FaSTTUBe/GST-Tensordecoder-ov_ep
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avwait: Send dropping=true message after all streams stopped

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Previously it was dispatched before the last video buffer, and audio
buffers would follow afterwards. It's misleading to send the
dropping=true message before both streams have really stopped, it can
lead to races when someone is e.g. waiting for that message to send EOS.

Also added some debug output.

https://bugzilla.gnome.org/show_bug.cgi?id=797145
This commit is contained in:
Vivia Nikolaidou 2018-09-14 16:05:20 +03:00
parent 1d6814f54b
commit b1b4a04338
2 changed files with 77 additions and 11 deletions
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87
gst/timecode/gstavwait.c
View File

@ -100,6 +100,15 @@ enum
#define DEFAULT_TARGET_RUNNING_TIME GST_CLOCK_TIME_NONE #define DEFAULT_TARGET_RUNNING_TIME GST_CLOCK_TIME_NONE
#define DEFAULT_MODE MODE_TIMECODE #define DEFAULT_MODE MODE_TIMECODE
/* flags for self->must_send_end_message */
enum
{
END_MESSAGE_NORMAL = 0,
END_MESSAGE_STREAM_ENDED = 1,
END_MESSAGE_VIDEO_PUSHED = 2,
END_MESSAGE_AUDIO_PUSHED = 4
};
static void gst_avwait_set_property (GObject * object, static void gst_avwait_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec); guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_avwait_get_property (GObject * object, static void gst_avwait_get_property (GObject * object,
@ -322,6 +331,7 @@ gst_avwait_change_state (GstElement * element, GstStateChange transition)
self->audio_eos_flag = FALSE; self->audio_eos_flag = FALSE;
self->video_flush_flag = FALSE; self->video_flush_flag = FALSE;
self->audio_flush_flag = FALSE; self->audio_flush_flag = FALSE;
self->must_send_end_message = END_MESSAGE_NORMAL;
g_mutex_unlock (&self->mutex); g_mutex_unlock (&self->mutex);
default: default:
break; break;
@ -700,6 +710,7 @@ gst_avwait_asink_event (GstPad * pad, GstObject * parent, GstEvent * event)
case GST_EVENT_EOS: case GST_EVENT_EOS:
g_mutex_lock (&self->mutex); g_mutex_lock (&self->mutex);
self->audio_eos_flag = TRUE; self->audio_eos_flag = TRUE;
self->must_send_end_message = END_MESSAGE_NORMAL;
g_cond_signal (&self->audio_cond); g_cond_signal (&self->audio_cond);
g_mutex_unlock (&self->mutex); g_mutex_unlock (&self->mutex);
break; break;
@ -738,6 +749,7 @@ gst_avwait_vsink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
GstVideoTimeCode *tc = NULL; GstVideoTimeCode *tc = NULL;
GstVideoTimeCodeMeta *tc_meta; GstVideoTimeCodeMeta *tc_meta;
gboolean retry = FALSE; gboolean retry = FALSE;
gboolean ret = GST_FLOW_OK;
timestamp = GST_BUFFER_TIMESTAMP (inbuf); timestamp = GST_BUFFER_TIMESTAMP (inbuf);
if (timestamp == GST_CLOCK_TIME_NONE) { if (timestamp == GST_CLOCK_TIME_NONE) {
@ -805,8 +817,7 @@ gst_avwait_vsink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
self->vsegment.position); self->vsegment.position);
if (self->recording) { if (self->recording) {
self->audio_running_time_to_end_at = self->running_time_to_end_at; self->audio_running_time_to_end_at = self->running_time_to_end_at;
gst_avwait_send_element_message (self, TRUE, self->must_send_end_message |= END_MESSAGE_STREAM_ENDED;
self->running_time_to_end_at);
} }
} }
gst_buffer_unref (inbuf); gst_buffer_unref (inbuf);
@ -868,7 +879,7 @@ gst_avwait_vsink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
&& running_time <= self->running_time_to_end_at) { && running_time <= self->running_time_to_end_at) {
/* We just stopped recording: synchronise the audio */ /* We just stopped recording: synchronise the audio */
self->audio_running_time_to_end_at = running_time; self->audio_running_time_to_end_at = running_time;
gst_avwait_send_element_message (self, TRUE, running_time); self->must_send_end_message |= END_MESSAGE_STREAM_ENDED;
} else if (running_time < self->running_time_to_wait_for } else if (running_time < self->running_time_to_wait_for
&& self->running_time_to_wait_for != GST_CLOCK_TIME_NONE) { && self->running_time_to_wait_for != GST_CLOCK_TIME_NONE) {
self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE; self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE;
@ -921,10 +932,33 @@ gst_avwait_vsink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
self->was_recording = self->recording; self->was_recording = self->recording;
g_cond_signal (&self->cond); g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex); g_mutex_unlock (&self->mutex);
if (inbuf) if (inbuf) {
return gst_pad_push (self->vsrcpad, inbuf); GST_WARNING_OBJECT (self, "Pass video buffer ending at %" GST_TIME_FORMAT,
else GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf) +
return GST_FLOW_OK; GST_BUFFER_DURATION (inbuf)));
ret = gst_pad_push (self->vsrcpad, inbuf);
}
g_mutex_lock (&self->mutex);
if (self->must_send_end_message & END_MESSAGE_AUDIO_PUSHED) {
self->must_send_end_message = END_MESSAGE_NORMAL;
g_mutex_unlock (&self->mutex);
gst_avwait_send_element_message (self, TRUE,
self->audio_running_time_to_end_at);
} else if (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) {
if (self->audio_eos_flag) {
self->must_send_end_message = END_MESSAGE_NORMAL;
g_mutex_unlock (&self->mutex);
gst_avwait_send_element_message (self, TRUE,
self->audio_running_time_to_end_at);
} else {
self->must_send_end_message |= END_MESSAGE_VIDEO_PUSHED;
g_mutex_unlock (&self->mutex);
}
} else {
g_mutex_unlock (&self->mutex);
}
return ret;
} }
/* /*
@ -955,6 +989,10 @@ gst_avwait_asink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
GstClockTime duration; GstClockTime duration;
GstClockTime running_time_at_end = GST_CLOCK_TIME_NONE; GstClockTime running_time_at_end = GST_CLOCK_TIME_NONE;
gint asign, vsign = 1, esign = 1; gint asign, vsign = 1, esign = 1;
GstFlowReturn ret = GST_FLOW_OK;
/* Make sure the video thread doesn't send the element message before we
* actually call gst_pad_push */
gboolean send_element_message = FALSE;
timestamp = GST_BUFFER_TIMESTAMP (inbuf); timestamp = GST_BUFFER_TIMESTAMP (inbuf);
if (timestamp == GST_CLOCK_TIME_NONE) { if (timestamp == GST_CLOCK_TIME_NONE) {
@ -1072,15 +1110,42 @@ gst_avwait_asink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf)
inbuf = inbuf =
gst_audio_buffer_clip (inbuf, &asegment2, self->ainfo.rate, gst_audio_buffer_clip (inbuf, &asegment2, self->ainfo.rate,
self->ainfo.bpf); self->ainfo.bpf);
if (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) {
send_element_message = TRUE;
}
} else { } else {
/* Programming error? Shouldn't happen */ /* Programming error? Shouldn't happen */
g_assert_not_reached (); g_assert_not_reached ();
} }
g_mutex_unlock (&self->mutex); g_mutex_unlock (&self->mutex);
if (inbuf) if (inbuf) {
return gst_pad_push (self->asrcpad, inbuf); GstClockTime new_duration =
else gst_util_uint64_scale (gst_buffer_get_size (inbuf) / self->ainfo.bpf,
return GST_FLOW_OK; GST_SECOND, self->ainfo.rate);
GstClockTime new_running_time_at_end =
gst_segment_to_running_time (&self->asegment, GST_FORMAT_TIME,
self->asegment.position + new_duration);
GST_WARNING_OBJECT (self, "Pass audio buffer ending at %" GST_TIME_FORMAT,
GST_TIME_ARGS (new_running_time_at_end));
ret = gst_pad_push (self->asrcpad, inbuf);
}
if (send_element_message) {
g_mutex_lock (&self->mutex);
if ((self->must_send_end_message & END_MESSAGE_VIDEO_PUSHED) ||
self->video_eos_flag) {
self->must_send_end_message = END_MESSAGE_NORMAL;
g_mutex_unlock (&self->mutex);
gst_avwait_send_element_message (self, TRUE,
self->audio_running_time_to_end_at);
} else if (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) {
self->must_send_end_message |= END_MESSAGE_AUDIO_PUSHED;
g_mutex_unlock (&self->mutex);
} else {
g_assert_not_reached ();
}
}
send_element_message = FALSE;
return ret;
} }
static GstIterator * static GstIterator *

1
gst/timecode/gstavwait.h
View File

@ -84,6 +84,7 @@ struct _GstAvWait
gboolean dropping; gboolean dropping;
gboolean recording; gboolean recording;
gboolean was_recording; gboolean was_recording;
gint must_send_end_message;
GCond cond; GCond cond;
GMutex mutex; GMutex mutex;