Fix: st30_pipeline don't close until mbuf is free

When tx_st30p_frame_done is called and subsequently framebuff->stat is set to ST30P_TX_FRAME_FREE, data from the framebuffer can still be in transport, already packetized and copied into rte_mbuf, waiting to be sent. Added sleep after last packet status is set to ST30P_TX_FRAME_FREE Signed-off-by: Kasiewicz, Marek <[email protected]>
OpenVisualCloud · Jan 30, 2025 · 70fd7ff · 70fd7ff
1 parent 6b056c8
commit 70fd7ff
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Showing 5 changed files with 51 additions and 33 deletions.
diff --git a/ecosystem/gstreamer_plugin/gst_mtl_common.c b/ecosystem/gstreamer_plugin/gst_mtl_common.c
@@ -193,7 +193,8 @@ gboolean gst_mtl_common_parse_audio_format(const char* format, enum st30_fmt* au
   return TRUE;
 }
 
-gboolean gst_mtl_common_parse_sampling(gint sampling, enum st30_sampling* st_sampling) {
+gboolean gst_mtl_common_gst_to_st_sampling(gint sampling,
+                                           enum st30_sampling* st_sampling) {
   if (!st_sampling) {
     GST_ERROR("Invalid st_sampling pointer");
     return FALSE;
@@ -210,6 +211,30 @@ gboolean gst_mtl_common_parse_sampling(gint sampling, enum st30_sampling* st_sam
       *st_sampling = ST30_SAMPLING_96K;
       return TRUE;
     default:
+      GST_ERROR("Unsupported sampling value");
+      return FALSE;
+  }
+}
+
+gboolean gst_mtl_common_st_to_gst_sampling(enum st30_sampling st_sampling,
+                                           gint* gst_sampling) {
+  if (!gst_sampling) {
+    GST_ERROR("Invalid gst_sampling pointer");
+    return FALSE;
+  }
+
+  switch (st_sampling) {
+    case ST31_SAMPLING_44K:
+      *gst_sampling = GST_MTL_SUPPORTED_AUDIO_SAMPLING_44_1K;
+      return TRUE;
+    case ST30_SAMPLING_48K:
+      *gst_sampling = GST_MTL_SUPPORTED_AUDIO_SAMPLING_48K;
+      return TRUE;
+    case ST30_SAMPLING_96K:
+      *gst_sampling = GST_MTL_SUPPORTED_AUDIO_SAMPLING_96K;
+      return TRUE;
+    default:
+      GST_ERROR("Unsupported st_sampling value");
       return FALSE;
   }
 }

diff --git a/ecosystem/gstreamer_plugin/gst_mtl_common.h b/ecosystem/gstreamer_plugin/gst_mtl_common.h
@@ -84,7 +84,10 @@ gboolean gst_mtl_common_parse_fps_code(gint fps_code, enum st_fps* fps);
 gboolean gst_mtl_common_parse_pixel_format(const char* format, enum st_frame_fmt* fmt);
 
 gboolean gst_mtl_common_parse_audio_format(const char* format, enum st30_fmt* audio);
-gboolean gst_mtl_common_parse_sampling(gint sampling, enum st30_sampling* st_sampling);
+gboolean gst_mtl_common_gst_to_st_sampling(gint sampling,
+                                           enum st30_sampling* st_sampling);
+gboolean gst_mtl_common_st_to_gst_sampling(enum st30_sampling st_sampling,
+                                           gint* gst_sampling);
 
 gboolean gst_mtl_common_parse_dev_arguments(struct mtl_init_params* mtl_init_params,
                                             StDevArgs* devArgs);

diff --git a/ecosystem/gstreamer_plugin/gst_mtl_st30p_rx.c b/ecosystem/gstreamer_plugin/gst_mtl_st30p_rx.c
@@ -208,7 +208,7 @@ static gboolean gst_mtl_st30p_rx_start(GstBaseSrc* basesrc) {
   ops_rx->ptime = ST30_PTIME_1MS;
   ops_rx->flags |= ST30P_RX_FLAG_BLOCK_GET;
 
-  if (!gst_mtl_common_parse_sampling(src->sampling, &ops_rx->sampling)) {
+  if (!gst_mtl_common_gst_to_st_sampling(src->sampling, &ops_rx->sampling)) {
     GST_ERROR("Failed to parse ops_rx sampling %d", src->sampling);
     return FALSE;
   }
@@ -353,25 +353,32 @@ static void gst_mtl_st30p_rx_get_property(GObject* object, guint prop_id, GValue
  */
 
 static gboolean gst_mtl_st30p_rx_negotiate(GstBaseSrc* basesrc) {
-  GstAudioInfo* info;
   Gst_Mtl_St30p_Rx* src = GST_MTL_ST30P_RX(basesrc);
   struct st30p_rx_ops* ops_rx = &src->ops_rx;
-  gint ret;
+  GstAudioInfo* info;
+  gint sampling;
   GstCaps* caps;
+  gint ret;
 
   info = gst_audio_info_new();
 
+  if (!gst_mtl_common_st_to_gst_sampling(ops_rx->sampling, &sampling)) {
+    GST_ERROR("Failed to convert sampling rate");
+    gst_audio_info_free(info);
+    return FALSE;
+  }
+
   switch (ops_rx->fmt) {
     case ST30_FMT_PCM24:
-      gst_audio_info_set_format(info, GST_AUDIO_FORMAT_S24LE, info->rate, info->channels,
+      gst_audio_info_set_format(info, GST_AUDIO_FORMAT_S24LE, sampling, ops_rx->channel,
                                 NULL);
       break;
     case ST30_FMT_PCM16:
-      gst_audio_info_set_format(info, GST_AUDIO_FORMAT_S16LE, info->rate, info->channels,
+      gst_audio_info_set_format(info, GST_AUDIO_FORMAT_S16LE, sampling, ops_rx->channel,
                                 NULL);
       break;
     case ST30_FMT_PCM8:
-      gst_audio_info_set_format(info, GST_AUDIO_FORMAT_S8, info->rate, info->channels,
+      gst_audio_info_set_format(info, GST_AUDIO_FORMAT_S8, sampling, ops_rx->channel,
                                 NULL);
       break;
     default:
@@ -380,9 +387,6 @@ static gboolean gst_mtl_st30p_rx_negotiate(GstBaseSrc* basesrc) {
       return FALSE;
   }
 
-  info->rate = ops_rx->sampling;
-  info->channels = ops_rx->channel;
-
   caps = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING,
                              gst_audio_format_to_string(info->finfo->format), "channels",
                              G_TYPE_INT, info->channels, "rate", G_TYPE_INT, info->rate,

diff --git a/ecosystem/gstreamer_plugin/gst_mtl_st30p_tx.c b/ecosystem/gstreamer_plugin/gst_mtl_st30p_tx.c
@@ -128,27 +128,6 @@ static GstFlowReturn gst_mtl_st30p_tx_chain(GstPad* pad, GstObject* parent,
 static gboolean gst_mtl_st30p_tx_start(GstBaseSink* bsink);
 static gboolean gst_mtl_st30p_tx_cur_frame_flush(Gst_Mtl_St30p_Tx* sink);
 
-static gboolean gst_mtl_st30p_tx_parse_sampling(gint sampling,
-                                                enum st30_sampling* st_sampling) {
-  if (!st_sampling) {
-    GST_ERROR("Invalid st_sampling pointer");
-    return FALSE;
-  }
-  switch (sampling) {
-    case GST_MTL_SUPPORTED_AUDIO_SAMPLING_44_1K:
-      *st_sampling = ST31_SAMPLING_44K;
-      return TRUE;
-    case GST_MTL_SUPPORTED_AUDIO_SAMPLING_48K:
-      *st_sampling = ST30_SAMPLING_48K;
-      return TRUE;
-    case GST_MTL_SUPPORTED_AUDIO_SAMPLING_96K:
-      *st_sampling = ST30_SAMPLING_96K;
-      return TRUE;
-    default:
-      return FALSE;
-  }
-}
-
 static void gst_mtl_st30p_tx_class_init(Gst_Mtl_St30p_TxClass* klass) {
   GObjectClass* gobject_class;
   GstElementClass* gstelement_class;
@@ -311,7 +290,7 @@ static gboolean gst_mtl_st30p_tx_session_create(Gst_Mtl_St30p_Tx* sink, GstCaps*
   }
   ops_tx.channel = info->channels;
 
-  if (!gst_mtl_st30p_tx_parse_sampling(info->rate, &ops_tx.sampling)) {
+  if (!gst_mtl_common_gst_to_st_sampling(info->rate, &ops_tx.sampling)) {
     GST_ERROR("Failed to parse sampling rate");
     gst_audio_info_free(info);
     return FALSE;

diff --git a/lib/src/st2110/pipeline/st30_pipeline_tx.c b/lib/src/st2110/pipeline/st30_pipeline_tx.c
@@ -350,6 +350,13 @@ static void tx_st30p_framebuffs_flush(struct st30p_tx_ctx* ctx) {
       mt_sleep_ms(10);
     }
   }
+  /* Workaround: When tx_st30p_frame_done is called and subsequently framebuff->stat is
+   * set to ST30P_TX_FRAME_FREE, data from the framebuffer can still be in transport,
+   * already packetized and copied into rte_mbuf, waiting to be sent.
+   * TODO: add synchronization mechanism to ensure all data is sent before freeing the
+   * session.
+   */
+  mt_sleep_ms(50);
 }
 
 struct st30_frame* st30p_tx_get_frame(st30p_tx_handle handle) {