Merge "dsp: add BT sink support in AFE driver" (8d72268c) · Commits · e / devices / android_kernel_fairphone_FP4

dsp/q6afe.c

+124 −23

Original line number	Diff line number	Diff line
		@@ -3442,15 +3442,21 @@ int afe_port_send_usb_dev_param(u16 port_id, union afe_port_config *afe_config)

		static int q6afe_send_dec_config(u16 port_id,
		union afe_port_config afe_config,
		struct afe_dec_config *cfg)
		struct afe_dec_config *cfg,
		u32 format,
		u16 afe_in_channels, u16 afe_in_bit_width)
		{
		struct afe_dec_media_fmt_t dec_media_fmt;
		struct avs_dec_depacketizer_id_param_t dec_depkt_id_param;
		struct avs_dec_congestion_buffer_param_t dec_buffer_id_param;
		struct afe_enc_dec_imc_info_param_t imc_info_param;
		struct afe_port_media_type_t media_type;
		struct param_hdr_v3 param_hdr;
		int ret;
		u32 dec_fmt;

		memset(&dec_depkt_id_param, 0, sizeof(dec_depkt_id_param));
		memset(&dec_media_fmt, 0, sizeof(dec_media_fmt));
		memset(&imc_info_param, 0, sizeof(imc_info_param));
		memset(&media_type, 0, sizeof(media_type));
		memset(&param_hdr, 0, sizeof(param_hdr));
		@@ -3462,11 +3468,11 @@ static int q6afe_send_dec_config(u16 port_id,
		__func__);
		param_hdr.param_id = AFE_DECODER_PARAM_ID_DEPACKETIZER_ID;
		param_hdr.param_size = sizeof(struct avs_dec_depacketizer_id_param_t);
		dec_depkt_id_param.dec_depacketizer_id =
		AFE_MODULE_ID_DEPACKETIZER_COP;
		if (cfg->format == ASM_MEDIA_FMT_APTX_ADAPTIVE)
		dec_depkt_id_param.dec_depacketizer_id =
		AFE_MODULE_ID_DEPACKETIZER_COP_V1;
		if (cfg->format == ENC_CODEC_TYPE_LDAC)
		dec_depkt_id_param.dec_depacketizer_id =
		AFE_MODULE_ID_DEPACKETIZER_COP;
		ret = q6afe_pack_and_set_param_in_band(port_id,
		q6audio_get_port_index(port_id),
		param_hdr,
		@@ -3477,10 +3483,41 @@ static int q6afe_send_dec_config(u16 port_id,
		goto exit;
		}

		switch (cfg->format) {
		case ASM_MEDIA_FMT_SBC:
		case ASM_MEDIA_FMT_AAC_V2:
		case ASM_MEDIA_FMT_MP3:
		if (port_id == SLIMBUS_9_TX) {
		dec_buffer_id_param.max_nr_buffers = 200;
		dec_buffer_id_param.pre_buffer_size = 200;
		} else {
		dec_buffer_id_param.max_nr_buffers = 0;
		dec_buffer_id_param.pre_buffer_size = 0;
		}
		pr_debug("%s: sending AFE_DECODER_PARAM_ID_CONGESTION_BUFFER_SIZE to DSP payload\n",
		__func__);
		param_hdr.param_id =
		AFE_DECODER_PARAM_ID_CONGESTION_BUFFER_SIZE;
		param_hdr.param_size =
		sizeof(struct avs_dec_congestion_buffer_param_t);
		dec_buffer_id_param.version = 0;
		ret = q6afe_pack_and_set_param_in_band(port_id,
		q6audio_get_port_index(port_id),
		param_hdr,
		(u8 *) &dec_buffer_id_param);
		if (ret) {
		pr_err("%s: AFE_DECODER_PARAM_ID_CONGESTION_BUFFER_SIZE for port 0x%x failed %d\n",
		__func__, port_id, ret);
		goto exit;
		}
		break;
		default:
		pr_debug("%s:sending AFE_ENCDEC_PARAM_ID_DEC_TO_ENC_COMMUNICATION to DSP payload\n",
		__func__);
		param_hdr.param_id = AFE_ENCDEC_PARAM_ID_DEC_TO_ENC_COMMUNICATION;
		param_hdr.param_size = sizeof(struct afe_enc_dec_imc_info_param_t);
		param_hdr.param_id =
		AFE_ENCDEC_PARAM_ID_DEC_TO_ENC_COMMUNICATION;
		param_hdr.param_size =
		sizeof(struct afe_enc_dec_imc_info_param_t);
		imc_info_param.imc_info = cfg->abr_dec_cfg.imc_info;
		ret = q6afe_pack_and_set_param_in_band(port_id,
		q6audio_get_port_index(port_id),
		@@ -3491,14 +3528,31 @@ static int q6afe_send_dec_config(u16 port_id,
		__func__, port_id, ret);
		goto exit;
		}
		break;
		}

		pr_debug("%s:Sending AFE_API_VERSION_PORT_MEDIA_TYPE to DSP", __func__);
		param_hdr.module_id = AFE_MODULE_PORT;
		param_hdr.param_id = AFE_PARAM_ID_PORT_MEDIA_TYPE;
		param_hdr.param_size = sizeof(struct afe_port_media_type_t);
		media_type.minor_version = AFE_API_VERSION_PORT_MEDIA_TYPE;
		media_type.sample_rate = afe_config.slim_sch.sample_rate;
		switch (cfg->format) {
		case ASM_MEDIA_FMT_AAC_V2:
		media_type.sample_rate =
		cfg->data.aac_config.sample_rate;
		break;
		default:
		media_type.sample_rate =
		afe_config.slim_sch.sample_rate;
		}
		if (afe_in_bit_width)
		media_type.bit_width = afe_in_bit_width;
		else
		media_type.bit_width = afe_config.slim_sch.bit_width;

		if (afe_in_channels)
		media_type.num_channels = afe_in_channels;
		else
		media_type.num_channels = afe_config.slim_sch.num_channels;
		media_type.data_format = AFE_PORT_DATA_FORMAT_PCM;
		media_type.reserved = 0;
		@@ -3512,6 +3566,50 @@ static int q6afe_send_dec_config(u16 port_id,
		goto exit;
		}

		if (format != ASM_MEDIA_FMT_SBC && format != ASM_MEDIA_FMT_AAC_V2) {
		pr_debug("%s:Unsuppported dec format. Ignore AFE config %u\n",
		__func__, format);
		goto exit;
		}
		pr_debug("%s: sending AFE_DECODER_PARAM_ID_DEC_MEDIA_FMT to DSP payload\n",
		__func__);
		param_hdr.module_id = AFE_MODULE_ID_DECODER;
		param_hdr.instance_id = INSTANCE_ID_0;
		param_hdr.param_id = AFE_DECODER_PARAM_ID_DEC_FMT_ID;
		param_hdr.param_size = sizeof(dec_fmt);
		dec_fmt = format;
		ret = q6afe_pack_and_set_param_in_band(port_id,
		q6audio_get_port_index(port_id),
		param_hdr, (u8 *) &dec_fmt);
		if (ret) {
		pr_err("%s: AFE_DECODER_PARAM_ID_DEC_MEDIA_FMT for port 0x%x failed %d\n",
		__func__, port_id, ret);
		goto exit;
		}

		switch (cfg->format) {
		case ASM_MEDIA_FMT_AAC_V2:
		param_hdr.param_size = sizeof(struct afe_dec_media_fmt_t);

		pr_debug("%s:send AFE_DECODER_PARAM_ID DEC_MEDIA_FMT to DSP payload\n",
		__func__);
		param_hdr.param_id = AVS_DECODER_PARAM_ID_DEC_MEDIA_FMT;
		dec_media_fmt.dec_media_config = cfg->data;
		ret = q6afe_pack_and_set_param_in_band(port_id,
		q6audio_get_port_index(port_id),
		param_hdr,
		(u8 *) &dec_media_fmt);
		if (ret) {
		pr_err("%s: AFE_DECODER_PARAM_ID DEC_MEDIA_FMT for port 0x%x failed %d\n",
		__func__, port_id, ret);
		goto exit;
		}
		break;
		default:
		pr_debug("%s:No need to send DEC_MEDIA_FMT to DSP payload\n",
		__func__);
		}

		exit:
		return ret;
		}
		@@ -3550,7 +3648,8 @@ static int q6afe_send_enc_config(u16 port_id,
		format != ASM_MEDIA_FMT_APTX && format != ASM_MEDIA_FMT_APTX_HD &&
		format != ASM_MEDIA_FMT_CELT && format != ASM_MEDIA_FMT_LDAC &&
		format != ASM_MEDIA_FMT_APTX_ADAPTIVE) {
		pr_err("%s:Unsuppported format Ignore AFE config\n", __func__);
		pr_err("%s:Unsuppported enc format. Ignore AFE config\n",
		__func__);
		return 0;
		}

		@@ -3582,7 +3681,7 @@ static int q6afe_send_enc_config(u16 port_id,
		enc_blk_param.enc_cfg_blk_size =
		sizeof(union afe_enc_config_data);
		}
		pr_debug("%s:send AFE_ENCODER_PARAM_ID_ENC_CFG_BLK to DSP payloadn",
		pr_debug("%s:send AFE_ENCODER_PARAM_ID_ENC_CFG_BLK to DSP payload\n",
		__func__);
		param_hdr.param_id = AFE_ENCODER_PARAM_ID_ENC_CFG_BLK;
		enc_blk_param.enc_blk_config = *cfg;
		@@ -4064,7 +4163,9 @@ static int __afe_port_start(u16 port_id, union afe_port_config *afe_config,
		pr_debug("%s: Found AFE decoder support for SLIMBUS format = %d\n",
		__func__, codec_format);
		ret = q6afe_send_dec_config(port_id, *afe_config,
		dec_cfg);
		dec_cfg, codec_format,
		afe_in_channels,
		afe_in_bit_width);
		if (ret) {
		pr_err("%s: AFE decoder config for port 0x%x failed %d\n",
		__func__, port_id, ret);

include/dsp/apr_audio-v2.h

+144 −1

Original line number	Diff line number	Diff line
		@@ -3875,6 +3875,12 @@ struct afe_id_aptx_adaptive_enc_init
		*/
		#define AFE_ENCODER_PARAM_ID_ENC_FMT_ID 0x0001322B

		/*
		* Decoder format ID parameter for the #AVS_MODULE_ID_DECODER module.
		* This parameter cannot be set runtime.
		*/
		#define AFE_DECODER_PARAM_ID_DEC_FMT_ID 0x00013234

		/*
		* Encoder scrambler parameter for the #AVS_MODULE_ID_ENCODER module.
		* This parameter cannot be set runtime.
		@@ -3915,6 +3921,12 @@ struct afe_id_aptx_adaptive_enc_init
		*/
		#define AFE_DECODER_PARAM_ID_DEPACKETIZER_ID 0x00013235

		/*
		* Decoder buffer ID parameter for the #AVS_MODULE_ID_DECODER module.
		* This parameter cannot be set runtime.
		*/
		#define AFE_DECODER_PARAM_ID_CONGESTION_BUFFER_SIZE 0x000132ec

		/*
		* Data format to send compressed data
		* is transmitted/received over Slimbus lines.
		@@ -4185,6 +4197,7 @@ struct asm_celt_enc_cfg_t {
		} __packed;

		#define ASM_MEDIA_FMT_LDAC 0x00013224
		#define ENC_CODEC_TYPE_LDAC 0x23000000
		struct asm_ldac_specific_enc_cfg_t {
		/*
		* This is used to calculate the encoder output
		@@ -4293,6 +4306,98 @@ struct afe_port_media_type_t {
		uint16_t reserved;
		} __packed;

		/*
		* Payload of the SBC decoder configuration parameters in the
		* #ASM_MEDIA_FMT_SBC media format.
		*/
		struct asm_sbc_dec_cfg_t {
		/* All configuration is extracted from the stream */
		} __packed;

		/*
		* Payload of the MP3 decoder configuration parameters in the
		* #ASM_MEDIA_FMT_MP3 media format.
		*/
		struct asm_mp3_dec_cfg_t {
		/* All configuration is extracted from the stream */
		} __packed;

		struct asm_aac_dec_cfg_v2_t {
		uint16_t aac_fmt_flag;
		/*
		* Bit stream format option.
		*
		* @values
		* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADTS
		* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LOAS
		* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_ADIF
		* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_RAW
		* - #ASM_MEDIA_FMT_AAC_FORMAT_FLAG_LATM
		*/

		uint16_t audio_obj_type;
		/*
		* Audio Object Type (AOT) present in the AAC stream.
		*
		* @values
		* - #ASM_MEDIA_FMT_AAC_AOT_LC
		* - #ASM_MEDIA_FMT_AAC_AOT_SBR
		* - #ASM_MEDIA_FMT_AAC_AOT_BSAC
		* - #ASM_MEDIA_FMT_AAC_AOT_PS
		*
		* Other values are not supported.
		*/

		uint16_t channel_config;
		/*
		* Number of channels present in the AAC stream.
		*
		* @values
		* - 0 -- PCE
		* - 1 -- Mono
		* - 2 -- Stereo
		* - 6 -- 5.1 content
		*/

		uint16_t total_size_of_PCE_bits;
		/*
		* For RAW formats and if channel_config=0 (PCE),
		* the client can send the bit stream containing PCE
		* immediately following this structure (in band).
		*
		* @values @ge 0 (does not include the bits required
		* for 32-bit alignment)
		*
		* If this field is set to 0, the PCE information is
		* assumed to be available in the audio bit stream
		* and not in band.
		*
		* If this field is greater than 0, the PCE information
		* follows this structure. Additional bits might
		* be required for 32-bit alignment.
		*/

		uint32_t sample_rate;
		/*
		* Number of samples per second.
		*
		* @values 8000, 11025, 12000, 16000, 22050, 24000, 32000,
		* 44100, 48000, 64000, 88200, 96000 Hz
		*
		* This field must be equal to the sample rate of the
		* AAC-LC decoder output.
		* - For MP4 or 3GP containers, this sample rate
		* is indicated by the
		* samplingFrequencyIndex field in the
		* AudioSpecificConfig element.
		* - For ADTS format, this sample rate is indicated by the
		* samplingFrequencyIndex in the ADTS fixed header.
		* - For ADIF format, this sample rate is indicated by the
		* samplingFrequencyIndex in the program_config_element
		* present in the ADIF header.
		*/
		} __packed;

		union afe_enc_config_data {
		struct asm_sbc_enc_cfg_t sbc_config;
		struct asm_aac_enc_cfg_v2_t aac_config;
		@@ -4309,9 +4414,16 @@ struct afe_enc_config {
		union afe_enc_config_data data;
		};

		union afe_dec_config_data {
		struct asm_sbc_dec_cfg_t sbc_config;
		struct asm_aac_dec_cfg_v2_t aac_config;
		struct asm_mp3_dec_cfg_t mp3_config;
		};

		struct afe_dec_config {
		u32 format;
		struct afe_abr_dec_cfg_t abr_dec_cfg;
		union afe_dec_config_data data;
		};

		struct afe_enc_cfg_blk_param_t {
		@@ -4322,6 +4434,14 @@ struct afe_enc_cfg_blk_param_t {
		union afe_enc_config_data enc_blk_config;
		};

		/*
		* Payload of the AVS_DECODER_PARAM_ID_DEC_MEDIA_FMT parameter used by
		* AVS_MODULE_ID_DECODER.
		*/
		struct afe_dec_media_fmt_t {
		union afe_dec_config_data dec_media_config;
		} __packed;

		/*
		* Payload of the AVS_ENCODER_PARAM_ID_PACKETIZER_ID parameter.
		*/
		@@ -4373,12 +4493,34 @@ struct afe_enc_dec_imc_info_param_t {
		struct avs_dec_depacketizer_id_param_t {
		/*
		* Supported values:
		* #AVS_MODULE_ID_DEPACKETIZER_COP
		* #AFE_MODULE_ID_DEPACKETIZER_COP
		* #AFE_MODULE_ID_DEPACKETIZER_COP_V1
		* Any OpenDSP supported values
		*/
		uint32_t dec_depacketizer_id;
		};

		struct avs_dec_congestion_buffer_param_t {
		uint32_t version;
		uint16_t max_nr_buffers;
		/*
		* Maximum number of 1ms buffers:
		* 0 - 256
		*/
		uint16_t pre_buffer_size;
		/*
		* Pre-buffering size in 1ms:
		* 1 - 128
		*/
		};

		/*
		* ID of the parameter used by #AVS_MODULE_ID_DECODER to configure
		* the decoder mode for the AFE module.
		* This parameter cannot be set at runtime.
		*/
		#define AVS_DECODER_PARAM_ID_DEC_MEDIA_FMT 0x00013232

		/* ID of the parameter used by #AFE_MODULE_AUDIO_DEV_INTERFACE to configure
		* the island mode for a given AFE port.
		*/
		@@ -4492,6 +4634,7 @@ union afe_port_config {
		struct avs_enc_packetizer_id_param_t enc_pkt_id_param;
		struct avs_enc_set_scrambler_param_t enc_set_scrambler_param;
		struct avs_dec_depacketizer_id_param_t dec_depkt_id_param;
		struct afe_dec_media_fmt_t dec_media_fmt;
		struct afe_enc_level_to_bitrate_map_param_t map_param;
		struct afe_enc_dec_imc_info_param_t imc_info_param;
		struct afe_param_id_cdc_dma_cfg_t cdc_dma;