From efb830ca3e89e4ed742de99f6ca583a88e4cbd6d Mon Sep 17 00:00:00 2001 From: Derek Chen Date: Thu, 24 Mar 2016 14:29:12 -0400 Subject: [PATCH] ASoC: msm: qdsp6v2: Add ASM low latency loopback support Add ASM low latency loopback support. ICC (in-car communication) use case needs the low latency ASM loopback to satisfy the latency requirement. CRs-fixed: 998118 Change-Id: If225e809072f4296bc22028da29e589137e5799d Signed-off-by: Derek Chen --- .../bindings/sound/qcom-audio-dev.txt | 16 ++- include/sound/apr_audio-v2.h | 132 ++++++++++++++++++ sound/soc/msm/qdsp6v2/msm-pcm-loopback-v2.c | 31 +++- sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c | 63 +++++++++ sound/soc/msm/qdsp6v2/q6asm.c | 78 ++++++++--- 5 files changed, 297 insertions(+), 23 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt b/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt index e8177c3a0952..7126c516c99a 100644 --- a/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt +++ b/Documentation/devicetree/bindings/sound/qcom-audio-dev.txt @@ -177,6 +177,11 @@ Required properties: - compatible : "qcom,msm-pcm-loopback" +Optional properties: + + - qcom,msm-pcm-loopback-low-latency : Flag indicating whether + the device node is of type low latency. + * msm-dai-q6 [First Level Nodes] @@ -415,6 +420,11 @@ Example: qcom,msm-pcm-low-latency; }; + qcom,msm-pcm-loopback-low-latency { + compatible = "qcom,msm-pcm-loopback"; + qcom,msm-pcm-loopback-low-latency; + }; + qcom,msm-pcm-routing { compatible = "qcom,msm-pcm-routing"; }; @@ -2116,13 +2126,15 @@ Example: asoc-platform = <&pcm0>, <&pcm1>, <&pcm2>, <&voip>, <&voice>, <&loopback>, <&compress>, <&hostless>, - <&afe>, <&lsm>, <&routing>, <&compr>; + <&afe>, <&lsm>, <&routing>, <&compr>, + <&loopback1>; asoc-platform-names = "msm-pcm-dsp.0", "msm-pcm-dsp.1", "msm-pcm-dsp.2", "msm-voip-dsp", "msm-pcm-voice", "msm-pcm-loopback", "msm-compress-dsp", "msm-pcm-hostless", "msm-pcm-afe", "msm-lsm-client", - "msm-pcm-routing", "msm-compr-dsp"; + "msm-pcm-routing", "msm-compr-dsp", + "msm-pcm-loopback.1"; asoc-cpu = <&dai_pri_auxpcm>, <&dai_sec_auxpcm>, <&dai_hdmi>, <&dai_mi2s>, <&dai_mi2s_quat>, <&afe_pcm_rx>, <&afe_pcm_tx>, diff --git a/include/sound/apr_audio-v2.h b/include/sound/apr_audio-v2.h index e098e2329ac6..ab77f5d99034 100644 --- a/include/sound/apr_audio-v2.h +++ b/include/sound/apr_audio-v2.h @@ -5812,6 +5812,138 @@ struct asm_stream_cmd_open_loopback_v2 { /* Reserved for future use. This field must be set to zero. */ } __packed; + +#define ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK 0x00010DBA + +/* Bitmask for the stream's Performance mode. */ +#define ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK \ + (0x70000000UL) + +/* Bit shift for the stream's Performance mode. */ +#define ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK 28 + +/* Bitmask for the decoder converter enable flag. */ +#define ASM_BIT_MASK_DECODER_CONVERTER_FLAG (0x00000078UL) + +/* Shift value for the decoder converter enable flag. */ +#define ASM_SHIFT_DECODER_CONVERTER_FLAG 3 + +/* Converter mode is None (Default). */ +#define ASM_CONVERTER_MODE_NONE 0 + +/* Converter mode is DDP-to-DD. */ +#define ASM_DDP_DD_CONVERTER_MODE 1 + +/* Identifies a special converter mode where source and sink formats + * are the same but postprocessing must applied. Therefore, Decode + * @rarrow Re-encode is necessary. + */ +#define ASM_POST_PROCESS_CONVERTER_MODE 2 + + +struct asm_stream_cmd_open_transcode_loopback_t { + struct apr_hdr hdr; + u32 mode_flags; +/* Mode Flags specifies the performance mode in which this stream + * is to be opened. + * Supported values{for bits 30 to 28}(stream_perf_mode flag) + * + * #ASM_LEGACY_STREAM_SESSION -- This mode ensures backward + * compatibility to the original behavior + * of ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK + * + * #ASM_LOW_LATENCY_STREAM_SESSION -- Opens a loopback session by using + * shortened buffers in low latency POPP + * - Recommendation: Do not enable high latency algorithms. They might + * negate the benefits of opening a low latency stream, and they + * might also suffer quality degradation from unexpected jitter. + * - This Low Latency mode is supported only for PCM In and PCM Out + * loopbacks. An error is returned if Low Latency mode is opened for + * other transcode loopback modes. + * - To configure this subfield, use + * ASM_BIT_MASK_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK and + * ASM_SHIFT_STREAM_PERF_MODE_FLAG_IN_OPEN_TRANSCODE_LOOPBACK. + * + * Supported values{for bits 6 to 3} (decoder-converter compatibility) + * #ASM_CONVERTER_MODE_NONE (0x0) -- Default + * #ASM_DDP_DD_CONVERTER_MODE (0x1) + * #ASM_POST_PROCESS_CONVERTER_MODE (0x2) + * 0x3-0xF -- Reserved for future use + * - Use #ASM_BIT_MASK_DECODER_CONVERTER_FLAG and + * ASM_SHIFT_DECODER_CONVERTER_FLAG to set this bit + * All other bits are reserved; clients must set them to 0. + */ + + u32 src_format_id; +/* Specifies the media format of the input audio stream. + * + * Supported values + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 + * - #ASM_MEDIA_FMT_DTS + * - #ASM_MEDIA_FMT_EAC3_DEC + * - #ASM_MEDIA_FMT_EAC3 + * - #ASM_MEDIA_FMT_AC3_DEC + * - #ASM_MEDIA_FMT_AC3 + */ + u32 sink_format_id; +/* Specifies the media format of the output stream. + * + * Supported values + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2 + * - #ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V3 + * - #ASM_MEDIA_FMT_DTS (not supported in Low Latency mode) + * - #ASM_MEDIA_FMT_EAC3_DEC (not supported in Low Latency mode) + * - #ASM_MEDIA_FMT_EAC3 (not supported in Low Latency mode) + * - #ASM_MEDIA_FMT_AC3_DEC (not supported in Low Latency mode) + * - #ASM_MEDIA_FMT_AC3 (not supported in Low Latency mode) + */ + + u32 audproc_topo_id; +/* Postprocessing topology ID, which specifies the topology (order of + * processing) of postprocessing algorithms. + * + * Supported values + * - #ASM_STREAM_POSTPROC_TOPO_ID_DEFAULT + * - #ASM_STREAM_POSTPROC_TOPO_ID_PEAKMETER + * - #ASM_STREAM_POSTPROC_TOPO_ID_MCH_PEAK_VOL + * - #ASM_STREAM_POSTPROC_TOPO_ID_NONE + * Topologies can be added through #ASM_CMD_ADD_TOPOLOGIES. + * This field is ignored for the Converter mode, in which no + * postprocessing is performed. + */ + + u16 src_endpoint_type; +/* Specifies the source endpoint that provides the input samples. + * + * Supported values + * - 0 -- Tx device matrix or stream router (gateway to the hardware + * ports) + * - All other values are reserved + * Clients must set this field to 0. Otherwise, an error is returned. + */ + + u16 sink_endpoint_type; +/* Specifies the sink endpoint type. + * + * Supported values + * - 0 -- Rx device matrix or stream router (gateway to the hardware + * ports) + * - All other values are reserved + * Clients must set this field to 0. Otherwise, an error is returned. + */ + + u16 bits_per_sample; +/* Number of bits per sample processed by the ASM modules. + * Supported values 16, 24 + */ + + u16 reserved; +/* This field must be set to 0. + */ +} __packed; + + #define ASM_STREAM_CMD_CLOSE 0x00010BCD #define ASM_STREAM_CMD_FLUSH 0x00010BCE diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-loopback-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-loopback-v2.c index 82458275b892..bd1468beedf6 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-loopback-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-loopback-v2.c @@ -25,6 +25,7 @@ #include #include #include +#include #include "msm-pcm-routing-v2.h" @@ -67,6 +68,10 @@ static struct fe_dai_session_map session_map[LOOPBACK_SESSION_MAX] = { static u32 hfp_tx_mute; +struct msm_pcm_pdata { + int perf_mode; +}; + static void stop_pcm(struct msm_pcm_loopback *pcm); static int msm_pcm_loopback_get_session(struct snd_soc_pcm_runtime *rtd, struct msm_pcm_loopback **pcm); @@ -244,6 +249,7 @@ static int msm_pcm_open(struct snd_pcm_substream *substream) struct msm_pcm_routing_evt event; struct asm_session_mtmx_strtr_param_window_v2_t asm_mtmx_strtr_window; uint32_t param_id; + struct msm_pcm_pdata *pdata; ret = msm_pcm_loopback_get_session(rtd, &pcm); if (ret) @@ -269,6 +275,15 @@ static int msm_pcm_open(struct snd_pcm_substream *substream) if (pcm->audio_client != NULL) stop_pcm(pcm); + pdata = (struct msm_pcm_pdata *) + dev_get_drvdata(rtd->platform->dev); + if (!pdata) { + dev_err(rtd->platform->dev, + "%s: platform data not populated\n", __func__); + mutex_unlock(&pcm->lock); + return -EINVAL; + } + pcm->audio_client = q6asm_audio_client_alloc( (app_cb)msm_pcm_loopback_event_handler, pcm); if (!pcm->audio_client) { @@ -278,7 +293,7 @@ static int msm_pcm_open(struct snd_pcm_substream *substream) return -ENOMEM; } pcm->session_id = pcm->audio_client->session; - pcm->audio_client->perf_mode = false; + pcm->audio_client->perf_mode = pdata->perf_mode; ret = q6asm_open_loopback_v2(pcm->audio_client, bits_per_sample); if (ret < 0) { @@ -745,9 +760,23 @@ static struct snd_soc_platform_driver msm_soc_platform = { static int msm_pcm_probe(struct platform_device *pdev) { + struct msm_pcm_pdata *pdata; + dev_dbg(&pdev->dev, "%s: dev name %s\n", __func__, dev_name(&pdev->dev)); + pdata = kzalloc(sizeof(struct msm_pcm_pdata), GFP_KERNEL); + if (!pdata) + return -ENOMEM; + + if (of_property_read_bool(pdev->dev.of_node, + "qcom,msm-pcm-loopback-low-latency")) + pdata->perf_mode = LOW_LATENCY_PCM_MODE; + else + pdata->perf_mode = LEGACY_PCM_MODE; + + dev_set_drvdata(&pdev->dev, pdata); + return snd_soc_register_platform(&pdev->dev, &msm_soc_platform); } diff --git a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c index 03994225edcf..d2666f46c8dc 100644 --- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c +++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c @@ -5242,6 +5242,57 @@ static const struct snd_kcontrol_new mmul8_mixer_controls[] = { msm_routing_put_audio_mixer), }; +static const struct snd_kcontrol_new mmul9_mixer_controls[] = { + SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("PRI_MI2S_TX", MSM_BACKEND_DAI_PRI_MI2S_TX, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_FM_TX", MSM_BACKEND_DAI_INT_FM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("INTERNAL_BT_SCO_TX", MSM_BACKEND_DAI_INT_BT_SCO_TX, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("AFE_PCM_TX", MSM_BACKEND_DAI_AFE_PCM_TX, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_DL", MSM_BACKEND_DAI_INCALL_RECORD_RX, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("VOC_REC_UL", MSM_BACKEND_DAI_INCALL_RECORD_TX, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("SLIM_6_TX", MSM_BACKEND_DAI_SLIMBUS_6_TX, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_0", MSM_BACKEND_DAI_TERT_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_1", MSM_BACKEND_DAI_TERT_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_2", MSM_BACKEND_DAI_TERT_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("TERT_TDM_TX_3", MSM_BACKEND_DAI_TERT_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_0", MSM_BACKEND_DAI_QUAT_TDM_TX_0, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_1", MSM_BACKEND_DAI_QUAT_TDM_TX_1, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_2", MSM_BACKEND_DAI_QUAT_TDM_TX_2, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), + SOC_SINGLE_EXT("QUAT_TDM_TX_3", MSM_BACKEND_DAI_QUAT_TDM_TX_3, + MSM_FRONTEND_DAI_MULTIMEDIA9, 1, 0, msm_routing_get_audio_mixer, + msm_routing_put_audio_mixer), +}; + static const struct snd_kcontrol_new mmul17_mixer_controls[] = { SOC_SINGLE_EXT("SLIM_0_TX", MSM_BACKEND_DAI_SLIMBUS_0_TX, MSM_FRONTEND_DAI_MULTIMEDIA17, 1, 0, msm_routing_get_audio_mixer, @@ -9210,6 +9261,8 @@ static const struct snd_soc_dapm_widget msm_qdsp6_widgets[] = { mmul6_mixer_controls, ARRAY_SIZE(mmul6_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia8 Mixer", SND_SOC_NOPM, 0, 0, mmul8_mixer_controls, ARRAY_SIZE(mmul8_mixer_controls)), + SND_SOC_DAPM_MIXER("MultiMedia9 Mixer", SND_SOC_NOPM, 0, 0, + mmul9_mixer_controls, ARRAY_SIZE(mmul9_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia17 Mixer", SND_SOC_NOPM, 0, 0, mmul17_mixer_controls, ARRAY_SIZE(mmul17_mixer_controls)), SND_SOC_DAPM_MIXER("MultiMedia18 Mixer", SND_SOC_NOPM, 0, 0, @@ -10251,6 +10304,15 @@ static const struct snd_soc_dapm_route intercon[] = { {"MultiMedia8 Mixer", "QUAT_TDM_TX_2", "QUAT_TDM_TX_2"}, {"MultiMedia8 Mixer", "QUAT_TDM_TX_3", "QUAT_TDM_TX_3"}, + {"MultiMedia9 Mixer", "TERT_TDM_TX_0", "TERT_TDM_TX_0"}, + {"MultiMedia9 Mixer", "TERT_TDM_TX_1", "TERT_TDM_TX_1"}, + {"MultiMedia9 Mixer", "TERT_TDM_TX_2", "TERT_TDM_TX_2"}, + {"MultiMedia9 Mixer", "TERT_TDM_TX_3", "TERT_TDM_TX_3"}, + {"MultiMedia9 Mixer", "QUAT_TDM_TX_0", "QUAT_TDM_TX_0"}, + {"MultiMedia9 Mixer", "QUAT_TDM_TX_1", "QUAT_TDM_TX_1"}, + {"MultiMedia9 Mixer", "QUAT_TDM_TX_2", "QUAT_TDM_TX_2"}, + {"MultiMedia9 Mixer", "QUAT_TDM_TX_3", "QUAT_TDM_TX_3"}, + {"MultiMedia1 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, {"MultiMedia2 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, {"MultiMedia4 Mixer", "USB_AUDIO_TX", "USB_AUDIO_TX"}, @@ -10365,6 +10427,7 @@ static const struct snd_soc_dapm_route intercon[] = { {"MM_UL5", NULL, "MultiMedia5 Mixer"}, {"MM_UL6", NULL, "MultiMedia6 Mixer"}, {"MM_UL8", NULL, "MultiMedia8 Mixer"}, + {"MM_UL9", NULL, "MultiMedia9 Mixer"}, {"MM_UL17", NULL, "MultiMedia17 Mixer"}, {"MM_UL18", NULL, "MultiMedia18 Mixer"}, {"MM_UL19", NULL, "MultiMedia19 Mixer"}, diff --git a/sound/soc/msm/qdsp6v2/q6asm.c b/sound/soc/msm/qdsp6v2/q6asm.c index 19105ffd9d4a..670e8f8ea84b 100644 --- a/sound/soc/msm/qdsp6v2/q6asm.c +++ b/sound/soc/msm/qdsp6v2/q6asm.c @@ -1710,6 +1710,7 @@ static int32_t q6asm_callback(struct apr_client_data *data, void *priv) case ASM_STREAM_CMD_OPEN_PUSH_MODE_READ: case ASM_STREAM_CMD_OPEN_READWRITE_V2: case ASM_STREAM_CMD_OPEN_LOOPBACK_V2: + case ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK: case ASM_DATA_CMD_MEDIA_FMT_UPDATE_V2: case ASM_STREAM_CMD_SET_ENCDEC_PARAM: case ASM_DATA_CMD_REMOVE_INITIAL_SILENCE: @@ -2982,7 +2983,6 @@ int q6asm_open_read_write_v2(struct audio_client *ac, uint32_t rd_format, int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample) { int rc = 0x00; - struct asm_stream_cmd_open_loopback_v2 open; if (ac == NULL) { pr_err("%s: APR handle NULL\n", __func__); @@ -2994,29 +2994,67 @@ int q6asm_open_loopback_v2(struct audio_client *ac, uint16_t bits_per_sample) } pr_debug("%s: session[%d]\n", __func__, ac->session); - q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE); - atomic_set(&ac->cmd_state, -1); - open.hdr.opcode = ASM_STREAM_CMD_OPEN_LOOPBACK_V2; + if (ac->perf_mode == LOW_LATENCY_PCM_MODE) { + struct asm_stream_cmd_open_transcode_loopback_t open; - open.mode_flags = 0; - open.src_endpointype = 0; - open.sink_endpointype = 0; - /* source endpoint : matrix */ - open.postprocopo_id = q6asm_get_asm_topology_cal(); + q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE); + atomic_set(&ac->cmd_state, -1); + open.hdr.opcode = ASM_STREAM_CMD_OPEN_TRANSCODE_LOOPBACK; - ac->app_type = q6asm_get_asm_app_type_cal(); - ac->topology = open.postprocopo_id; - open.bits_per_sample = bits_per_sample; - open.reserved = 0; + open.mode_flags = 0; + open.src_endpoint_type = 0; + open.sink_endpoint_type = 0; + open.src_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2; + open.sink_format_id = ASM_MEDIA_FMT_MULTI_CHANNEL_PCM_V2; + /* source endpoint : matrix */ + open.audproc_topo_id = q6asm_get_asm_topology_cal(); - rc = apr_send_pkt(ac->apr, (uint32_t *) &open); - if (rc < 0) { - pr_err("%s: open failed op[0x%x]rc[%d]\n", __func__, - open.hdr.opcode, rc); - rc = -EINVAL; - goto fail_cmd; + ac->app_type = q6asm_get_asm_app_type_cal(); + if (ac->perf_mode == LOW_LATENCY_PCM_MODE) + open.mode_flags |= ASM_LOW_LATENCY_STREAM_SESSION; + else + open.mode_flags |= ASM_LEGACY_STREAM_SESSION; + ac->topology = open.audproc_topo_id; + open.bits_per_sample = bits_per_sample; + open.reserved = 0; + pr_debug("%s: opening a transcode_loopback with mode_flags =[%d] session[%d]\n", + __func__, open.mode_flags, ac->session); + + rc = apr_send_pkt(ac->apr, (uint32_t *) &open); + if (rc < 0) { + pr_err("%s: open failed op[0x%x]rc[%d]\n", + __func__, open.hdr.opcode, rc); + rc = -EINVAL; + goto fail_cmd; + } + } else {/*if(ac->perf_mode == LEGACY_PCM_MODE)*/ + struct asm_stream_cmd_open_loopback_v2 open; + + q6asm_add_hdr(ac, &open.hdr, sizeof(open), TRUE); + atomic_set(&ac->cmd_state, -1); + open.hdr.opcode = ASM_STREAM_CMD_OPEN_LOOPBACK_V2; + + open.mode_flags = 0; + open.src_endpointype = 0; + open.sink_endpointype = 0; + /* source endpoint : matrix */ + open.postprocopo_id = q6asm_get_asm_topology_cal(); + + ac->app_type = q6asm_get_asm_app_type_cal(); + ac->topology = open.postprocopo_id; + open.bits_per_sample = bits_per_sample; + open.reserved = 0; + pr_debug("%s: opening a loopback_v2 with mode_flags =[%d] session[%d]\n", + __func__, open.mode_flags, ac->session); + + rc = apr_send_pkt(ac->apr, (uint32_t *) &open); + if (rc < 0) { + pr_err("%s: open failed op[0x%x]rc[%d]\n", + __func__, open.hdr.opcode, rc); + rc = -EINVAL; + goto fail_cmd; + } } - rc = wait_event_timeout(ac->cmd_wait, (atomic_read(&ac->cmd_state) >= 0), 5*HZ); if (!rc) {