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decklinkaudiosrc: Implement timestamp jitter and discont handling

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This commit is contained in:
Sebastian Dröge 2014-12-19 13:24:10 +01:00
parent 7c0fb92f1b
commit 112fdd266a
2 changed files with 107 additions and 17 deletions
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114
sys/decklink/gstdecklinkaudiosrc.cpp
View file

@ -30,11 +30,15 @@ GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_src_debug);
#define MAX_QUEUE_LENGTH 5
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
enum
{
PROP_0,
PROP_MODE,
PROP_DEVICE_NUMBER
PROP_DEVICE_NUMBER,
PROP_ALIGNMENT_THRESHOLD,
PROP_DISCONT_WAIT
};
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("src",
@ -131,19 +135,25 @@ gst_decklink_audio_src_class_init (GstDecklinkAudioSrcClass * klass)
pushsrc_class->create = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_create);
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Playback Mode",
"Audio Mode to use for playback",
GST_TYPE_DECKLINK_MODE, GST_DECKLINK_MODE_NTSC,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class, PROP_DEVICE_NUMBER,
g_param_spec_int ("device-number", "Device number",
"Output device instance to use", 0, G_MAXINT, 0,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
G_PARAM_CONSTRUCT)));
g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
"Timestamp alignment threshold in nanoseconds", 0,
G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
g_param_spec_uint64 ("discont-wait", "Discont Wait",
"Window of time in nanoseconds to wait before "
"creating a discontinuity", 0,
G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
@ -158,8 +168,9 @@ gst_decklink_audio_src_class_init (GstDecklinkAudioSrcClass * klass)
static void
gst_decklink_audio_src_init (GstDecklinkAudioSrc * self)
{
self->mode = GST_DECKLINK_MODE_NTSC;
self->device_number = 0;
self->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
self->discont_wait = DEFAULT_DISCONT_WAIT;
gst_base_src_set_live (GST_BASE_SRC (self), TRUE);
gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME);
@ -177,12 +188,15 @@ gst_decklink_audio_src_set_property (GObject * object, guint property_id,
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
switch (property_id) {
case PROP_MODE:
self->mode = (GstDecklinkModeEnum) g_value_get_enum (value);
break;
case PROP_DEVICE_NUMBER:
self->device_number = g_value_get_int (value);
break;
case PROP_ALIGNMENT_THRESHOLD:
self->alignment_threshold = g_value_get_uint64 (value);
break;
case PROP_DISCONT_WAIT:
self->discont_wait = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
@ -196,12 +210,15 @@ gst_decklink_audio_src_get_property (GObject * object, guint property_id,
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
switch (property_id) {
case PROP_MODE:
g_value_set_enum (value, self->mode);
break;
case PROP_DEVICE_NUMBER:
g_value_set_int (value, self->device_number);
break;
case PROP_ALIGNMENT_THRESHOLD:
g_value_set_uint64 (value, self->alignment_threshold);
break;
case PROP_DISCONT_WAIT:
g_value_set_uint64 (value, self->discont_wait);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
@ -325,6 +342,9 @@ gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
CapturePacket *p;
AudioPacket *ap;
GstClockTime timestamp, duration;
GstClockTime start_time, end_time;
guint64 start_offset, end_offset;
gboolean discont = FALSE;
g_mutex_lock (&self->lock);
while (g_queue_is_empty (&self->current_packets) && !self->flushing) {
@ -356,7 +376,6 @@ gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
ap->input = self->input->input;
ap->input->AddRef ();
// TODO: Jitter/discont handling
duration =
gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate);
// Our capture time is the end timestamp, subtract the
@ -366,6 +385,66 @@ gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
else
timestamp = 0;
// Jitter and discontinuity handling, based on audiobasesrc
start_time = timestamp;
end_time = p->capture_time;
// Convert to the sample numbers
end_offset = gst_util_uint64_scale (end_time, self->info.rate, GST_SECOND);
if (end_offset >= sample_count)
start_offset = end_offset - sample_count;
else
start_offset = 0;
if (self->next_offset == -1) {
discont = TRUE;
} else {
guint64 diff, max_sample_diff;
// Check discont
if (start_offset <= self->next_offset)
diff = self->next_offset - start_offset;
else
diff = start_offset - self->next_offset;
max_sample_diff =
gst_util_uint64_scale_int (self->alignment_threshold, self->info.rate,
GST_SECOND);
// Discont!
if (G_UNLIKELY (diff >= max_sample_diff)) {
if (self->discont_wait > 0) {
if (self->discont_time == GST_CLOCK_TIME_NONE) {
self->discont_time = start_time;
} else if (start_time - self->discont_time >= self->discont_wait) {
discont = TRUE;
self->discont_time = GST_CLOCK_TIME_NONE;
}
} else {
discont = TRUE;
}
} else if (G_UNLIKELY (self->discont_time != GST_CLOCK_TIME_NONE)) {
// we have had a discont, but are now back on track!
self->discont_time = GST_CLOCK_TIME_NONE;
}
}
if (discont) {
// Have discont, need resync and use the capture timestamps
if (self->next_offset != -1)
GST_INFO_OBJECT (self, "Have discont. Expected %"
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
self->next_offset, start_offset);
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
self->next_offset = end_offset;
} else {
// No discont, just keep counting
self->discont_time = GST_CLOCK_TIME_NONE;
timestamp = gst_util_uint64_scale (self->next_offset, GST_SECOND, self->info.rate);
self->next_offset += sample_count;
duration = gst_util_uint64_scale (self->next_offset, GST_SECOND, self->info.rate) - timestamp;
}
GST_BUFFER_TIMESTAMP (*buffer) = timestamp;
GST_BUFFER_DURATION (*buffer) = duration;
@ -546,6 +625,7 @@ gst_decklink_audio_src_change_state (GstElement * element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
self->input->clock, TRUE));
self->flushing = FALSE;
self->next_offset = -1;
break;
}
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:{

10
sys/decklink/gstdecklinkaudiosrc.h
View file

@ -60,6 +60,16 @@ struct _GstDecklinkAudioSrc
GMutex lock;
gboolean flushing;
GQueue current_packets;
/* properties for handling jittery timestamps */
GstClockTime alignment_threshold;
GstClockTime discont_wait;
/* counter to keep track of timestamps */
gint64 next_offset;
/* Last time we noticed a discont */
GstClockTime discont_time;
};
struct _GstDecklinkAudioSrcClass