ASoC: msm: qdsp6v2: support decode of g711 clips

Add support in audio stream manager to configure decode
of g711 related clips.

CRs-Fixed: 1094107
Change-Id: Ie90fd68e24e7e793aaac64290e3c1e41682d6d5a
Signed-off-by: Yamit Mehta <ymehta@codeaurora.org>
Signed-off-by: Surendar karka <sukark@codeaurora.org>

include/sound/apr_audio-v2.h

@@ -3581,6 +3581,10 @@ struct asm_alac_cfg {
 	u32 channel_layout_tag;
 };
 
+struct asm_g711_dec_cfg {
+	u32 sample_rate;
+};
+
 struct asm_vorbis_cfg {
 	u32 bit_stream_fmt;
 };
@@ -4309,6 +4313,12 @@ struct asm_alac_fmt_blk_v2 {
 
 } __packed;
 
+struct asm_g711_dec_fmt_blk_v2 {
+	struct apr_hdr hdr;
+	struct asm_data_cmd_media_fmt_update_v2 fmtblk;
+	u32 sample_rate;
+} __packed;
+
 struct asm_ape_fmt_blk_v2 {
 	struct apr_hdr hdr;
 	struct asm_data_cmd_media_fmt_update_v2 fmtblk;

include/sound/q6asm-v2.h

@@ -534,6 +534,9 @@ int q6asm_stream_media_format_block_flac(struct audio_client *ac,
 int q6asm_media_format_block_alac(struct audio_client *ac,
 			struct asm_alac_cfg *cfg, int stream_id);
 
+int q6asm_media_format_block_g711(struct audio_client *ac,
+			struct asm_g711_dec_cfg *cfg, int stream_id);
+
 int q6asm_stream_media_format_block_vorbis(struct audio_client *ac,
 			struct asm_vorbis_cfg *cfg, int stream_id);
 

sound/soc/msm/qdsp6v2/q6asm.c

@@ -2863,6 +2863,11 @@ static int __q6asm_open_read_write(struct audio_client *ac, uint32_t rd_format,
 		break;
 	case FORMAT_DSD:
 		open.dec_fmt_id = ASM_MEDIA_FMT_DSD;
+	case FORMAT_G711_ALAW_FS:
+		open.dec_fmt_id = ASM_MEDIA_FMT_G711_ALAW_FS;
+		break;
+	case FORMAT_G711_MLAW_FS:
+		open.dec_fmt_id = ASM_MEDIA_FMT_G711_MLAW_FS;
 		break;
 	default:
 		pr_err("%s: Invalid format 0x%x\n",
@@ -5485,6 +5490,75 @@ fail_cmd:
 	return rc;
 }
 
+/*
+ * q6asm_media_format_block_g711 - sends g711 decoder configuration
+ *                                            parameters
+ * @ac: Client session handle
+ * @cfg: Audio stream manager configuration parameters
+ * @stream_id: Stream id
+ */
+int q6asm_media_format_block_g711(struct audio_client *ac,
+				struct asm_g711_dec_cfg *cfg, int stream_id)
+{
+	struct asm_g711_dec_fmt_blk_v2 fmt;
+	int rc = 0;
+
+	if (!ac) {
+		pr_err("%s: audio client is null\n", __func__);
+		return -EINVAL;
+	}
+	if (!cfg) {
+		pr_err("%s: Invalid ASM config\n", __func__);
+		return -EINVAL;
+	}
+
+	if (stream_id <= 0) {
+		pr_err("%s: Invalid stream id\n", __func__);
+		return -EINVAL;
+	}
+
+	pr_debug("%s :session[%d]rate[%d]\n", __func__,
+		ac->session, cfg->sample_rate);
+
+	memset(&fmt, 0, sizeof(struct asm_g711_dec_fmt_blk_v2));
+
+	q6asm_stream_add_hdr(ac, &fmt.hdr, sizeof(fmt), TRUE, stream_id);
+	atomic_set(&ac->cmd_state, -1);
+
+	fmt.hdr.opcode = ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2;
+	fmt.fmtblk.fmt_blk_size = sizeof(fmt) - sizeof(fmt.hdr) -
+						sizeof(fmt.fmtblk);
+
+	fmt.sample_rate = cfg->sample_rate;
+
+	rc = apr_send_pkt(ac->apr, (uint32_t *) &fmt);
+	if (rc < 0) {
+		pr_err("%s :Command media format update failed %d\n",
+				__func__, rc);
+		goto fail_cmd;
+	}
+	rc = wait_event_timeout(ac->cmd_wait,
+				(atomic_read(&ac->cmd_state) >= 0), 5*HZ);
+	if (!rc) {
+		pr_err("%s :timeout. waited for FORMAT_UPDATE\n", __func__);
+		rc = -ETIMEDOUT;
+		goto fail_cmd;
+	}
+
+	if (atomic_read(&ac->cmd_state) > 0) {
+		pr_err("%s: DSP returned error[%s]\n",
+				__func__, adsp_err_get_err_str(
+				atomic_read(&ac->cmd_state)));
+		rc = adsp_err_get_lnx_err_code(
+				atomic_read(&ac->cmd_state));
+		goto fail_cmd;
+	}
+	return 0;
+fail_cmd:
+	return rc;
+}
+EXPORT_SYMBOL(q6asm_media_format_block_g711);
+
 int q6asm_stream_media_format_block_vorbis(struct audio_client *ac,
 				struct asm_vorbis_cfg *cfg, int stream_id)
 {