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decklinksrc: Sample the pipeline clock for the timestamps instead of coming up with our own
If we just count the frames and calculate timestamps from that, all frames will arrive late in the sink as we have a live source here. Instead take the pipeline clock at capture time as reference.
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parent
ee8766b342
commit
50c4ea54f6
3 changed files with 35 additions and 11 deletions
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@ -78,6 +78,8 @@ HRESULT
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videoFrame, IDeckLinkAudioInputPacket * audioFrame)
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{
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GstDecklinkSrc *decklinksrc;
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GstClock *clock;
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GstClockTime base_time, clock_time, capture_time;
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const char *timecodeString = NULL;
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g_return_val_if_fail (priv != NULL, S_OK);
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@ -109,9 +111,32 @@ HRESULT
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}
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}
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GST_DEBUG_OBJECT (decklinksrc, "Frame received [%s] - %s - Size: %li bytes",
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timecodeString != NULL ? timecodeString : "No timecode",
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"Valid Frame", videoFrame->GetRowBytes () * videoFrame->GetHeight ());
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GST_OBJECT_LOCK (decklinksrc);
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if ((clock = GST_ELEMENT_CLOCK (decklinksrc))) {
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base_time = GST_ELEMENT (decklinksrc)->base_time;
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gst_object_ref (clock);
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} else {
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base_time = GST_CLOCK_TIME_NONE;
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}
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GST_OBJECT_UNLOCK (decklinksrc);
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if (clock) {
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clock_time = gst_clock_get_time (clock);
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gst_object_unref (clock);
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} else {
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clock_time = GST_CLOCK_TIME_NONE;
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}
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if (base_time != GST_CLOCK_TIME_NONE) {
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capture_time = clock_time - base_time;
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} else {
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capture_time = GST_CLOCK_TIME_NONE;
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}
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GST_DEBUG_OBJECT (decklinksrc,
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"Frame received [%s] - %s - %" GST_TIME_FORMAT "Size: %li bytes",
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timecodeString != NULL ? timecodeString : "No timecode", "Valid Frame",
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GST_TIME_ARGS (capture_time),
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videoFrame->GetRowBytes () * videoFrame->GetHeight ());
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if (timecodeString)
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FREE_COM_STRING (timecodeString);
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@ -130,6 +155,7 @@ HRESULT
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audioFrame->AddRef ();
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decklinksrc->audio_frame = audioFrame;
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}
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decklinksrc->capture_time = capture_time;
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/* increment regardless whether frame was dropped or not */
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decklinksrc->frame_num++;
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@ -756,6 +756,7 @@ gst_decklink_src_task (void *priv)
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GstFlowReturn video_flow, audio_flow, flow;
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const GstDecklinkMode *mode;
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gboolean discont = FALSE;
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GstClockTime capture_time;
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GST_DEBUG_OBJECT (decklinksrc, "task");
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@ -766,6 +767,7 @@ gst_decklink_src_task (void *priv)
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}
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video_frame = decklinksrc->video_frame;
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audio_frame = decklinksrc->audio_frame;
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capture_time = decklinksrc->capture_time;
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decklinksrc->video_frame = NULL;
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decklinksrc->audio_frame = NULL;
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g_mutex_unlock (&decklinksrc->mutex);
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@ -846,12 +848,9 @@ gst_decklink_src_task (void *priv)
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vf->input->AddRef ();
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}
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GST_BUFFER_TIMESTAMP (buffer) =
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gst_util_uint64_scale_int (decklinksrc->frame_num * GST_SECOND,
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GST_BUFFER_TIMESTAMP (buffer) = capture_time;
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GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (GST_SECOND,
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mode->fps_d, mode->fps_n);
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GST_BUFFER_DURATION (buffer) =
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gst_util_uint64_scale_int ((decklinksrc->frame_num + 1) * GST_SECOND,
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mode->fps_d, mode->fps_n) - GST_BUFFER_TIMESTAMP (buffer);
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GST_BUFFER_OFFSET (buffer) = decklinksrc->frame_num;
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GST_BUFFER_OFFSET_END (buffer) = decklinksrc->frame_num; /* FIXME: +1? */
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@ -873,9 +872,7 @@ gst_decklink_src_task (void *priv)
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audio_buffer = gst_buffer_new_and_alloc (n_samples * 2 * 2);
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gst_buffer_fill (audio_buffer, 0, data, n_samples * 2 * 2);
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GST_BUFFER_TIMESTAMP (audio_buffer) =
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gst_util_uint64_scale_int (decklinksrc->num_audio_samples * GST_SECOND,
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1, 48000);
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GST_BUFFER_TIMESTAMP (audio_buffer) = capture_time;
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/* FIXME: should be next_timestamp - timestamp for perfect stream */
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GST_BUFFER_DURATION (audio_buffer) =
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gst_util_uint64_scale_int (n_samples * GST_SECOND, 1, 48000);
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@ -60,6 +60,7 @@ struct _GstDecklinkSrc
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int dropped_frames;
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int dropped_frames_old;
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gboolean stop;
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GstClockTime capture_time;
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IDeckLinkVideoInputFrame *video_frame;
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IDeckLinkAudioInputPacket * audio_frame;
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