* time has been reached. It will then pass-through both audio and video,
* starting from that specific timecode or running time, making sure that
* audio starts as early as possible after the video (or at the same time as
- * the video). In the "audio-after-video" mode, it only drops audio buffers
- * until video has started.
+ * the video). In the "video-first" mode, it only drops audio buffers until
+ * video has started.
+ *
+ * The "recording" property acts essentially like a valve connected before
+ * everything else. If recording is FALSE, all buffers are dropped regardless
+ * of settings. If recording is TRUE, the other settings (mode,
+ * target-timecode, target-running-time, etc) are taken into account. Audio
+ * will always start and end together with the video, as long as the stream
+ * itself doesn't start too late or end too early.
*
* ## Example launch line
* |[
PROP_TARGET_TIME_CODE_STRING,
PROP_TARGET_RUNNING_TIME,
PROP_END_TIME_CODE,
+ PROP_RECORDING,
PROP_MODE
};
g_param_spec_boxed ("target-timecode", "Target timecode (object)",
"Timecode to wait for in timecode mode (object)",
GST_TYPE_VIDEO_TIME_CODE,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
+ G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TARGET_RUNNING_TIME,
g_param_spec_uint64 ("target-running-time", "Target running time",
"Running time to wait for in running-time mode",
0, G_MAXUINT64,
DEFAULT_TARGET_RUNNING_TIME,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
+ G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode",
"Operation mode: What to wait for",
GST_TYPE_AVWAIT_MODE,
- DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ DEFAULT_MODE,
+ GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
+ G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_END_TIME_CODE,
g_param_spec_boxed ("end-timecode", "End timecode (object)",
GST_TYPE_VIDEO_TIME_CODE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_RECORDING,
+ g_param_spec_boolean ("recording",
+ "Recording state",
+ "Whether the element is stopped or recording. "
+ "If set to FALSE, all buffers will be dropped regardless of settings.",
+ TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
gobject_class->finalize = gst_avwait_finalize;
gstelement_class->change_state = gst_avwait_change_state;
self->tc = gst_video_time_code_new_empty ();
self->end_tc = NULL;
self->running_time_to_end_at = GST_CLOCK_TIME_NONE;
+ self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE;
+ self->audio_running_time_to_end_at = GST_CLOCK_TIME_NONE;
+ self->recording = TRUE;
self->target_running_time = DEFAULT_TARGET_RUNNING_TIME;
self->mode = DEFAULT_MODE;
"dropping", G_TYPE_BOOLEAN, dropping,
"running-time", GST_TYPE_CLOCK_TIME, running_time, NULL)))) {
GST_ERROR_OBJECT (self, "Unable to send element message!");
+ g_assert_not_reached ();
}
}
GST_DEBUG_OBJECT (self, "First time reset in paused to ready");
self->running_time_to_wait_for = GST_CLOCK_TIME_NONE;
self->running_time_to_end_at = GST_CLOCK_TIME_NONE;
+ self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE;
+ self->audio_running_time_to_end_at = GST_CLOCK_TIME_NONE;
}
if (!self->dropping) {
self->dropping = TRUE;
self->vsegment.position = GST_CLOCK_TIME_NONE;
gst_video_info_init (&self->vinfo);
self->last_seen_video_running_time = GST_CLOCK_TIME_NONE;
+ if (self->last_seen_tc)
+ gst_video_time_code_free (self->last_seen_tc);
self->last_seen_tc = NULL;
g_mutex_unlock (&self->mutex);
break;
g_value_set_uint64 (value, self->target_running_time);
break;
}
+ case PROP_RECORDING:{
+ g_value_set_boolean (value, self->recording);
+ break;
+ }
case PROP_MODE:{
g_value_set_enum (value, self->mode);
break;
self->target_running_time = g_value_get_uint64 (value);
if (self->mode == MODE_RUNNING_TIME) {
self->running_time_to_wait_for = self->target_running_time;
+ if (self->recording) {
+ self->audio_running_time_to_wait_for = self->running_time_to_wait_for;
+ }
if (self->target_running_time < self->last_seen_video_running_time) {
self->dropping = TRUE;
- gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE);
}
}
break;
self->tc) < 0) {
self->running_time_to_wait_for = GST_CLOCK_TIME_NONE;
self->dropping = TRUE;
- gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE);
}
break;
case MODE_RUNNING_TIME:
self->running_time_to_wait_for = self->target_running_time;
+ if (self->recording) {
+ self->audio_running_time_to_wait_for =
+ self->running_time_to_wait_for;
+ }
if (self->target_running_time < self->last_seen_video_running_time) {
self->dropping = TRUE;
- gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE);
}
break;
/* Let the chain functions handle the rest */
}
break;
}
+ case PROP_RECORDING:{
+ g_mutex_lock (&self->mutex);
+ self->recording = g_value_get_boolean (value);
+ g_mutex_unlock (&self->mutex);
+ break;
+ }
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
GST_DEBUG_OBJECT (self, "First time reset in video segment");
self->running_time_to_wait_for = GST_CLOCK_TIME_NONE;
self->running_time_to_end_at = GST_CLOCK_TIME_NONE;
+ self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE;
+ self->audio_running_time_to_end_at = GST_CLOCK_TIME_NONE;
if (!self->dropping) {
self->dropping = TRUE;
gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE);
GST_DEBUG_OBJECT (self, "First time reset in video flush");
self->running_time_to_wait_for = GST_CLOCK_TIME_NONE;
self->running_time_to_end_at = GST_CLOCK_TIME_NONE;
+ self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE;
+ self->audio_running_time_to_end_at = GST_CLOCK_TIME_NONE;
if (!self->dropping) {
self->dropping = TRUE;
gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE);
tc_meta = gst_buffer_get_video_time_code_meta (inbuf);
if (tc_meta) {
- tc = &tc_meta->tc;
+ tc = gst_video_time_code_copy (&tc_meta->tc);
+ if (self->last_seen_tc) {
+ gst_video_time_code_free (self->last_seen_tc);
+ }
self->last_seen_tc = tc;
}
switch (self->mode) {
self->running_time_to_wait_for =
gst_segment_to_running_time (&self->vsegment, GST_FORMAT_TIME,
self->vsegment.position);
+ if (self->recording) {
+ self->audio_running_time_to_wait_for =
+ self->running_time_to_wait_for;
+ }
}
if (self->end_tc && gst_video_time_code_compare (tc, self->end_tc) >= 0) {
if (self->running_time_to_end_at == GST_CLOCK_TIME_NONE) {
self->running_time_to_end_at =
gst_segment_to_running_time (&self->vsegment, GST_FORMAT_TIME,
self->vsegment.position);
- gst_avwait_send_element_message (self, TRUE,
- self->running_time_to_end_at);
+ if (self->recording) {
+ self->audio_running_time_to_end_at = self->running_time_to_end_at;
+ gst_avwait_send_element_message (self, TRUE,
+ self->running_time_to_end_at);
+ }
}
gst_buffer_unref (inbuf);
inbuf = NULL;
- } else if (emit_passthrough_signal) {
+ } else if (emit_passthrough_signal && self->recording) {
gst_avwait_send_element_message (self, FALSE,
self->running_time_to_wait_for);
}
} else {
if (self->dropping) {
self->dropping = FALSE;
- gst_avwait_send_element_message (self, FALSE, running_time);
+ if (self->recording)
+ gst_avwait_send_element_message (self, FALSE, running_time);
}
GST_INFO_OBJECT (self,
"Have %" GST_TIME_FORMAT ", waiting for %" GST_TIME_FORMAT,
self->vsegment.position);
GST_DEBUG_OBJECT (self, "First video running time is %" GST_TIME_FORMAT,
GST_TIME_ARGS (self->running_time_to_wait_for));
+ if (self->recording) {
+ self->audio_running_time_to_wait_for = self->running_time_to_wait_for;
+ }
if (self->dropping) {
self->dropping = FALSE;
- gst_avwait_send_element_message (self, FALSE,
- self->running_time_to_wait_for);
+ if (self->recording)
+ gst_avwait_send_element_message (self, FALSE,
+ self->running_time_to_wait_for);
}
}
break;
}
}
+
+ if (!self->recording) {
+ if (self->was_recording) {
+ GST_INFO_OBJECT (self, "Recording stopped at %" GST_TIME_FORMAT,
+ GST_TIME_ARGS (running_time));
+ if (running_time > self->running_time_to_wait_for
+ && running_time <= self->running_time_to_end_at) {
+ /* We just stopped recording: synchronise the audio */
+ self->audio_running_time_to_end_at = running_time;
+ gst_avwait_send_element_message (self, TRUE, running_time);
+ } else if (running_time < self->running_time_to_wait_for
+ && self->running_time_to_wait_for != GST_CLOCK_TIME_NONE) {
+ /* We should set audio_running_time_to_wait_for to a value far enough
+ * in the future, so that it will never be reached. However, setting
+ * it to GST_CLOCK_TIME_NONE would eternally trigger the g_cond_wait
+ * in the audio chain function, causing audio upstream to be queued up
+ * forever. There is already code in place to ensure that audio will
+ * not exceed the video at the same place, so we just set it to
+ * GST_CLOCK_TIME_NONE - 1 here to ensure it will never be reached,
+ * but still not trigger the eternal waiting code */
+ self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE - 1;
+ }
+ }
+ /* Recording is FALSE: we drop all buffers */
+ if (inbuf) {
+ gst_buffer_unref (inbuf);
+ inbuf = NULL;
+ }
+ } else {
+ if (!self->was_recording) {
+ GST_INFO_OBJECT (self,
+ "Recording started at %" GST_TIME_FORMAT " waiting for %"
+ GST_TIME_FORMAT " inbuf %p", GST_TIME_ARGS (running_time),
+ GST_TIME_ARGS (self->running_time_to_wait_for), inbuf);
+ if (running_time < self->running_time_to_end_at ||
+ self->running_time_to_end_at == GST_CLOCK_TIME_NONE) {
+ /* We are before the end of the recording. Check if we just actually
+ * started */
+ if (running_time > self->running_time_to_wait_for) {
+ /* We just started recording: synchronise the audio */
+ self->audio_running_time_to_wait_for = running_time;
+ gst_avwait_send_element_message (self, FALSE, running_time);
+ } else {
+ /* We will start in the future when running_time_to_wait_for is
+ * reached */
+ self->audio_running_time_to_wait_for = self->running_time_to_wait_for;
+ }
+ self->audio_running_time_to_end_at = self->running_time_to_end_at;
+ }
+ }
+ }
+
+ self->was_recording = self->recording;
g_cond_signal (&self->cond);
g_mutex_unlock (&self->mutex);
if (inbuf)
video_running_time = GST_CLOCK_TIME_NONE;
}
}
+ duration =
+ gst_util_uint64_scale (gst_buffer_get_size (inbuf) / self->ainfo.bpf,
+ GST_SECOND, self->ainfo.rate);
+ if (duration != GST_CLOCK_TIME_NONE) {
+ esign =
+ gst_segment_to_running_time_full (&self->asegment, GST_FORMAT_TIME,
+ self->asegment.position + duration, &running_time_at_end);
+ if (esign == 0) {
+ g_mutex_unlock (&self->mutex);
+ GST_ERROR_OBJECT (self, "Could not get running time at end");
+ gst_buffer_unref (inbuf);
+ return GST_FLOW_ERROR;
+ }
+ }
while (!(self->video_eos_flag || self->audio_flush_flag
|| self->shutdown_flag) &&
/* Start at timecode */
(video_running_time == GST_CLOCK_TIME_NONE
/* Wait if audio is after the video: dunno what to do */
|| gst_avwait_compare_guint64_with_signs (asign,
- current_running_time, vsign, video_running_time) == 1
+ running_time_at_end, vsign, video_running_time) == 1
/* Wait if we don't even know what to wait for yet */
- || self->running_time_to_wait_for == GST_CLOCK_TIME_NONE)) {
+ || self->audio_running_time_to_wait_for == GST_CLOCK_TIME_NONE)) {
g_cond_wait (&self->cond, &self->mutex);
vsign =
gst_segment_to_running_time_full (&self->vsegment, GST_FORMAT_TIME,
g_mutex_unlock (&self->mutex);
return GST_FLOW_FLUSHING;
}
- duration =
- gst_util_uint64_scale (gst_buffer_get_size (inbuf) / self->ainfo.bpf,
- GST_SECOND, self->ainfo.rate);
- if (duration != GST_CLOCK_TIME_NONE) {
- esign =
- gst_segment_to_running_time_full (&self->asegment, GST_FORMAT_TIME,
- self->asegment.position + duration, &running_time_at_end);
- if (esign == 0) {
- g_mutex_unlock (&self->mutex);
- GST_ERROR_OBJECT (self, "Could not get running time at end");
- gst_buffer_unref (inbuf);
- return GST_FLOW_ERROR;
- }
- }
- if (self->running_time_to_wait_for == GST_CLOCK_TIME_NONE
+ if (self->audio_running_time_to_wait_for == GST_CLOCK_TIME_NONE
/* Audio ends before start : drop */
|| gst_avwait_compare_guint64_with_signs (esign,
- running_time_at_end, 1, self->running_time_to_wait_for) == -1
+ running_time_at_end, 1, self->audio_running_time_to_wait_for) == -1
/* Audio starts after end: drop */
- || current_running_time >= self->running_time_to_end_at) {
+ || current_running_time >= self->audio_running_time_to_end_at) {
GST_DEBUG_OBJECT (self,
"Dropped an audio buf at %" GST_TIME_FORMAT " waiting for %"
GST_TIME_FORMAT " video time %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_running_time),
- GST_TIME_ARGS (self->running_time_to_wait_for),
+ GST_TIME_ARGS (self->audio_running_time_to_wait_for),
GST_TIME_ARGS (video_running_time));
GST_DEBUG_OBJECT (self, "Would have ended at %i %" GST_TIME_FORMAT,
esign, GST_TIME_ARGS (running_time_at_end));
gst_buffer_unref (inbuf);
inbuf = NULL;
} else if (gst_avwait_compare_guint64_with_signs (esign, running_time_at_end,
- 1, self->running_time_to_wait_for) >= 0
+ 1, self->audio_running_time_to_wait_for) >= 0
&& gst_avwait_compare_guint64_with_signs (esign, running_time_at_end, 1,
- self->running_time_to_end_at) == -1) {
+ self->audio_running_time_to_end_at) == -1) {
/* Audio ends after start, but before end: clip */
GstSegment asegment2 = self->asegment;
gst_segment_set_running_time (&asegment2, GST_FORMAT_TIME,
- self->running_time_to_wait_for);
+ self->audio_running_time_to_wait_for);
inbuf =
gst_audio_buffer_clip (inbuf, &asegment2, self->ainfo.rate,
self->ainfo.bpf);
} else if (gst_avwait_compare_guint64_with_signs (esign, running_time_at_end,
- 1, self->running_time_to_end_at) >= 0) {
+ 1, self->audio_running_time_to_end_at) >= 0) {
/* Audio starts after start, but before end: clip from the other side */
GstSegment asegment2 = self->asegment;
guint64 stop;
ssign =
gst_segment_position_from_running_time_full (&asegment2,
- GST_FORMAT_TIME, self->running_time_to_end_at, &stop);
+ GST_FORMAT_TIME, self->audio_running_time_to_end_at, &stop);
if (ssign > 0) {
asegment2.stop = stop;
} else {
GstClockTime last_seen_video_running_time;
GstVideoTimeCode *last_seen_tc;
+ /* If running_time_to_wait_for has been reached but we are
+ * not recording, audio shouldn't start running. It should
+ * instead start synchronised with the video when we start
+ * recording. Similarly when stopping recording manually vs
+ * when the target timecode has been reached. So we use
+ * different variables for the audio */
+ GstClockTime audio_running_time_to_wait_for;
+ GstClockTime audio_running_time_to_end_at;
+
gboolean video_eos_flag;
gboolean audio_flush_flag;
gboolean shutdown_flag;
gboolean dropping;
+ gboolean recording;
+ gboolean was_recording;
GCond cond;
GMutex mutex;
elements/videoframe-audiolevel \
elements/autoconvert \
elements/autovideoconvert \
+ elements/avwait \
elements/asfmux \
elements/camerabin \
elements/gdppay \
$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \
$(GST_AUDIO_LIBS)
+elements_avwait_CFLAGS = \
+ $(GST_PLUGINS_BASE_CFLAGS) \
+ $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS)
+elements_avwait_LDADD = \
+ $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) $(LDADD) \
+ $(GST_AUDIO_LIBS) $(GST_VIDEO_LIBS)
+
elements_faad_CFLAGS = \
$(GST_PLUGINS_BASE_CFLAGS) \
$(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS)
--- /dev/null
+/* GStreamer unit test for avwait
+ *
+ * Copyright (C) 2018 Vivia Nikolaidou <vivia@toolsonair.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include <config.h>
+#endif
+
+#include <gst/check/gstcheck.h>
+#include <gst/audio/audio.h>
+#include <gst/video/video.h>
+
+typedef enum _SwitchType
+{
+ DO_NOT_SWITCH = -1,
+ SWITCH_FALSE = 0,
+ SWITCH_TRUE = 1
+} SwitchType;
+
+static guint audio_buffer_count, video_buffer_count;
+static SwitchType switch_after_2s;
+static GstVideoTimeCode *target_tc;
+static GstVideoTimeCode *end_tc;
+static GstClockTime target_running_time;
+static gboolean recording;
+static gint mode;
+
+static GstAudioInfo ainfo;
+
+static guint n_abuffers, n_vbuffers;
+static GstClockTime first_audio_timestamp, last_audio_timestamp;
+static GstClockTime first_video_timestamp, last_video_timestamp;
+
+typedef struct _ElementPadAndSwitchType
+{
+ GstElement *element;
+ GstPad *pad;
+ SwitchType switch_after_2s;
+} ElementPadAndSwitchType;
+
+static void
+set_default_params (void)
+{
+ n_abuffers = 16;
+ n_vbuffers = 160;
+ switch_after_2s = DO_NOT_SWITCH;
+ target_tc = NULL;
+ end_tc = NULL;
+ target_running_time = GST_CLOCK_TIME_NONE;
+ recording = TRUE;
+ mode = 2;
+
+ first_audio_timestamp = GST_CLOCK_TIME_NONE;
+ last_audio_timestamp = GST_CLOCK_TIME_NONE;
+ first_video_timestamp = GST_CLOCK_TIME_NONE;
+ last_video_timestamp = GST_CLOCK_TIME_NONE;
+};
+
+static GstFlowReturn
+output_achain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+ GstClockTime timestamp, duration;
+
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+ duration =
+ gst_util_uint64_scale (gst_buffer_get_size (buffer) / ainfo.bpf,
+ GST_SECOND, ainfo.rate);
+
+ if (first_audio_timestamp == GST_CLOCK_TIME_NONE)
+ first_audio_timestamp = timestamp;
+
+ last_audio_timestamp = timestamp + duration;
+
+ audio_buffer_count++;
+ gst_buffer_unref (buffer);
+ return GST_FLOW_OK;
+}
+
+static GstFlowReturn
+output_vchain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+ GstClockTime timestamp;
+
+ timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+ if (first_video_timestamp == GST_CLOCK_TIME_NONE)
+ first_video_timestamp = timestamp;
+
+ last_video_timestamp = timestamp + GST_BUFFER_DURATION (buffer);
+
+ video_buffer_count++;
+ gst_buffer_unref (buffer);
+ return GST_FLOW_OK;
+}
+
+static gpointer
+push_abuffers (gpointer data)
+{
+ GstSegment segment;
+ GstPad *pad = data;
+ gint i;
+ GstClockTime timestamp = 0;
+ GstCaps *caps;
+ guint buf_size = 1000;
+ guint channels = 2;
+
+ gst_pad_send_event (pad, gst_event_new_stream_start ("test"));
+
+ gst_audio_info_set_format (&ainfo, GST_AUDIO_FORMAT_S8, buf_size, channels,
+ NULL);
+ caps = gst_audio_info_to_caps (&ainfo);
+ gst_pad_send_event (pad, gst_event_new_caps (caps));
+ gst_caps_unref (caps);
+
+ gst_segment_init (&segment, GST_FORMAT_TIME);
+ gst_pad_send_event (pad, gst_event_new_segment (&segment));
+
+ for (i = 0; i < n_abuffers; i++) {
+ GstBuffer *buf = gst_buffer_new_and_alloc (channels * buf_size);
+
+ gst_buffer_memset (buf, 0, 0, channels * buf_size);
+
+ GST_BUFFER_TIMESTAMP (buf) = timestamp;
+ timestamp += 1 * GST_SECOND;
+ GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf);
+
+ fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK);
+ }
+ gst_pad_send_event (pad, gst_event_new_eos ());
+
+ return NULL;
+}
+
+static gpointer
+push_vbuffers (gpointer data)
+{
+ GstSegment segment;
+ ElementPadAndSwitchType *e = data;
+ GstPad *pad = e->pad;
+ gint i;
+ GstClockTime timestamp = 0;
+ GstVideoTimeCode *tc;
+
+ gst_pad_send_event (pad, gst_event_new_stream_start ("test"));
+ gst_segment_init (&segment, GST_FORMAT_TIME);
+ gst_pad_send_event (pad, gst_event_new_segment (&segment));
+ tc = gst_video_time_code_new (40, 1, NULL, GST_VIDEO_TIME_CODE_FLAGS_NONE, 0,
+ 0, 0, 0, 0);
+
+ for (i = 0; i < n_vbuffers; i++) {
+ GstBuffer *buf = gst_buffer_new_and_alloc (1000);
+
+ gst_buffer_memset (buf, 0, i, 1);
+
+ GST_BUFFER_TIMESTAMP (buf) = timestamp;
+ timestamp += 25 * GST_MSECOND;
+ GST_BUFFER_DURATION (buf) = timestamp - GST_BUFFER_TIMESTAMP (buf);
+ gst_buffer_add_video_time_code_meta (buf, tc);
+ gst_video_time_code_increment_frame (tc);
+
+ fail_unless (gst_pad_chain (pad, buf) == GST_FLOW_OK);
+
+ if (timestamp == 2 * GST_SECOND && e->switch_after_2s != DO_NOT_SWITCH) {
+ g_object_set (e->element, "recording", ! !e->switch_after_2s, NULL);
+ }
+ }
+ gst_pad_send_event (pad, gst_event_new_eos ());
+ gst_video_time_code_free (tc);
+
+ return NULL;
+}
+
+static void
+test_avwait_generic (void)
+{
+ GstElement *avwait;
+ GstPad *asink, *vsink, *asrc, *vsrc, *aoutput_sink, *voutput_sink;
+ GThread *athread, *vthread;
+ GstBus *bus;
+ ElementPadAndSwitchType *e;
+
+ audio_buffer_count = 0;
+ video_buffer_count = 0;
+
+ avwait = gst_element_factory_make ("avwait", NULL);
+ fail_unless (avwait != NULL);
+ g_object_set (avwait, "mode", mode,
+ "target-running-time", target_running_time, "recording", recording, NULL);
+ if (target_tc != NULL)
+ g_object_set (avwait, "target-timecode", target_tc, NULL);
+ if (end_tc != NULL)
+ g_object_set (avwait, "end-timecode", end_tc, NULL);
+
+ bus = gst_bus_new ();
+ gst_element_set_bus (avwait, bus);
+
+ asink = gst_element_get_static_pad (avwait, "asink");
+ fail_unless (asink != NULL);
+
+ vsink = gst_element_get_static_pad (avwait, "vsink");
+ fail_unless (vsink != NULL);
+
+ asrc = gst_element_get_static_pad (avwait, "asrc");
+ aoutput_sink = gst_pad_new ("sink", GST_PAD_SINK);
+ fail_unless (aoutput_sink != NULL);
+ fail_unless (gst_pad_link (asrc, aoutput_sink) == GST_PAD_LINK_OK);
+
+ vsrc = gst_element_get_static_pad (avwait, "vsrc");
+ voutput_sink = gst_pad_new ("sink", GST_PAD_SINK);
+ fail_unless (voutput_sink != NULL);
+ fail_unless (gst_pad_link (vsrc, voutput_sink) == GST_PAD_LINK_OK);
+
+ gst_pad_set_chain_function (aoutput_sink, output_achain);
+
+ gst_pad_set_chain_function (voutput_sink, output_vchain);
+
+ gst_pad_set_active (aoutput_sink, TRUE);
+ gst_pad_set_active (voutput_sink, TRUE);
+ fail_unless (gst_element_set_state (avwait,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
+ e = g_new0 (ElementPadAndSwitchType, 1);
+ e->element = avwait;
+ e->pad = vsink;
+ e->switch_after_2s = switch_after_2s;
+
+ athread = g_thread_new ("athread", (GThreadFunc) push_abuffers, asink);
+ vthread = g_thread_new ("vthread", (GThreadFunc) push_vbuffers, e);
+
+ g_thread_join (vthread);
+ g_thread_join (athread);
+
+ /* teardown */
+ gst_element_set_state (avwait, GST_STATE_NULL);
+ gst_bus_set_flushing (bus, TRUE);
+ gst_object_unref (bus);
+ g_free (e);
+ gst_pad_unlink (asrc, aoutput_sink);
+ gst_object_unref (asrc);
+ gst_pad_unlink (vsrc, voutput_sink);
+ gst_object_unref (vsrc);
+ gst_object_unref (asink);
+ gst_object_unref (vsink);
+ gst_pad_set_active (aoutput_sink, FALSE);
+ gst_object_unref (aoutput_sink);
+ gst_pad_set_active (voutput_sink, FALSE);
+ gst_object_unref (voutput_sink);
+ gst_object_unref (avwait);
+}
+
+GST_START_TEST (test_avwait_switch_to_true)
+{
+ set_default_params ();
+ recording = FALSE;
+ switch_after_2s = SWITCH_TRUE;
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, 2 * GST_SECOND);
+ fail_unless_equals_uint64 (first_video_timestamp, 2 * GST_SECOND);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_switch_to_false)
+{
+ set_default_params ();
+ recording = TRUE;
+ switch_after_2s = SWITCH_FALSE;
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, 0);
+ fail_unless_equals_uint64 (first_video_timestamp, 0);
+ fail_unless_equals_uint64 (last_video_timestamp, 2 * GST_SECOND);
+ fail_unless_equals_uint64 (last_audio_timestamp, 2 * GST_SECOND);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_1s_switch_to_true)
+{
+ set_default_params ();
+ recording = FALSE;
+ switch_after_2s = SWITCH_TRUE;
+ mode = 1;
+ target_running_time = 1 * GST_SECOND;
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, 2 * GST_SECOND);
+ fail_unless_equals_uint64 (first_video_timestamp, 2 * GST_SECOND);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_1s_switch_to_false)
+{
+ set_default_params ();
+ recording = TRUE;
+ switch_after_2s = SWITCH_FALSE;
+ mode = 1;
+ target_running_time = 1 * GST_SECOND;
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, 1 * GST_SECOND);
+ fail_unless_equals_uint64 (first_video_timestamp, 1 * GST_SECOND);
+ fail_unless_equals_uint64 (last_video_timestamp, 2 * GST_SECOND);
+ fail_unless_equals_uint64 (last_audio_timestamp, 2 * GST_SECOND);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_3s_switch_to_true)
+{
+ set_default_params ();
+ recording = FALSE;
+ switch_after_2s = SWITCH_TRUE;
+ mode = 1;
+ target_running_time = 3 * GST_SECOND;
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, 3 * GST_SECOND);
+ fail_unless_equals_uint64 (first_video_timestamp, 3 * GST_SECOND);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_3s_switch_to_false)
+{
+ set_default_params ();
+ recording = TRUE;
+ switch_after_2s = SWITCH_FALSE;
+ mode = 1;
+ target_running_time = 3 * GST_SECOND;
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, GST_CLOCK_TIME_NONE);
+ fail_unless_equals_uint64 (first_video_timestamp, GST_CLOCK_TIME_NONE);
+ fail_unless_equals_uint64 (last_audio_timestamp, GST_CLOCK_TIME_NONE);
+ fail_unless_equals_uint64 (last_video_timestamp, GST_CLOCK_TIME_NONE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_1stc_switch_to_true)
+{
+ set_default_params ();
+ recording = FALSE;
+ switch_after_2s = SWITCH_TRUE;
+ mode = 0;
+ target_tc =
+ gst_video_time_code_new (40, 1, NULL, GST_VIDEO_TIME_CODE_FLAGS_NONE, 0,
+ 0, 1, 0, 0);
+ end_tc =
+ gst_video_time_code_new (40, 1, NULL, GST_VIDEO_TIME_CODE_FLAGS_NONE, 0,
+ 0, 3, 0, 0);
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, 2 * GST_SECOND);
+ fail_unless_equals_uint64 (first_video_timestamp, 2 * GST_SECOND);
+ fail_unless_equals_uint64 (last_video_timestamp, 3 * GST_SECOND);
+ fail_unless_equals_uint64 (last_audio_timestamp, 3 * GST_SECOND);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_1stc_switch_to_false)
+{
+ set_default_params ();
+ recording = TRUE;
+ switch_after_2s = SWITCH_FALSE;
+ mode = 0;
+ target_tc =
+ gst_video_time_code_new (40, 1, NULL, GST_VIDEO_TIME_CODE_FLAGS_NONE, 0,
+ 0, 1, 0, 0);
+ end_tc =
+ gst_video_time_code_new (40, 1, NULL, GST_VIDEO_TIME_CODE_FLAGS_NONE, 0,
+ 0, 3, 0, 0);
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, 1 * GST_SECOND);
+ fail_unless_equals_uint64 (first_video_timestamp, 1 * GST_SECOND);
+ fail_unless_equals_uint64 (last_video_timestamp, 2 * GST_SECOND);
+ fail_unless_equals_uint64 (last_audio_timestamp, 2 * GST_SECOND);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_3stc_switch_to_true)
+{
+ set_default_params ();
+ recording = FALSE;
+ switch_after_2s = SWITCH_TRUE;
+ mode = 0;
+ target_tc =
+ gst_video_time_code_new (40, 1, NULL, GST_VIDEO_TIME_CODE_FLAGS_NONE, 0,
+ 0, 3, 0, 0);
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, 3 * GST_SECOND);
+ fail_unless_equals_uint64 (first_video_timestamp, 3 * GST_SECOND);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_avwait_3stc_switch_to_false)
+{
+ set_default_params ();
+ recording = TRUE;
+ switch_after_2s = SWITCH_FALSE;
+ mode = 0;
+ target_tc =
+ gst_video_time_code_new (40, 1, NULL, GST_VIDEO_TIME_CODE_FLAGS_NONE, 0,
+ 0, 3, 0, 0);
+ test_avwait_generic ();
+ fail_unless_equals_uint64 (first_audio_timestamp, GST_CLOCK_TIME_NONE);
+ fail_unless_equals_uint64 (first_video_timestamp, GST_CLOCK_TIME_NONE);
+ fail_unless_equals_uint64 (last_audio_timestamp, GST_CLOCK_TIME_NONE);
+ fail_unless_equals_uint64 (last_video_timestamp, GST_CLOCK_TIME_NONE);
+}
+
+GST_END_TEST;
+
+
+static Suite *
+avwait_suite (void)
+{
+ Suite *s = suite_create ("avwait");
+ TCase *tc_chain;
+
+ tc_chain = tcase_create ("avwait");
+ tcase_add_test (tc_chain, test_avwait_switch_to_true);
+ tcase_add_test (tc_chain, test_avwait_switch_to_false);
+ tcase_add_test (tc_chain, test_avwait_1s_switch_to_true);
+ tcase_add_test (tc_chain, test_avwait_1s_switch_to_false);
+ tcase_add_test (tc_chain, test_avwait_3s_switch_to_true);
+ tcase_add_test (tc_chain, test_avwait_3s_switch_to_false);
+ tcase_add_test (tc_chain, test_avwait_1stc_switch_to_true);
+ tcase_add_test (tc_chain, test_avwait_1stc_switch_to_false);
+ tcase_add_test (tc_chain, test_avwait_3stc_switch_to_true);
+ tcase_add_test (tc_chain, test_avwait_3stc_switch_to_false);
+ suite_add_tcase (s, tc_chain);
+
+ return s;
+}
+
+GST_CHECK_MAIN (avwait);
projects / platform / upstream / gstreamer.git / commitdiff