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Fix: st30_pipeline don't close until mbuf is free #1049

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merged 1 commit into from
Jan 30, 2025
Merged

Fix: st30_pipeline don't close until mbuf is free #1049

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27 changes: 26 additions & 1 deletion ecosystem/gstreamer_plugin/gst_mtl_common.c
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Original file line number Diff line number Diff line change
Expand Up @@ -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;
Expand All @@ -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;
}
}
Expand Down
5 changes: 4 additions & 1 deletion ecosystem/gstreamer_plugin/gst_mtl_common.h
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Original file line number Diff line number Diff line change
Expand Up @@ -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);
Expand Down
22 changes: 13 additions & 9 deletions ecosystem/gstreamer_plugin/gst_mtl_st30p_rx.c
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Original file line number Diff line number Diff line change
Expand Up @@ -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;
}
Expand Down Expand Up @@ -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:
Expand All @@ -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,
Expand Down
23 changes: 1 addition & 22 deletions ecosystem/gstreamer_plugin/gst_mtl_st30p_tx.c
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Original file line number Diff line number Diff line change
Expand Up @@ -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;
Expand Down Expand Up @@ -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;
Expand Down
7 changes: 7 additions & 0 deletions lib/src/st2110/pipeline/st30_pipeline_tx.c
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Original file line number Diff line number Diff line change
Expand Up @@ -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) {
Expand Down
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