitstream.

	 * Supported values: 16, 32

	 */

	uint16_t bits_per_sample;



	/*

	 * Nominal sampling rate of the incoming bitstream.

	 * Supported values: 8000, 11025, 16000, 22050, 24000, 32000,

	 *                   44100, 48000, 88200, 96000, 176400, 192000,

	 *                   352800, 384000

	 */

	uint32_t sampling_rate;



} __packed;



struct asm_multi_channel_pcm_fmt_blk_v2 {

	struct apr_hdr hdr;

	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
@@ -10318,6 +10360,7 @@ enum { 
	COMPRESSED_PASSTHROUGH_CONVERT,

	COMPRESSED_PASSTHROUGH_DSD,

	LISTEN,

	COMPRESSED_PASSTHROUGH_GEN,

};



#define AUDPROC_MODULE_ID_COMPRESSED_MUTE                0x00010770

include/sound/q6afe-v2.h

+1 −1
Original line number Diff line number Diff line
@@ -364,7 +364,7 @@ int afe_send_custom_tdm_header_cfg( 
	struct afe_param_id_custom_tdm_header_cfg *custom_tdm_header_cfg,

	u16 port_id);

int afe_tdm_port_start(u16 port_id, struct afe_tdm_port_config *tdm_port,

		u32 rate);

		       u32 rate, u16 num_groups);

void afe_set_routing_callback(routing_cb cb);

int afe_get_av_dev_drift(struct afe_param_id_dev_timing_stats *timing_stats,

		u16 port);

include/sound/q6asm-v2.h

+6 −0
Original line number Diff line number Diff line
@@ -54,6 +54,7 @@ 
#define FORMAT_DTS          0x001c

#define FORMAT_DSD          0x001d

#define FORMAT_APTX         0x001e

#define FORMAT_GEN_COMPR    0x001f



#define ENCDEC_SBCBITRATE   0x0001

#define ENCDEC_IMMEDIATE_DECODE 0x0002
@@ -500,6 +501,11 @@ int q6asm_media_format_block_multi_ch_pcm_v2( 
			uint32_t rate, uint32_t channels,

			bool use_default_chmap, char *channel_map,

			uint16_t bits_per_sample);

int q6asm_media_format_block_gen_compr(

			struct audio_client *ac,

			uint32_t rate, uint32_t channels,

			bool use_default_chmap, char *channel_map,

			uint16_t bits_per_sample);



int q6asm_media_format_block_multi_ch_pcm_v3(struct audio_client *ac,

					     uint32_t rate, uint32_t channels,