ASoC: msm: qdsp6v2: add support for latest version of media format command

Add support for ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4
command. This command adds support to playback/record 32 bit data
in 32 bit word and also provides a way to inform DSP about the
endianness of the data.

Change-Id: I3b013bedde8ccfa97a02e255e237df0cf2de13b8
Signed-off-by: Ashish Jain <ashishj@codeaurora.org>

2 files changed, 197 insertions, 0 deletions

diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h
index 1a58a146c3b0..06b72b262395 100644
--- a/include/sound/apr_audio-v2.h
+++ b/include/sound/apr_audio-v2.h
@@ -3678,6 +3678,8 @@ struct asm_softvolume_params {
 
 #define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 0x00010DDC
 
+#define ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 0x0001320C
+
 #define ASM_MEDIA_FMT_EVRCB_FS 0x00010BEF
 
 #define ASM_MEDIA_FMT_EVRCWB_FS 0x00010BF0
@@ -3780,6 +3782,56 @@ struct asm_multi_channel_pcm_fmt_blk_v3 {
  */
 } __packed;
 
+struct asm_multi_channel_pcm_fmt_blk_v4 {
+	uint16_t                num_channels;
+/*
+ * Number of channels
+ * Supported values: 1 to 8
+ */
+
+	uint16_t                bits_per_sample;
+/*
+ * Number of bits per sample per channel
+ * Supported values: 16, 24, 32
+ */
+
+	uint32_t                sample_rate;
+/*
+ * Number of samples per second
+ * Supported values: 2000 to 48000, 96000,192000 Hz
+ */
+
+	uint16_t                is_signed;
+/* Flag that indicates that PCM samples are signed (1) */
+
+	uint16_t                sample_word_size;
+/*
+ * Size in bits of the word that holds a sample of a channel.
+ * Supported values: 12,24,32
+ */
+
+	uint8_t                 channel_mapping[8];
+/*
+ * Each element, i, in the array describes channel i inside the buffer where
+ * 0 <= i < num_channels. Unused channels are set to 0.
+ */
+	uint16_t                endianness;
+/*
+ * Flag to indicate the endianness of the pcm sample
+ * Supported values: 0 - Little endian (all other formats)
+ *                   1 - Big endian (AIFF)
+ */
+	uint16_t                mode;
+/*
+ * Mode to provide additional info about the pcm input data.
+ * Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
+ *                       Q31 for unpacked 24b or 32b)
+ *                  15 - for 16 bit
+ *                  23 - for 24b packed or 8.24 format
+ *                  31 - for 24b unpacked or 32bit
+ */
+} __packed;
+
 /*
  * Payload of the multichannel PCM configuration parameters in
  * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 media format.
@@ -3790,6 +3842,16 @@ struct asm_multi_channel_pcm_fmt_blk_param_v3 {
 	struct asm_multi_channel_pcm_fmt_blk_v3 param;
 } __packed;
 
+/*
+ * Payload of the multichannel PCM configuration parameters in
+ * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
+ */
+struct asm_multi_channel_pcm_fmt_blk_param_v4 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmt_blk;
+	struct asm_multi_channel_pcm_fmt_blk_v4 param;
+} __packed;
+
 struct asm_stream_cmd_set_encdec_param {
 	u32                  param_id;
 	/* ID of the parameter. */
@@ -3825,6 +3887,79 @@ struct asm_dec_ddp_endp_param_v2 {
 	int endp_param_value;
 } __packed;
 
+/*
+ * Payload of the multichannel PCM encoder configuration parameters in
+ * the ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V4 media format.
+ */
+
+struct asm_multi_channel_pcm_enc_cfg_v4 {
+	struct apr_hdr hdr;
+	struct asm_stream_cmd_set_encdec_param encdec;
+	struct asm_enc_cfg_blk_param_v2 encblk;
+	uint16_t num_channels;
+	/*
+	 * Number of PCM channels.
+	 * @values
+	 * - 0 -- Native mode
+	 * - 1 -- 8 channels
+	 * Native mode indicates that encoding must be performed with the number
+	 * of channels at the input.
+	 */
+	uint16_t  bits_per_sample;
+	/*
+	 * Number of bits per sample per channel.
+	 * @values 16, 24
+	 */
+	uint32_t  sample_rate;
+	/*
+	 * Number of samples per second.
+	 * @values 0, 8000 to 48000 Hz
+	 * A value of 0 indicates the native sampling rate. Encoding is
+	 * performed at the input sampling rate.
+	 */
+	uint16_t  is_signed;
+	/*
+	 * Flag that indicates the PCM samples are signed (1). Currently, only
+	 * signed PCM samples are supported.
+	 */
+	uint16_t    sample_word_size;
+	/*
+	 * The size in bits of the word that holds a sample of a channel.
+	 * @values 16, 24, 32
+	 * 16-bit samples are always placed in 16-bit words:
+	 * sample_word_size = 1.
+	 * 24-bit samples can be placed in 32-bit words or in consecutive
+	 * 24-bit words.
+	 * - If sample_word_size = 32, 24-bit samples are placed in the
+	 * most significant 24 bits of a 32-bit word.
+	 * - If sample_word_size = 24, 24-bit samples are placed in
+	 * 24-bit words. @tablebulletend
+	 */
+	uint8_t   channel_mapping[8];
+	/*
+	 * Channel mapping array expected at the encoder output.
+	 *  Channel[i] mapping describes channel i inside the buffer, where
+	 *  0 @le i < num_channels. All valid used channels must be present at
+	 *  the beginning of the array.
+	 * If Native mode is set for the channels, this field is ignored.
+	 * @values See Section @xref{dox:PcmChannelDefs}
+	 */
+	uint16_t                endianness;
+	/*
+	 * Flag to indicate the endianness of the pcm sample
+	 * Supported values: 0 - Little endian (all other formats)
+	 *                   1 - Big endian (AIFF)
+	 */
+	uint16_t                mode;
+	/*
+	 * Mode to provide additional info about the pcm input data.
+	 * Supported values: 0 - Default QFs (Q15 for 16b, Q23 for packed 24b,
+	 *                       Q31 for unpacked 24b or 32b)
+	 *                  15 - for 16 bit
+	 *                  23 - for 24b packed or 8.24 format
+	 *                  31 - for 24b unpacked or 32bit
+	 */
+} __packed;
 
 /*
  * Payload of the multichannel PCM encoder configuration parameters in
diff --git a/include/sound/q6asm-v2.h b/include/sound/q6asm-v2.h
index 00129eb08888..f08bd73edb59 100644
--- a/include/sound/q6asm-v2.h
+++ b/include/sound/q6asm-v2.h
@@ -97,6 +97,24 @@
 #define ASM_SHIFT_GAPLESS_MODE_FLAG	31
 #define ASM_SHIFT_LAST_BUFFER_FLAG	30
 
+#define ASM_LITTLE_ENDIAN 0
+#define ASM_BIG_ENDIAN 1
+
+/* PCM_MEDIA_FORMAT_Version */
+enum {
+	PCM_MEDIA_FORMAT_V2 = 0,
+	PCM_MEDIA_FORMAT_V3,
+	PCM_MEDIA_FORMAT_V4,
+};
+
+/* PCM format modes in DSP */
+enum {
+	DEFAULT_QF = 0,
+	Q15 = 15,
+	Q23 = 23,
+	Q31 = 31,
+};
+
 /* payload structure bytes */
 #define READDONE_IDX_STATUS 0
 #define READDONE_IDX_BUFADD_LSW 1
@@ -245,6 +263,9 @@ int q6asm_open_read_v2(struct audio_client *ac, uint32_t format,
 int q6asm_open_read_v3(struct audio_client *ac, uint32_t format,
 		       uint16_t bits_per_sample);
 
+int q6asm_open_read_v4(struct audio_client *ac, uint32_t format,
+		       uint16_t bits_per_sample);
+
 int q6asm_open_write(struct audio_client *ac, uint32_t format
 		/*, uint16_t bits_per_sample*/);
 
@@ -257,6 +278,9 @@ int q6asm_open_shared_io(struct audio_client *ac,
 int q6asm_open_write_v3(struct audio_client *ac, uint32_t format,
 			uint16_t bits_per_sample);
 
+int q6asm_open_write_v4(struct audio_client *ac, uint32_t format,
+			uint16_t bits_per_sample);
+
 int q6asm_stream_open_write_v2(struct audio_client *ac, uint32_t format,
 			       uint16_t bits_per_sample, int32_t stream_id,
 			       bool is_gapless_mode);
@@ -265,6 +289,10 @@ int q6asm_stream_open_write_v3(struct audio_client *ac, uint32_t format,
 			       uint16_t bits_per_sample, int32_t stream_id,
 			       bool is_gapless_mode);
 
+int q6asm_stream_open_write_v4(struct audio_client *ac, uint32_t format,
+			       uint16_t bits_per_sample, int32_t stream_id,
+			       bool is_gapless_mode);
+
 int q6asm_open_write_compressed(struct audio_client *ac, uint32_t format,
 				uint32_t passthrough_flag);
 
@@ -369,6 +397,13 @@ int q6asm_enc_cfg_blk_pcm_v3(struct audio_client *ac,
 			     bool use_back_flavor, u8 *channel_map,
 			     uint16_t sample_word_size);
 
+int q6asm_enc_cfg_blk_pcm_v4(struct audio_client *ac,
+			     uint32_t rate, uint32_t channels,
+			     uint16_t bits_per_sample, bool use_default_chmap,
+			     bool use_back_flavor, u8 *channel_map,
+			     uint16_t sample_word_size, uint16_t endianness,
+			     uint16_t mode);
+
 int q6asm_enc_cfg_blk_pcm_format_support(struct audio_client *ac,
 			uint32_t rate, uint32_t channels,
 			uint16_t bits_per_sample);
@@ -378,6 +413,13 @@ int q6asm_enc_cfg_blk_pcm_format_support_v3(struct audio_client *ac,
 					    uint16_t bits_per_sample,
 					    uint16_t sample_word_size);
 
+int q6asm_enc_cfg_blk_pcm_format_support_v4(struct audio_client *ac,
+					    uint32_t rate, uint32_t channels,
+					    uint16_t bits_per_sample,
+					    uint16_t sample_word_size,
+					    uint16_t endianness,
+					    uint16_t mode);
+
 int q6asm_set_encdec_chan_map(struct audio_client *ac,
 		uint32_t num_channels);
 
@@ -427,6 +469,17 @@ int q6asm_media_format_block_pcm_format_support_v3(struct audio_client *ac,
 						   char *channel_map,
 						   uint16_t sample_word_size);
 
+int q6asm_media_format_block_pcm_format_support_v4(struct audio_client *ac,
+						   uint32_t rate,
+						   uint32_t channels,
+						   uint16_t bits_per_sample,
+						   int stream_id,
+						   bool use_default_chmap,
+						   char *channel_map,
+						   uint16_t sample_word_size,
+						   uint16_t endianness,
+						   uint16_t mode);
+
 int q6asm_media_format_block_multi_ch_pcm(struct audio_client *ac,
 			uint32_t rate, uint32_t channels,
 			bool use_default_chmap, char *channel_map);
@@ -444,6 +497,15 @@ int q6asm_media_format_block_multi_ch_pcm_v3(struct audio_client *ac,
 					     uint16_t bits_per_sample,
 					     uint16_t sample_word_size);
 
+int q6asm_media_format_block_multi_ch_pcm_v4(struct audio_client *ac,
+					     uint32_t rate, uint32_t channels,
+					     bool use_default_chmap,
+					     char *channel_map,
+					     uint16_t bits_per_sample,
+					     uint16_t sample_word_size,
+					     uint16_t endianness,
+					     uint16_t mode);
+
 int q6asm_media_format_block_aac(struct audio_client *ac,
 			struct asm_aac_cfg *cfg);