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:
Sebastian Dröge 2024-07-11 17:23:43 +03:00 committed by GStreamer Marge Bot
parent 89096dde38
commit 3eec4703c6

View file

@ -28,17 +28,22 @@
*
* ```sh
* gst-launch-1.0 ajasrc video-format=1080p-3000 ! ajasrcdemux name=d \
* d.video ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! videoconvert ! autovideosink \
* d.audio ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! audioconvert ! audioresample ! autoaudiosink
* d.video ! queue max-size-bytes=0 max-size-buffers=0
* max-size-time=1000000000 ! videoconvert ! autovideosink \ d.audio ! queue
* max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! audioconvert !
* audioresample ! autoaudiosink
* ```
*
* Capture 1080p30 audio/video and directly output it again on the same card
*
* ```sh
* gst-launch-1.0 ajasrc video-format=1080p-3000 channel=1 input-source=sdi-1 audio-system=2 ! ajasrcdemux name=d \
* d.video ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! c.video \
* d.audio ! queue max-size-bytes=0 max-size-buffers=0 max-size-time=1000000000 ! c.audio \
* ajasinkcombiner name=c ! ajasink channel=0 reference-source=input-1
* gst-launch-1.0 ajasrc video-format=1080p-3000 channel=1 input-source=sdi-1
* audio-system=2 ! ajasrcdemux name=d \
* d.video ! queue max-size-bytes=0 max-size-buffers=0
* max-size-time=1000000000 ! c.video \
* d.audio ! queue max-size-bytes=0 max-size-buffers=0
* max-size-time=1000000000 ! c.audio \ ajasinkcombiner name=c ! ajasink
* channel=0 reference-source=input-1
* ```
*
* Since: 1.24
@ -2283,7 +2288,7 @@ next_item:
g_free(__dbg); \
__msg = gst_message_new_error(GST_OBJECT(el), __err, __fmt_dbg); \
QueueItem item = {.type = QUEUE_ITEM_TYPE_ERROR, .error{.msg = __msg}}; \
gst_vec_deque_push_tail_struct(el->queue, &item); \
gst_vec_deque_push_tail_struct(el->queue, &item); \
g_cond_signal(&el->queue_cond); \
} \
G_STMT_END;
@ -2293,9 +2298,6 @@ static void capture_thread_func(AJAThread *thread, void *data) {
GstClock *clock = NULL;
GstClock *real_time_clock;
AUTOCIRCULATE_TRANSFER transfer;
guint64 frames_dropped_last = G_MAXUINT64;
gboolean have_signal = TRUE, discont = TRUE;
guint iterations_without_frame = 0;
NTV2VideoFormat last_detected_video_format = ::NTV2_FORMAT_UNKNOWN;
if (self->capture_cpu_core != G_MAXUINT) {
@ -2317,12 +2319,29 @@ static void capture_thread_func(AJAThread *thread, void *data) {
real_time_clock = GST_CLOCK(g_object_new(GST_TYPE_SYSTEM_CLOCK, "clock-type",
GST_CLOCK_TYPE_REALTIME, NULL));
// if we currently have signal
bool have_signal = true;
// if the *next* output frame should have the discont flag set
bool discont = true;
// if the pipeline clock is the monotonic system clock
bool clock_is_monotonic_system_clock = false;
// if the next frame is the first one after autocirculate was started
bool first_frame_after_start = true;
bool reset_clock = false;
GstClockTime first_frame_time = 0;
guint64 first_frame_processed_plus_dropped_minus_buffered = 0;
// acFrameTime of the last captured frame. Used to detect
// if a frame is actually missing or the signal changed
GstClockTime last_frame_time = GST_CLOCK_TIME_NONE;
// Base time of the first frame after a clock observation reset
// Internal frame time is calculated from this and the number of frames
// captured so far.
GstClockTime first_frame_time = GST_CLOCK_TIME_NONE;
guint64 num_frames = 0;
// Number of frames to drop. After signal loss the first few frames
// often have invalid acFrameTime so we drop them to calculate valid
// timestamps afterwards
guint frames_to_drop = 0;
// Number of vsync iterations without a captured frame. If this becomes too
// high we assume signal loss.
guint iterations_without_frame = 0;
g_mutex_lock(&self->queue_lock);
restart:
@ -2352,12 +2371,19 @@ restart:
gst_clear_object(&system_clock);
}
// Reset all local state after restart
have_signal = true;
discont = true;
first_frame_after_start = true;
first_frame_time = 0;
frames_dropped_last = G_MAXUINT64;
have_signal = TRUE;
last_frame_time = GST_CLOCK_TIME_NONE;
first_frame_time = GST_CLOCK_TIME_NONE;
num_frames = 0;
frames_to_drop = 0;
iterations_without_frame = 0;
g_mutex_lock(&self->queue_lock);
self->video_format = ::NTV2_FORMAT_UNKNOWN;
while (self->playing && !self->shutdown) {
// If we don't have a video format configured, configure the device now
// and potentially auto-detect the video format
@ -2381,7 +2407,6 @@ restart:
if (self->video_format == ::NTV2_FORMAT_UNKNOWN) {
GST_DEBUG_OBJECT(self, "No signal, waiting");
frames_dropped_last = G_MAXUINT64;
if (have_signal) {
QueueItem item = {
.type = QUEUE_ITEM_TYPE_SIGNAL_CHANGE,
@ -2390,8 +2415,7 @@ restart:
.vpid = 0}};
gst_vec_deque_push_tail_struct(self->queue, &item);
g_cond_signal(&self->queue_cond);
have_signal = FALSE;
discont = TRUE;
have_signal = false;
}
self->device->device->WaitForInputVerticalInterrupt(self->channel);
continue;
@ -2505,7 +2529,6 @@ restart:
GST_DEBUG_OBJECT(self, "No signal, waiting");
g_mutex_unlock(&self->queue_lock);
frames_dropped_last = G_MAXUINT64;
if (have_signal || current_video_format != last_detected_video_format) {
QueueItem item = {
.type = QUEUE_ITEM_TYPE_SIGNAL_CHANGE,
@ -2515,8 +2538,7 @@ restart:
last_detected_video_format = ::NTV2_FORMAT_UNKNOWN;
gst_vec_deque_push_tail_struct(self->queue, &item);
g_cond_signal(&self->queue_cond);
have_signal = FALSE;
discont = TRUE;
have_signal = false;
}
self->device->device->WaitForInputVerticalInterrupt(self->channel);
g_mutex_lock(&self->queue_lock);
@ -2542,7 +2564,6 @@ restart:
current_string.c_str(), configured_string.c_str(),
effective_string.c_str());
g_mutex_unlock(&self->queue_lock);
frames_dropped_last = G_MAXUINT64;
if (have_signal || current_video_format != last_detected_video_format) {
QueueItem item = {
.type = QUEUE_ITEM_TYPE_SIGNAL_CHANGE,
@ -2552,8 +2573,7 @@ restart:
last_detected_video_format = current_video_format;
gst_vec_deque_push_tail_struct(self->queue, &item);
g_cond_signal(&self->queue_cond);
have_signal = FALSE;
discont = TRUE;
have_signal = false;
}
self->device->device->WaitForInputVerticalInterrupt(self->channel);
g_mutex_lock(&self->queue_lock);
@ -2592,7 +2612,7 @@ restart:
GST_TIME_ARGS(status.acRDTSCCurrentTime * 100),
status.acFramesProcessed, status.acFramesDropped, status.acBufferLevel);
if (status.IsRunning() && status.acBufferLevel > 1) {
if (status.IsRunning() && status.HasAvailableInputFrame()) {
GstBuffer *video_buffer = NULL;
GstBuffer *audio_buffer = NULL;
GstBuffer *anc_buffer = NULL, *anc_buffer2 = NULL;
@ -2610,8 +2630,8 @@ restart:
.vpid = vpid_a}};
gst_vec_deque_push_tail_struct(self->queue, &item);
g_cond_signal(&self->queue_cond);
have_signal = TRUE;
reset_clock = true;
have_signal = true;
last_frame_time = GST_CLOCK_TIME_NONE;
// Drop the next frames after signal recovery as the capture times
// are generally just wrong.
@ -2699,16 +2719,6 @@ restart:
continue;
}
if (frames_to_drop > 0) {
GST_TRACE_OBJECT(self, "Dropping frame");
frames_to_drop -= 1;
gst_clear_buffer(&anc_buffer2);
gst_clear_buffer(&anc_buffer);
gst_clear_buffer(&audio_buffer);
gst_clear_buffer(&video_buffer);
continue;
}
const AUTOCIRCULATE_TRANSFER_STATUS &transfer_status =
transfer.GetTransferStatus();
const FRAME_STAMP &frame_stamp = transfer_status.GetFrameStamp();
@ -2735,6 +2745,16 @@ restart:
transfer_status.acFramesDropped,
transfer_status.acBufferLevel);
if (frames_to_drop > 0) {
GST_TRACE_OBJECT(self, "Dropping frame");
frames_to_drop -= 1;
gst_clear_buffer(&anc_buffer2);
gst_clear_buffer(&anc_buffer);
gst_clear_buffer(&audio_buffer);
gst_clear_buffer(&video_buffer);
continue;
}
GstClockTime frame_time_real = frame_stamp.acFrameTime * 100;
// Convert capture time from real-time clock to monotonic clock by
@ -2758,19 +2778,55 @@ restart:
frame_time_monotonic = frame_time_real + diff;
}
// Detect frame drop: backwards capture time is clearly a problem,
// otherwise consider a frame being dropped if more than 1.75 frame
// durations are between two frames
if (last_frame_time != GST_CLOCK_TIME_NONE) {
GstClockTime frame_drop_threshold =
(7 * self->configured_info.fps_d * GST_SECOND) /
(4 * self->configured_info.fps_n);
if (last_frame_time >= frame_time_monotonic) {
GST_ERROR_OBJECT(self, "Frame capture time went backwards");
last_frame_time = GST_CLOCK_TIME_NONE;
} else if (frame_time_monotonic - last_frame_time >
frame_drop_threshold) {
GstClockTime timestamp =
last_frame_time + self->configured_info.fps_d * GST_SECOND /
self->configured_info.fps_n;
GstClockTime timestamp_end = frame_time_monotonic;
GST_WARNING_OBJECT(self,
"Frame drop of %" GST_TIME_FORMAT " detected",
GST_TIME_ARGS(timestamp_end - timestamp));
QueueItem item = {.type = QUEUE_ITEM_TYPE_FRAMES_DROPPED,
.frames_dropped = {.driver_side = TRUE,
.timestamp_start = timestamp,
.timestamp_end = timestamp_end}};
gst_vec_deque_push_tail_struct(self->queue, &item);
g_cond_signal(&self->queue_cond);
last_frame_time = GST_CLOCK_TIME_NONE;
} else {
GST_TRACE_OBJECT(
self, "Time since last frame: %" GST_TIME_FORMAT,
GST_TIME_ARGS(frame_time_monotonic - last_frame_time));
}
}
GstClockTime frame_src_time;
// Update clock mapping
if (first_frame_after_start || reset_clock) {
if (first_frame_after_start || last_frame_time == GST_CLOCK_TIME_NONE) {
GstClockTime internal, external;
guint64 num, denom;
// Keep observations if there was only temporary signal loss or a
// format change as the source is either using the same clock as
// before, or it's different but then our previous configuration would
// be as good/bad as the local monotonic system clock and over some
// frames we would converge to the new clock.
if (!first_frame_after_start) {
// Keep observations if there was only temporary signal loss as the
// source is either using the same clock as before, or it's different
// but then our previous configuration would be as good/bad as the
// local monotonic system clock and over some frames we would converge
// to the new clock.
if (first_frame_after_start) {
// FIXME: Workaround to get rid of all previous observations
g_object_set(self->clock, "window-size", 32, NULL);
}
@ -2787,36 +2843,30 @@ restart:
&denom);
first_frame_time = frame_src_time = gst_clock_adjust_with_calibration(
NULL, frame_time_monotonic, internal, external, num, denom);
first_frame_processed_plus_dropped_minus_buffered =
transfer_status.acFramesProcessed +
transfer_status.acFramesDropped - transfer_status.acBufferLevel;
GST_TRACE_OBJECT(
self,
"Initializing clock with first frame time %" GST_TIME_FORMAT
" and initial frame count %" G_GUINT64_FORMAT,
GST_TIME_ARGS(first_frame_time),
first_frame_processed_plus_dropped_minus_buffered);
self, "Initializing clock with first frame time %" GST_TIME_FORMAT,
GST_TIME_ARGS(first_frame_time));
first_frame_after_start = false;
reset_clock = false;
discont = TRUE;
num_frames = 0;
} else {
gdouble r_squared;
frame_src_time =
first_frame_time +
gst_util_uint64_scale_ceil(
transfer_status.acFramesProcessed +
transfer_status.acFramesDropped -
transfer_status.acBufferLevel -
first_frame_processed_plus_dropped_minus_buffered,
self->configured_info.fps_d * GST_SECOND,
self->configured_info.fps_n);
gst_util_uint64_scale_ceil(num_frames,
self->configured_info.fps_d * GST_SECOND,
self->configured_info.fps_n);
gst_clock_add_observation(self->clock, frame_time_monotonic,
frame_src_time, &r_squared);
}
last_frame_time = frame_time_monotonic;
num_frames += 1;
GstClockTime capture_time;
if (self->clock == clock) {
// If the pipeline is using our clock then we can directly use the
@ -2881,33 +2931,6 @@ restart:
GST_BUFFER_DURATION(audio_buffer) = gst_util_uint64_scale(
GST_SECOND, self->configured_info.fps_d, self->configured_info.fps_n);
if (frames_dropped_last == G_MAXUINT64) {
frames_dropped_last = transfer_status.acFramesDropped;
} else if (frames_dropped_last < transfer_status.acFramesDropped) {
GST_WARNING_OBJECT(
self, "Dropped %" G_GUINT64_FORMAT " frames",
transfer_status.acFramesDropped - frames_dropped_last);
GstClockTime timestamp = gst_util_uint64_scale(
transfer_status.acFramesProcessed + frames_dropped_last,
self->configured_info.fps_n,
self->configured_info.fps_d * GST_SECOND);
GstClockTime timestamp_end = gst_util_uint64_scale(
transfer_status.acFramesProcessed + transfer_status.acFramesDropped,
self->configured_info.fps_n,
self->configured_info.fps_d * GST_SECOND);
QueueItem item = {.type = QUEUE_ITEM_TYPE_FRAMES_DROPPED,
.frames_dropped = {.driver_side = TRUE,
.timestamp_start = timestamp,
.timestamp_end = timestamp_end}};
gst_vec_deque_push_tail_struct(self->queue, &item);
g_cond_signal(&self->queue_cond);
frames_dropped_last = transfer_status.acFramesDropped;
discont = TRUE;
}
gst_buffer_set_size(audio_buffer, transfer.GetCapturedAudioByteCount());
if (anc_buffer)
gst_buffer_set_size(anc_buffer,
@ -2964,7 +2987,7 @@ restart:
gst_vec_deque_drop_struct(self->queue, i, NULL);
gst_vec_deque_push_tail_struct(self->queue, &item);
self->queue_num_frames -= 1;
discont = TRUE;
discont = true;
g_cond_signal(&self->queue_cond);
break;
}
@ -2974,7 +2997,7 @@ restart:
if (discont) {
GST_BUFFER_FLAG_SET(video_buffer, GST_BUFFER_FLAG_DISCONT);
GST_BUFFER_FLAG_SET(audio_buffer, GST_BUFFER_FLAG_DISCONT);
discont = FALSE;
discont = false;
}
QueueItem item = {
@ -3006,7 +3029,6 @@ restart:
if (have_signal && iterations_without_frame < 32) {
iterations_without_frame++;
} else {
frames_dropped_last = G_MAXUINT64;
if (have_signal || last_detected_video_format != current_video_format) {
QueueItem item = {
.type = QUEUE_ITEM_TYPE_SIGNAL_CHANGE,
@ -3016,8 +3038,7 @@ restart:
last_detected_video_format = current_video_format;
gst_vec_deque_push_tail_struct(self->queue, &item);
g_cond_signal(&self->queue_cond);
have_signal = FALSE;
discont = TRUE;
have_signal = false;
}
}