mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-13 04:41:16 +00:00
ajasrc: Fix handling of timestamps and don't rely on driver frame counters
The driver frame counters (processed, dropped, buffer level) are not always correct apparently, and don't allow reliably assigning a frame number to captured frames. Instead of relying on them, count the number of frames directly here and detect dropped frames based on the capture times of the frames: if more than 1.75 frame durations are between two frames, then there must've been a dropped frame. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7163>
This commit is contained in:
parent
89096dde38
commit
3eec4703c6
1 changed files with 114 additions and 93 deletions
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@ -28,17 +28,22 @@
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*
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* ```sh
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* gst-launch-1.0 ajasrc video-format=1080p-3000 ! ajasrcdemux name=d \
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* d.video ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! videoconvert ! autovideosink \
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* d.audio ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! audioconvert ! audioresample ! autoaudiosink
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* d.video ! queue max-size-bytes=0 max-size-buffers=0
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* max-size-time=1000000000 ! videoconvert ! autovideosink \ d.audio ! queue
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* max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! audioconvert !
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* audioresample ! autoaudiosink
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* ```
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*
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* Capture 1080p30 audio/video and directly output it again on the same card
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*
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* ```sh
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* gst-launch-1.0 ajasrc video-format=1080p-3000 channel=1 input-source=sdi-1 audio-system=2 ! ajasrcdemux name=d \
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* d.video ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! c.video \
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* d.audio ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! c.audio \
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* ajasinkcombiner name=c ! ajasink channel=0 reference-source=input-1
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* gst-launch-1.0 ajasrc video-format=1080p-3000 channel=1 input-source=sdi-1
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* audio-system=2 ! ajasrcdemux name=d \
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* d.video ! queue max-size-bytes=0 max-size-buffers=0
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* max-size-time=1000000000 ! c.video \
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* d.audio ! queue max-size-bytes=0 max-size-buffers=0
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* max-size-time=1000000000 ! c.audio \ ajasinkcombiner name=c ! ajasink
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* channel=0 reference-source=input-1
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* ```
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*
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* Since: 1.24
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@ -2283,7 +2288,7 @@ next_item:
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g_free(__dbg); \
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__msg = gst_message_new_error(GST_OBJECT(el), __err, __fmt_dbg); \
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QueueItem item = {.type = QUEUE_ITEM_TYPE_ERROR, .error{.msg = __msg}}; \
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gst_vec_deque_push_tail_struct(el->queue, &item); \
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gst_vec_deque_push_tail_struct(el->queue, &item); \
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g_cond_signal(&el->queue_cond); \
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} \
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G_STMT_END;
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@ -2293,9 +2298,6 @@ static void capture_thread_func(AJAThread *thread, void *data) {
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GstClock *clock = NULL;
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GstClock *real_time_clock;
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AUTOCIRCULATE_TRANSFER transfer;
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guint64 frames_dropped_last = G_MAXUINT64;
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gboolean have_signal = TRUE, discont = TRUE;
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guint iterations_without_frame = 0;
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NTV2VideoFormat last_detected_video_format = ::NTV2_FORMAT_UNKNOWN;
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if (self->capture_cpu_core != G_MAXUINT) {
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@ -2317,12 +2319,29 @@ static void capture_thread_func(AJAThread *thread, void *data) {
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real_time_clock = GST_CLOCK(g_object_new(GST_TYPE_SYSTEM_CLOCK, "clock-type",
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GST_CLOCK_TYPE_REALTIME, NULL));
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// if we currently have signal
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bool have_signal = true;
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// if the *next* output frame should have the discont flag set
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bool discont = true;
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// if the pipeline clock is the monotonic system clock
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bool clock_is_monotonic_system_clock = false;
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// if the next frame is the first one after autocirculate was started
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bool first_frame_after_start = true;
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bool reset_clock = false;
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GstClockTime first_frame_time = 0;
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guint64 first_frame_processed_plus_dropped_minus_buffered = 0;
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// acFrameTime of the last captured frame. Used to detect
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// if a frame is actually missing or the signal changed
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GstClockTime last_frame_time = GST_CLOCK_TIME_NONE;
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// Base time of the first frame after a clock observation reset
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// Internal frame time is calculated from this and the number of frames
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// captured so far.
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GstClockTime first_frame_time = GST_CLOCK_TIME_NONE;
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guint64 num_frames = 0;
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// Number of frames to drop. After signal loss the first few frames
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// often have invalid acFrameTime so we drop them to calculate valid
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// timestamps afterwards
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guint frames_to_drop = 0;
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// Number of vsync iterations without a captured frame. If this becomes too
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// high we assume signal loss.
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guint iterations_without_frame = 0;
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g_mutex_lock(&self->queue_lock);
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restart:
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@ -2352,12 +2371,19 @@ restart:
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gst_clear_object(&system_clock);
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}
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// Reset all local state after restart
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have_signal = true;
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discont = true;
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first_frame_after_start = true;
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first_frame_time = 0;
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frames_dropped_last = G_MAXUINT64;
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have_signal = TRUE;
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last_frame_time = GST_CLOCK_TIME_NONE;
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first_frame_time = GST_CLOCK_TIME_NONE;
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num_frames = 0;
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frames_to_drop = 0;
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iterations_without_frame = 0;
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g_mutex_lock(&self->queue_lock);
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self->video_format = ::NTV2_FORMAT_UNKNOWN;
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while (self->playing && !self->shutdown) {
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// If we don't have a video format configured, configure the device now
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// and potentially auto-detect the video format
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@ -2381,7 +2407,6 @@ restart:
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if (self->video_format == ::NTV2_FORMAT_UNKNOWN) {
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GST_DEBUG_OBJECT(self, "No signal, waiting");
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frames_dropped_last = G_MAXUINT64;
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if (have_signal) {
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QueueItem item = {
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.type = QUEUE_ITEM_TYPE_SIGNAL_CHANGE,
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@ -2390,8 +2415,7 @@ restart:
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.vpid = 0}};
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gst_vec_deque_push_tail_struct(self->queue, &item);
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g_cond_signal(&self->queue_cond);
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have_signal = FALSE;
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discont = TRUE;
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have_signal = false;
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}
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self->device->device->WaitForInputVerticalInterrupt(self->channel);
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continue;
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@ -2505,7 +2529,6 @@ restart:
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GST_DEBUG_OBJECT(self, "No signal, waiting");
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g_mutex_unlock(&self->queue_lock);
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frames_dropped_last = G_MAXUINT64;
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if (have_signal || current_video_format != last_detected_video_format) {
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QueueItem item = {
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.type = QUEUE_ITEM_TYPE_SIGNAL_CHANGE,
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last_detected_video_format = ::NTV2_FORMAT_UNKNOWN;
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gst_vec_deque_push_tail_struct(self->queue, &item);
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g_cond_signal(&self->queue_cond);
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have_signal = FALSE;
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discont = TRUE;
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have_signal = false;
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}
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self->device->device->WaitForInputVerticalInterrupt(self->channel);
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g_mutex_lock(&self->queue_lock);
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@ -2542,7 +2564,6 @@ restart:
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current_string.c_str(), configured_string.c_str(),
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effective_string.c_str());
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g_mutex_unlock(&self->queue_lock);
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frames_dropped_last = G_MAXUINT64;
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if (have_signal || current_video_format != last_detected_video_format) {
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QueueItem item = {
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.type = QUEUE_ITEM_TYPE_SIGNAL_CHANGE,
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last_detected_video_format = current_video_format;
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gst_vec_deque_push_tail_struct(self->queue, &item);
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g_cond_signal(&self->queue_cond);
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have_signal = FALSE;
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discont = TRUE;
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have_signal = false;
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}
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self->device->device->WaitForInputVerticalInterrupt(self->channel);
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g_mutex_lock(&self->queue_lock);
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@ -2592,7 +2612,7 @@ restart:
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GST_TIME_ARGS(status.acRDTSCCurrentTime * 100),
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status.acFramesProcessed, status.acFramesDropped, status.acBufferLevel);
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if (status.IsRunning() && status.acBufferLevel > 1) {
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if (status.IsRunning() && status.HasAvailableInputFrame()) {
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GstBuffer *video_buffer = NULL;
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GstBuffer *audio_buffer = NULL;
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GstBuffer *anc_buffer = NULL, *anc_buffer2 = NULL;
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.vpid = vpid_a}};
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gst_vec_deque_push_tail_struct(self->queue, &item);
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g_cond_signal(&self->queue_cond);
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have_signal = TRUE;
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reset_clock = true;
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have_signal = true;
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last_frame_time = GST_CLOCK_TIME_NONE;
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// Drop the next frames after signal recovery as the capture times
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// are generally just wrong.
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@ -2699,16 +2719,6 @@ restart:
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continue;
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}
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if (frames_to_drop > 0) {
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GST_TRACE_OBJECT(self, "Dropping frame");
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frames_to_drop -= 1;
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gst_clear_buffer(&anc_buffer2);
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gst_clear_buffer(&anc_buffer);
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gst_clear_buffer(&audio_buffer);
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gst_clear_buffer(&video_buffer);
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continue;
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}
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const AUTOCIRCULATE_TRANSFER_STATUS &transfer_status =
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transfer.GetTransferStatus();
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const FRAME_STAMP &frame_stamp = transfer_status.GetFrameStamp();
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transfer_status.acFramesDropped,
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transfer_status.acBufferLevel);
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if (frames_to_drop > 0) {
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GST_TRACE_OBJECT(self, "Dropping frame");
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frames_to_drop -= 1;
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gst_clear_buffer(&anc_buffer2);
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gst_clear_buffer(&anc_buffer);
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gst_clear_buffer(&audio_buffer);
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gst_clear_buffer(&video_buffer);
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continue;
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}
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GstClockTime frame_time_real = frame_stamp.acFrameTime * 100;
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// Convert capture time from real-time clock to monotonic clock by
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frame_time_monotonic = frame_time_real + diff;
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}
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// Detect frame drop: backwards capture time is clearly a problem,
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// otherwise consider a frame being dropped if more than 1.75 frame
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// durations are between two frames
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if (last_frame_time != GST_CLOCK_TIME_NONE) {
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GstClockTime frame_drop_threshold =
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(7 * self->configured_info.fps_d * GST_SECOND) /
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(4 * self->configured_info.fps_n);
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if (last_frame_time >= frame_time_monotonic) {
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GST_ERROR_OBJECT(self, "Frame capture time went backwards");
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last_frame_time = GST_CLOCK_TIME_NONE;
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} else if (frame_time_monotonic - last_frame_time >
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frame_drop_threshold) {
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GstClockTime timestamp =
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last_frame_time + self->configured_info.fps_d * GST_SECOND /
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self->configured_info.fps_n;
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GstClockTime timestamp_end = frame_time_monotonic;
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GST_WARNING_OBJECT(self,
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"Frame drop of %" GST_TIME_FORMAT " detected",
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GST_TIME_ARGS(timestamp_end - timestamp));
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QueueItem item = {.type = QUEUE_ITEM_TYPE_FRAMES_DROPPED,
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.frames_dropped = {.driver_side = TRUE,
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.timestamp_start = timestamp,
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.timestamp_end = timestamp_end}};
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gst_vec_deque_push_tail_struct(self->queue, &item);
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g_cond_signal(&self->queue_cond);
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last_frame_time = GST_CLOCK_TIME_NONE;
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} else {
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GST_TRACE_OBJECT(
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self, "Time since last frame: %" GST_TIME_FORMAT,
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GST_TIME_ARGS(frame_time_monotonic - last_frame_time));
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}
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}
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GstClockTime frame_src_time;
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// Update clock mapping
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if (first_frame_after_start || reset_clock) {
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if (first_frame_after_start || last_frame_time == GST_CLOCK_TIME_NONE) {
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GstClockTime internal, external;
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guint64 num, denom;
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// Keep observations if there was only temporary signal loss or a
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// format change as the source is either using the same clock as
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// before, or it's different but then our previous configuration would
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// be as good/bad as the local monotonic system clock and over some
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// frames we would converge to the new clock.
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if (!first_frame_after_start) {
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// Keep observations if there was only temporary signal loss as the
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// source is either using the same clock as before, or it's different
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// but then our previous configuration would be as good/bad as the
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// local monotonic system clock and over some frames we would converge
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// to the new clock.
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if (first_frame_after_start) {
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// FIXME: Workaround to get rid of all previous observations
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g_object_set(self->clock, "window-size", 32, NULL);
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}
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&denom);
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first_frame_time = frame_src_time = gst_clock_adjust_with_calibration(
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NULL, frame_time_monotonic, internal, external, num, denom);
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first_frame_processed_plus_dropped_minus_buffered =
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transfer_status.acFramesProcessed +
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transfer_status.acFramesDropped - transfer_status.acBufferLevel;
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GST_TRACE_OBJECT(
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self,
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"Initializing clock with first frame time %" GST_TIME_FORMAT
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" and initial frame count %" G_GUINT64_FORMAT,
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GST_TIME_ARGS(first_frame_time),
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first_frame_processed_plus_dropped_minus_buffered);
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self, "Initializing clock with first frame time %" GST_TIME_FORMAT,
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GST_TIME_ARGS(first_frame_time));
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first_frame_after_start = false;
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reset_clock = false;
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discont = TRUE;
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num_frames = 0;
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} else {
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gdouble r_squared;
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frame_src_time =
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first_frame_time +
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gst_util_uint64_scale_ceil(
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transfer_status.acFramesProcessed +
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transfer_status.acFramesDropped -
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transfer_status.acBufferLevel -
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first_frame_processed_plus_dropped_minus_buffered,
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self->configured_info.fps_d * GST_SECOND,
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self->configured_info.fps_n);
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gst_util_uint64_scale_ceil(num_frames,
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self->configured_info.fps_d * GST_SECOND,
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self->configured_info.fps_n);
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gst_clock_add_observation(self->clock, frame_time_monotonic,
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frame_src_time, &r_squared);
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}
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last_frame_time = frame_time_monotonic;
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num_frames += 1;
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GstClockTime capture_time;
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if (self->clock == clock) {
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// If the pipeline is using our clock then we can directly use the
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GST_BUFFER_DURATION(audio_buffer) = gst_util_uint64_scale(
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GST_SECOND, self->configured_info.fps_d, self->configured_info.fps_n);
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if (frames_dropped_last == G_MAXUINT64) {
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frames_dropped_last = transfer_status.acFramesDropped;
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} else if (frames_dropped_last < transfer_status.acFramesDropped) {
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GST_WARNING_OBJECT(
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self, "Dropped %" G_GUINT64_FORMAT " frames",
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transfer_status.acFramesDropped - frames_dropped_last);
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GstClockTime timestamp = gst_util_uint64_scale(
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transfer_status.acFramesProcessed + frames_dropped_last,
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self->configured_info.fps_n,
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self->configured_info.fps_d * GST_SECOND);
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GstClockTime timestamp_end = gst_util_uint64_scale(
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transfer_status.acFramesProcessed + transfer_status.acFramesDropped,
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self->configured_info.fps_n,
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self->configured_info.fps_d * GST_SECOND);
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QueueItem item = {.type = QUEUE_ITEM_TYPE_FRAMES_DROPPED,
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.frames_dropped = {.driver_side = TRUE,
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.timestamp_start = timestamp,
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.timestamp_end = timestamp_end}};
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gst_vec_deque_push_tail_struct(self->queue, &item);
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g_cond_signal(&self->queue_cond);
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frames_dropped_last = transfer_status.acFramesDropped;
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discont = TRUE;
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}
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gst_buffer_set_size(audio_buffer, transfer.GetCapturedAudioByteCount());
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if (anc_buffer)
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gst_buffer_set_size(anc_buffer,
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@ -2964,7 +2987,7 @@ restart:
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gst_vec_deque_drop_struct(self->queue, i, NULL);
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gst_vec_deque_push_tail_struct(self->queue, &item);
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self->queue_num_frames -= 1;
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discont = TRUE;
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discont = true;
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g_cond_signal(&self->queue_cond);
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break;
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}
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@ -2974,7 +2997,7 @@ restart:
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if (discont) {
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GST_BUFFER_FLAG_SET(video_buffer, GST_BUFFER_FLAG_DISCONT);
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GST_BUFFER_FLAG_SET(audio_buffer, GST_BUFFER_FLAG_DISCONT);
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discont = FALSE;
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discont = false;
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}
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QueueItem item = {
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@ -3006,7 +3029,6 @@ restart:
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if (have_signal && iterations_without_frame < 32) {
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iterations_without_frame++;
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} else {
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frames_dropped_last = G_MAXUINT64;
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if (have_signal || last_detected_video_format != current_video_format) {
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QueueItem item = {
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.type = QUEUE_ITEM_TYPE_SIGNAL_CHANGE,
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@ -3016,8 +3038,7 @@ restart:
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last_detected_video_format = current_video_format;
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gst_vec_deque_push_tail_struct(self->queue, &item);
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g_cond_signal(&self->queue_cond);
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have_signal = FALSE;
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discont = TRUE;
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have_signal = false;
|
||||
}
|
||||
}
|
||||
|
||||
|
|
Loading…
Reference in a new issue