ASoC: msm: qdsp6v2: support for AFE encoder

Add support for AFE encoder configuration.
Add new mixer controls to support AFE input channel
and encoder configuration.
Add support in AFE to set the encoder and input
channel configuration.If encoder configuration is set
ADSP will encode the PCM steam and send over backend.

Change-Id: I56581533ffff3b61f5b5ee128841ab61b62674d7
Signed-off-by: Naresh Tanniru <ntanniru@codeaurora.org>

2 files changed, 334 insertions, 0 deletions

diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
index 8d9016e3557c..3464726c408a 100644
--- a/include/sound/apr_audio-v2.h
+++ b/include/sound/apr_audio-v2.h
@@ -2738,6 +2738,333 @@ struct afe_param_id_set_topology_cfg {
 	u32		topology_id;
 } __packed;
 
+
+/*
+ * Generic encoder module ID.
+ * This module supports the following parameter IDs:
+ * #AVS_ENCODER_PARAM_ID_ENC_FMT_ID (cannot be set run time)
+ * #AVS_ENCODER_PARAM_ID_ENC_CFG_BLK (may be set run time)
+ * #AVS_ENCODER_PARAM_ID_ENC_BITRATE (may be set run time)
+ * #AVS_ENCODER_PARAM_ID_PACKETIZER_ID (cannot be set run time)
+ * Opcode - AVS_MODULE_ID_ENCODER
+ * AFE Command AFE_PORT_CMD_SET_PARAM_V2 supports this module ID.
+ */
+#define AFE_MODULE_ID_ENCODER        0x00013229
+
+/* Macro for defining the packetizer ID: COP. */
+#define AFE_MODULE_ID_PACKETIZER_COP 0x0001322A
+
+/*
+ * Packetizer type parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter cannot be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_PACKETIZER_ID 0x0001322E
+
+/*
+ * Encoder config block  parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter may be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_ENC_CFG_BLK 0x0001322C
+
+/*
+ * Encoder format ID parameter for the #AVS_MODULE_ID_ENCODER module.
+ * This parameter cannot be set runtime.
+ */
+#define AFE_ENCODER_PARAM_ID_ENC_FMT_ID         0x0001322B
+
+/*
+ * Data format to send compressed data
+ * is transmitted/received over Slimbus lines.
+ */
+#define AFE_SB_DATA_FORMAT_GENERIC_COMPRESSED    0x3
+
+/*
+ * ID for AFE port module. This will be used to define port properties.
+ * This module supports following parameter IDs:
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ * To configure the port property, the client must use the
+ * #AFE_PORT_CMD_SET_PARAM_V2 command,
+ * and fill the module ID with the respective parameter IDs as listed above.
+ * @apr_hdr_fields
+ * Opcode -- AFE_MODULE_PORT
+ */
+#define AFE_MODULE_PORT                          0x000102a6
+
+/*
+ * ID of the parameter used by #AFE_MODULE_PORT to set the port media type.
+ * parameter ID is currently supported using#AFE_PORT_CMD_SET_PARAM_V2 command.
+ */
+#define AFE_PARAM_ID_PORT_MEDIA_TYPE              0x000102a7
+
+/*
+ * Macros for defining the "data_format" field in the
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ */
+#define AFE_PORT_DATA_FORMAT_PCM                  0x0
+#define AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED   0x1
+
+/*
+ * Macro for defining the "minor_version" field in the
+ * #AFE_PARAM_ID_PORT_MEDIA_TYPE
+ */
+#define AFE_API_VERSION_PORT_MEDIA_TYPE           0x1
+
+#define ASM_MEDIA_FMT_NONE                        0x0
+
+/*
+ * Media format ID for SBC encode configuration.
+ * @par SBC encode configuration (asm_sbc_enc_cfg_t)
+ * @table{weak__asm__sbc__enc__cfg__t}
+ */
+#define ASM_MEDIA_FMT_SBC                         0x00010BF2
+
+/* SBC channel Mono mode.*/
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO                     1
+
+/* SBC channel Stereo mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO                   2
+
+/* SBC channel Dual Mono mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO                8
+
+/* SBC channel Joint Stereo mode. */
+#define ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO             9
+
+/* SBC bit allocation method = loudness. */
+#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS            0
+
+/* SBC bit allocation method = SNR. */
+#define ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR                 1
+
+
+/*
+ * Payload of the SBC encoder configuration parameters in the
+ * #ASM_MEDIA_FMT_SBC media format.
+ */
+struct asm_sbc_enc_cfg_t {
+	/*
+	 * Number of subbands.
+	 * @values 4, 8
+	 */
+	uint32_t    num_subbands;
+
+	/*
+	 * Size of the encoded block in samples.
+	 * @values 4, 8, 12, 16
+	 */
+	uint32_t    blk_len;
+
+	/*
+	 * Mode used to allocate bits between channels.
+	 * @values
+	 * 0 (Native mode)
+	 * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_MONO
+	 * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_STEREO
+	 * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_DUAL_MONO
+	 * #ASM_MEDIA_FMT_SBC_CHANNEL_MODE_JOINT_STEREO
+	 * Native mode indicates that encoding must be performed with the number
+	 * of channels at the input.
+	 * If postprocessing outputs one-channel data, Mono mode is used. If
+	 * postprocessing outputs two-channel data, Stereo mode is used.
+	 * The number of channels must not change during encoding.
+	 */
+	uint32_t    channel_mode;
+
+	/*
+	 * Encoder bit allocation method.
+	 * @values
+	 * #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_LOUDNESS
+	 * #ASM_MEDIA_FMT_SBC_ALLOCATION_METHOD_SNR @tablebulletend
+	 */
+	uint32_t    alloc_method;
+
+	/*
+	 * Number of encoded bits per second.
+	 * @values
+	 * Mono channel -- Maximum of 320 kbps
+	 * Stereo channel -- Maximum of 512 kbps @tablebulletend
+	 */
+	uint32_t    bit_rate;
+
+	/*
+	 * Number of samples per second.
+	 * @values 0 (Native mode), 16000, 32000, 44100, 48000 Hz
+	 * Native mode indicates that encoding must be performed with the
+	 * sampling rate at the input.
+	 * The sampling rate must not change during encoding.
+	 */
+	uint32_t    sample_rate;
+};
+
+#define ASM_MEDIA_FMT_AAC_AOT_LC            2
+#define ASM_MEDIA_FMT_AAC_AOT_SBR           5
+#define ASM_MEDIA_FMT_AAC_AOT_PS            29
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS  0
+#define ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW   3
+
+struct asm_aac_enc_cfg_v2_t {
+
+	/* Encoding rate in bits per second.*/
+	uint32_t     bit_rate;
+
+	/*
+	 * Encoding mode.
+	 * Supported values:
+	 * #ASM_MEDIA_FMT_AAC_AOT_LC
+	 * #ASM_MEDIA_FMT_AAC_AOT_SBR
+	 * #ASM_MEDIA_FMT_AAC_AOT_PS
+	 */
+	uint32_t     enc_mode;
+
+	/*
+	 * AAC format flag.
+	 * Supported values:
+	 * #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
+	 * #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
+	 */
+	uint16_t     aac_fmt_flag;
+
+	/*
+	 * Number of channels to encode.
+	 * Supported values:
+	 * 0 - Native mode
+	 * 1 - Mono
+	 * 2 - Stereo
+	 * Other values are not supported.
+	 * @note1hang The eAAC+ encoder mode supports only stereo.
+	 * Native mode indicates that encoding must be performed with the
+	 * number of channels at the input.
+	 * The number of channels must not change during encoding.
+	 */
+	uint32_t     channel_cfg;
+
+	/*
+	 * Number of samples per second.
+	 * Supported values: - 0 -- Native mode - For other values,
+	 * Native mode indicates that encoding must be performed with the
+	 * sampling rate at the input.
+	 * The sampling rate must not change during encoding.
+	 */
+	uint32_t     sample_rate;
+} __packed;
+
+/* FMT ID for apt-X Classic */
+#define ASM_MEDIA_FMT_APTX 0x000131ff
+
+/* FMT ID for apt-X HD */
+#define ASM_MEDIA_FMT_APTX_HD 0x00013200
+
+#define PCM_CHANNEL_L         1
+#define PCM_CHANNEL_R         2
+#define PCM_CHANNEL_C         3
+
+struct asm_custom_enc_cfg_aptx_t {
+	uint32_t    sample_rate;
+	/* Mono or stereo */
+	uint16_t    num_channels;
+	uint16_t    reserved;
+	/* num_ch == 1, then PCM_CHANNEL_C,
+	 * num_ch == 2, then {PCM_CHANNEL_L, PCM_CHANNEL_R}
+	 */
+	uint8_t     channel_mapping[8];
+	uint32_t    custom_size;
+} __packed;
+
+struct afe_enc_fmt_id_param_t {
+	/*
+	 * Supported values:
+	 *  #ASM_MEDIA_FMT_SBC
+	 *  #ASM_MEDIA_FMT_AAC_V2
+	 * Any OpenDSP supported values
+	 */
+	uint32_t    fmt_id;
+} __packed;
+
+struct afe_port_media_type_t {
+	/*
+	 * Minor version
+	 * @values #AFE_API_VERSION_PORT_MEDIA_TYPE.
+	 */
+	uint32_t    minor_version;
+
+	/*
+	 * Sampling rate of the port.
+	 * @values
+	 * #AFE_PORT_SAMPLE_RATE_8K
+	 * #AFE_PORT_SAMPLE_RATE_11_025K
+	 * #AFE_PORT_SAMPLE_RATE_12K
+	 * #AFE_PORT_SAMPLE_RATE_16K
+	 * #AFE_PORT_SAMPLE_RATE_22_05K
+	 * #AFE_PORT_SAMPLE_RATE_24K
+	 * #AFE_PORT_SAMPLE_RATE_32K
+	 * #AFE_PORT_SAMPLE_RATE_44_1K
+	 * #AFE_PORT_SAMPLE_RATE_48K
+	 * #AFE_PORT_SAMPLE_RATE_88_2K
+	 * #AFE_PORT_SAMPLE_RATE_96K
+	 * #AFE_PORT_SAMPLE_RATE_176_4K
+	 * #AFE_PORT_SAMPLE_RATE_192K
+	 * #AFE_PORT_SAMPLE_RATE_352_8K
+	 * #AFE_PORT_SAMPLE_RATE_384K
+	 */
+	uint32_t    sample_rate;
+
+	/*
+	 * Bit width of the sample.
+	 * @values 16, 24
+	 */
+	uint16_t    bit_width;
+
+	/*
+	 * Number of channels.
+	 * @values 1 to #AFE_PORT_MAX_AUDIO_CHAN_CNT
+	 */
+	uint16_t    num_channels;
+
+	/*
+	 * Data format supported by this port.
+	 * If the port media type and device media type are different,
+	 * it signifies a encoding/decoding use case
+	 * @values
+	 * #AFE_PORT_DATA_FORMAT_PCM
+	 * #AFE_PORT_DATA_FORMAT_GENERIC_COMPRESSED
+	 */
+	uint16_t   data_format;
+
+	/*This field must be set to zero.*/
+	uint16_t   reserved;
+} __packed;
+
+union afe_enc_config_data {
+	struct asm_sbc_enc_cfg_t sbc_config;
+	struct asm_aac_enc_cfg_v2_t aac_config;
+	struct asm_custom_enc_cfg_aptx_t  aptx_config;
+};
+
+struct afe_enc_config {
+	u32 format;
+	union afe_enc_config_data data;
+};
+
+struct afe_enc_cfg_blk_param_t {
+	uint32_t enc_cfg_blk_size;
+	/*
+	 *Size of the encoder configuration block that follows this member
+	 */
+	union afe_enc_config_data enc_blk_config;
+};
+
+/*
+ * Payload of the AVS_ENCODER_PARAM_ID_PACKETIZER_ID parameter.
+ */
+struct avs_enc_packetizer_id_param_t {
+	/*
+	 * Supported values:
+	 * #AVS_MODULE_ID_PACKETIZER_COP
+	 * Any OpenDSP supported values
+	 */
+	uint32_t enc_packetizer_id;
+};
+
 union afe_port_config {
 	struct afe_param_id_pcm_cfg               pcm;
 	struct afe_param_id_i2s_cfg               i2s;
@@ -2751,6 +3078,10 @@ union afe_port_config {
 	struct afe_param_id_set_topology_cfg      topology;
 	struct afe_param_id_tdm_cfg               tdm;
 	struct afe_param_id_usb_audio_cfg         usb_audio;
+	struct afe_enc_fmt_id_param_t             enc_fmt;
+	struct afe_port_media_type_t              media_type;
+	struct afe_enc_cfg_blk_param_t            enc_blk_param;
+	struct avs_enc_packetizer_id_param_t      enc_pkt_id_param;
 } __packed;
 
 struct afe_audioif_config_command_no_payload {
diff --git a/include/sound/q6afe-v2.h b/include/sound/q6afe-v2.h
index 5cd65357dd95..6be903a4c8d0 100644
--- a/include/sound/q6afe-v2.h
+++ b/include/sound/q6afe-v2.h
@@ -274,6 +274,9 @@ int afe_rt_proxy_port_read(phys_addr_t buf_addr_p,
 void afe_set_cal_mode(u16 port_id, enum afe_cal_mode afe_cal_mode);
 int afe_port_start(u16 port_id, union afe_port_config *afe_config,
 	u32 rate);
+int afe_port_start_v2(u16 port_id, union afe_port_config *afe_config,
+		      u32 rate, u16 afe_in_channels,
+		      struct afe_enc_config *enc_config);
 int afe_spk_prot_feed_back_cfg(int src_port, int dst_port,
 	int l_ch, int r_ch, u32 enable);
 int afe_spk_prot_get_calib_data(struct afe_spkr_prot_get_vi_calib *calib);