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-rw-r--r--Documentation/devicetree/bindings/sound/qcom-audio-dev.txt16
-rw-r--r--include/sound/apr_audio-v2.h132
-rw-r--r--sound/soc/msm/qdsp6v2/msm-pcm-loopback-v2.c31
-rw-r--r--sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c63
-rw-r--r--sound/soc/msm/qdsp6v2/q6asm.c78
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 f79f49608743..3687ce33b700 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";
};
@@ -2121,13 +2131,15 @@ 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 ceba9f7d759a..e75f3319f158 100644
--- a/include/sound/apr_audio-v2.h
+++ b/include/sound/apr_audio-v2.h
@@ -5955,6 +5955,138 @@ 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 <sound/control.h>
#include <sound/tlv.h>
#include <asm/dma.h>
+#include <sound/q6audio-v2.h>
#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 41f68e05e075..8feaf1fe5e2f 100644
--- a/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c
+++ b/sound/soc/msm/qdsp6v2/msm-pcm-routing-v2.c
@@ -5257,6 +5257,57 @@ 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,
@@ -9225,6 +9276,8 @@ 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,
@@ -10260,6 +10313,15 @@ 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"},
@@ -10374,6 +10436,7 @@ 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) {
generated by cgit v1.2.3 (git 2.25.1) at 2025-12-24 13:44:01 +0000