/* * GStreamer * Copyright (C) 2016 Vivia Nikolaidou * * Based on gstvideoframe-audiolevel.c: * Copyright (C) 2015 Vivia Nikolaidou * * 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. */ /** * SECTION:element-avwait * @title: avwait * * This element will drop all buffers until a specific timecode or running * 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 "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 * |[ * gst-launch-1.0 filesrc location="my_file" ! decodebin name=d ! "audio/x-raw" ! avwait name=l target-timecode-str="00:00:04:00" ! autoaudiosink d. ! "video/x-raw" ! timecodestamper ! l. l. ! queue ! timeoverlay time-mode=time-code ! autovideosink * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstavwait.h" #define GST_CAT_DEFAULT gst_avwait_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); static GstStaticPadTemplate audio_sink_template = GST_STATIC_PAD_TEMPLATE ("asink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw") ); static GstStaticPadTemplate audio_src_template = GST_STATIC_PAD_TEMPLATE ("asrc", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw") ); static GstStaticPadTemplate video_sink_template = GST_STATIC_PAD_TEMPLATE ("vsink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/x-raw") ); static GstStaticPadTemplate video_src_template = GST_STATIC_PAD_TEMPLATE ("vsrc", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("video/x-raw") ); #define parent_class gst_avwait_parent_class G_DEFINE_TYPE (GstAvWait, gst_avwait, GST_TYPE_ELEMENT); GST_ELEMENT_REGISTER_DEFINE (avwait, "avwait", GST_RANK_NONE, GST_TYPE_AVWAIT); enum { PROP_0, PROP_TARGET_TIME_CODE, PROP_TARGET_TIME_CODE_STRING, PROP_TARGET_RUNNING_TIME, PROP_END_TIME_CODE, PROP_END_RUNNING_TIME, PROP_RECORDING, PROP_MODE }; #define DEFAULT_TARGET_TIMECODE_STR "00:00:00:00" #define DEFAULT_TARGET_RUNNING_TIME GST_CLOCK_TIME_NONE #define DEFAULT_END_RUNNING_TIME GST_CLOCK_TIME_NONE #define DEFAULT_MODE MODE_TIMECODE /* flags for self->must_send_end_message */ enum { END_MESSAGE_NORMAL = 0, END_MESSAGE_STREAM_ENDED = 1, END_MESSAGE_VIDEO_PUSHED = 2, END_MESSAGE_AUDIO_PUSHED = 4 }; static void gst_avwait_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_avwait_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstFlowReturn gst_avwait_asink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf); static GstFlowReturn gst_avwait_vsink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf); static gboolean gst_avwait_asink_event (GstPad * pad, GstObject * parent, GstEvent * event); static gboolean gst_avwait_vsink_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstIterator *gst_avwait_iterate_internal_links (GstPad * pad, GstObject * parent); static void gst_avwait_finalize (GObject * gobject); static GstStateChangeReturn gst_avwait_change_state (GstElement * element, GstStateChange transition); static GType gst_avwait_mode_get_type (void) { static GType gtype = 0; if (gtype == 0) { static const GEnumValue values[] = { {MODE_TIMECODE, "time code (default)", "timecode"}, {MODE_RUNNING_TIME, "running time", "running-time"}, {MODE_VIDEO_FIRST, "video first", "video-first"}, {0, NULL, NULL} }; gtype = g_enum_register_static ("GstAvWaitMode", values); } return gtype; } static void gst_avwait_class_init (GstAvWaitClass * klass) { GstElementClass *gstelement_class; GObjectClass *gobject_class = (GObjectClass *) klass; GST_DEBUG_CATEGORY_INIT (gst_avwait_debug, "avwait", 0, "avwait"); gstelement_class = (GstElementClass *) klass; gst_element_class_set_static_metadata (gstelement_class, "Timecode Wait", "Filter/Audio/Video", "Drops all audio/video until a specific timecode or running time has been reached", "Vivia Nikolaidou "); gobject_class->set_property = gst_avwait_set_property; gobject_class->get_property = gst_avwait_get_property; g_object_class_install_property (gobject_class, PROP_TARGET_TIME_CODE_STRING, g_param_spec_string ("target-timecode-string", "Target timecode (string)", "Timecode to wait for in timecode mode (string). Must take the form 00:00:00:00", DEFAULT_TARGET_TIMECODE_STR, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_TARGET_TIME_CODE, g_param_spec_boxed ("target-timecode", "Target timecode (object)", "Timecode to wait for in timecode mode (object)", GST_TYPE_VIDEO_TIME_CODE, 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, 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, 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)", "Timecode to end at in timecode mode (object)", GST_TYPE_VIDEO_TIME_CODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_END_RUNNING_TIME, g_param_spec_uint64 ("end-running-time", "End running time", "Running time to end at in running-time mode", 0, G_MAXUINT64, DEFAULT_END_RUNNING_TIME, GST_PARAM_MUTABLE_READY | 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; gst_element_class_add_static_pad_template (gstelement_class, &audio_src_template); gst_element_class_add_static_pad_template (gstelement_class, &audio_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &video_src_template); gst_element_class_add_static_pad_template (gstelement_class, &video_sink_template); gst_type_mark_as_plugin_api (GST_TYPE_AVWAIT_MODE, 0); } static void gst_avwait_init (GstAvWait * self) { self->asinkpad = gst_pad_new_from_static_template (&audio_sink_template, "asink"); gst_pad_set_chain_function (self->asinkpad, GST_DEBUG_FUNCPTR (gst_avwait_asink_chain)); gst_pad_set_event_function (self->asinkpad, GST_DEBUG_FUNCPTR (gst_avwait_asink_event)); gst_pad_set_iterate_internal_links_function (self->asinkpad, GST_DEBUG_FUNCPTR (gst_avwait_iterate_internal_links)); gst_element_add_pad (GST_ELEMENT (self), self->asinkpad); self->vsinkpad = gst_pad_new_from_static_template (&video_sink_template, "vsink"); gst_pad_set_chain_function (self->vsinkpad, GST_DEBUG_FUNCPTR (gst_avwait_vsink_chain)); gst_pad_set_event_function (self->vsinkpad, GST_DEBUG_FUNCPTR (gst_avwait_vsink_event)); gst_pad_set_iterate_internal_links_function (self->vsinkpad, GST_DEBUG_FUNCPTR (gst_avwait_iterate_internal_links)); gst_element_add_pad (GST_ELEMENT (self), self->vsinkpad); self->asrcpad = gst_pad_new_from_static_template (&audio_src_template, "asrc"); gst_pad_set_iterate_internal_links_function (self->asrcpad, GST_DEBUG_FUNCPTR (gst_avwait_iterate_internal_links)); gst_element_add_pad (GST_ELEMENT (self), self->asrcpad); self->vsrcpad = gst_pad_new_from_static_template (&video_src_template, "vsrc"); gst_pad_set_iterate_internal_links_function (self->vsrcpad, GST_DEBUG_FUNCPTR (gst_avwait_iterate_internal_links)); gst_element_add_pad (GST_ELEMENT (self), self->vsrcpad); GST_PAD_SET_PROXY_CAPS (self->asinkpad); GST_PAD_SET_PROXY_ALLOCATION (self->asinkpad); GST_PAD_SET_PROXY_CAPS (self->asrcpad); GST_PAD_SET_PROXY_SCHEDULING (self->asrcpad); GST_PAD_SET_PROXY_CAPS (self->vsinkpad); GST_PAD_SET_PROXY_ALLOCATION (self->vsinkpad); GST_PAD_SET_PROXY_CAPS (self->vsrcpad); GST_PAD_SET_PROXY_SCHEDULING (self->vsrcpad); self->running_time_to_wait_for = GST_CLOCK_TIME_NONE; self->last_seen_video_running_time = GST_CLOCK_TIME_NONE; self->first_audio_running_time = GST_CLOCK_TIME_NONE; self->last_seen_tc = NULL; self->video_eos_flag = FALSE; self->audio_eos_flag = FALSE; self->video_flush_flag = FALSE; self->audio_flush_flag = FALSE; self->shutdown_flag = FALSE; self->dropping = TRUE; 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->end_running_time = DEFAULT_TARGET_RUNNING_TIME; self->mode = DEFAULT_MODE; gst_video_info_init (&self->vinfo); g_mutex_init (&self->mutex); g_cond_init (&self->cond); g_cond_init (&self->audio_cond); } static void gst_avwait_send_element_message (GstAvWait * self, gboolean dropping, GstClockTime running_time) { if (!gst_element_post_message (GST_ELEMENT (self), gst_message_new_element (GST_OBJECT (self), gst_structure_new ("avwait-status", "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 (); } } static GstStateChangeReturn gst_avwait_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret; GstAvWait *self = GST_AVWAIT (element); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: g_mutex_lock (&self->mutex); self->shutdown_flag = TRUE; g_cond_signal (&self->cond); g_cond_signal (&self->audio_cond); g_mutex_unlock (&self->mutex); break; case GST_STATE_CHANGE_READY_TO_PAUSED: g_mutex_lock (&self->mutex); self->shutdown_flag = FALSE; self->video_eos_flag = FALSE; self->audio_eos_flag = FALSE; self->video_flush_flag = FALSE; self->audio_flush_flag = FALSE; self->must_send_end_message = END_MESSAGE_NORMAL; g_mutex_unlock (&self->mutex); default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY:{ gboolean send_message = FALSE; g_mutex_lock (&self->mutex); if (self->mode != MODE_RUNNING_TIME) { 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; send_message = TRUE; } gst_segment_init (&self->asegment, GST_FORMAT_UNDEFINED); self->asegment.position = GST_CLOCK_TIME_NONE; gst_segment_init (&self->vsegment, GST_FORMAT_UNDEFINED); self->vsegment.position = GST_CLOCK_TIME_NONE; gst_video_info_init (&self->vinfo); self->last_seen_video_running_time = GST_CLOCK_TIME_NONE; self->first_audio_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); if (send_message) gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE); break; } default: break; } return ret; } static void gst_avwait_finalize (GObject * object) { GstAvWait *self = GST_AVWAIT (object); if (self->tc) { gst_video_time_code_free (self->tc); self->tc = NULL; } if (self->end_tc) { gst_video_time_code_free (self->end_tc); self->end_tc = NULL; } g_mutex_clear (&self->mutex); g_cond_clear (&self->cond); g_cond_clear (&self->audio_cond); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_avwait_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAvWait *self = GST_AVWAIT (object); switch (prop_id) { case PROP_TARGET_TIME_CODE_STRING:{ g_mutex_lock (&self->mutex); if (self->tc) g_value_take_string (value, gst_video_time_code_to_string (self->tc)); else g_value_set_string (value, DEFAULT_TARGET_TIMECODE_STR); g_mutex_unlock (&self->mutex); break; } case PROP_TARGET_TIME_CODE:{ g_mutex_lock (&self->mutex); g_value_set_boxed (value, self->tc); g_mutex_unlock (&self->mutex); break; } case PROP_END_TIME_CODE:{ g_mutex_lock (&self->mutex); g_value_set_boxed (value, self->end_tc); g_mutex_unlock (&self->mutex); break; } case PROP_TARGET_RUNNING_TIME:{ g_mutex_lock (&self->mutex); g_value_set_uint64 (value, self->target_running_time); g_mutex_unlock (&self->mutex); break; } case PROP_END_RUNNING_TIME:{ g_mutex_lock (&self->mutex); g_value_set_uint64 (value, self->end_running_time); g_mutex_unlock (&self->mutex); break; } case PROP_RECORDING:{ g_mutex_lock (&self->mutex); g_value_set_boolean (value, self->recording); g_mutex_unlock (&self->mutex); break; } case PROP_MODE:{ g_mutex_lock (&self->mutex); g_value_set_enum (value, self->mode); g_mutex_unlock (&self->mutex); break; } default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_avwait_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAvWait *self = GST_AVWAIT (object); switch (prop_id) { case PROP_TARGET_TIME_CODE_STRING:{ gchar **parts; const gchar *tc_str; guint hours, minutes, seconds, frames; tc_str = g_value_get_string (value); parts = g_strsplit (tc_str, ":", 4); if (!parts || parts[3] == NULL) { GST_ERROR_OBJECT (self, "Error: Could not parse timecode %s. Please input a timecode in the form 00:00:00:00", tc_str); g_strfreev (parts); return; } hours = g_ascii_strtoll (parts[0], NULL, 10); minutes = g_ascii_strtoll (parts[1], NULL, 10); seconds = g_ascii_strtoll (parts[2], NULL, 10); frames = g_ascii_strtoll (parts[3], NULL, 10); g_mutex_lock (&self->mutex); if (self->tc) gst_video_time_code_free (self->tc); self->tc = gst_video_time_code_new (0, 1, NULL, 0, hours, minutes, seconds, frames, 0); if (GST_VIDEO_INFO_FORMAT (&self->vinfo) != GST_VIDEO_FORMAT_UNKNOWN && self->vinfo.fps_n != 0) { self->tc->config.fps_n = self->vinfo.fps_n; self->tc->config.fps_d = self->vinfo.fps_d; } g_mutex_unlock (&self->mutex); g_strfreev (parts); break; } case PROP_TARGET_TIME_CODE:{ g_mutex_lock (&self->mutex); if (self->tc) gst_video_time_code_free (self->tc); self->tc = g_value_dup_boxed (value); if (self->tc && self->tc->config.fps_n == 0 && GST_VIDEO_INFO_FORMAT (&self->vinfo) != GST_VIDEO_FORMAT_UNKNOWN && self->vinfo.fps_n != 0) { self->tc->config.fps_n = self->vinfo.fps_n; self->tc->config.fps_d = self->vinfo.fps_d; } g_mutex_unlock (&self->mutex); break; } case PROP_END_TIME_CODE:{ g_mutex_lock (&self->mutex); if (self->end_tc) gst_video_time_code_free (self->end_tc); self->end_tc = g_value_dup_boxed (value); if (self->end_tc && self->end_tc->config.fps_n == 0 && GST_VIDEO_INFO_FORMAT (&self->vinfo) != GST_VIDEO_FORMAT_UNKNOWN && self->vinfo.fps_n != 0) { self->end_tc->config.fps_n = self->vinfo.fps_n; self->end_tc->config.fps_d = self->vinfo.fps_d; } g_mutex_unlock (&self->mutex); break; } case PROP_TARGET_RUNNING_TIME:{ g_mutex_lock (&self->mutex); self->target_running_time = g_value_get_uint64 (value); g_mutex_unlock (&self->mutex); break; } case PROP_END_RUNNING_TIME:{ g_mutex_lock (&self->mutex); self->end_running_time = g_value_get_uint64 (value); g_mutex_unlock (&self->mutex); break; } case PROP_MODE:{ GstAvWaitMode old_mode; g_mutex_lock (&self->mutex); old_mode = self->mode; self->mode = g_value_get_enum (value); if (self->mode != old_mode) { GST_DEBUG_OBJECT (self, "Changing mode from %d to %d", old_mode, self->mode); } g_mutex_unlock (&self->mutex); 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; } } static gboolean gst_avwait_vsink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstAvWait *self = GST_AVWAIT (parent); GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEGMENT:{ GstSegment segment; gboolean send_message = FALSE; gboolean segment_changed; g_mutex_lock (&self->mutex); gst_event_copy_segment (event, &segment); segment.position = self->vsegment.position; segment_changed = !gst_segment_is_equal (&segment, &self->vsegment); self->vsegment = segment; if (self->vsegment.format != GST_FORMAT_TIME) { GST_ERROR_OBJECT (self, "Invalid segment format"); g_mutex_unlock (&self->mutex); gst_event_unref (event); return FALSE; } if (segment_changed) { 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; send_message = TRUE; } self->vsegment.position = GST_CLOCK_TIME_NONE; } g_mutex_unlock (&self->mutex); if (send_message) gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE); break; } case GST_EVENT_GAP: gst_event_unref (event); return TRUE; case GST_EVENT_EOS:{ GstClockTime running_time; gboolean send_message = FALSE; GstClockTime audio_running_time_to_end_at; g_mutex_lock (&self->mutex); self->video_eos_flag = TRUE; /* If we were recording then we'd be done with it at EOS of the video * pad once the audio has caught up, if it has to */ running_time = self->last_seen_video_running_time; if (self->was_recording) { GST_INFO_OBJECT (self, "Recording stopped at EOS 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; self->must_send_end_message |= END_MESSAGE_STREAM_ENDED; } else if (running_time < self->running_time_to_wait_for && self->running_time_to_wait_for != GST_CLOCK_TIME_NONE) { self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE; } } g_cond_signal (&self->cond); if (self->must_send_end_message & END_MESSAGE_AUDIO_PUSHED) { self->must_send_end_message = END_MESSAGE_NORMAL; send_message = TRUE; audio_running_time_to_end_at = self->audio_running_time_to_end_at; } else if (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) { self->must_send_end_message |= END_MESSAGE_VIDEO_PUSHED; } g_mutex_unlock (&self->mutex); if (send_message) gst_avwait_send_element_message (self, TRUE, audio_running_time_to_end_at); break; } case GST_EVENT_FLUSH_START: g_mutex_lock (&self->mutex); self->video_flush_flag = TRUE; g_cond_signal (&self->audio_cond); g_mutex_unlock (&self->mutex); break; case GST_EVENT_FLUSH_STOP:{ gboolean send_message = FALSE; g_mutex_lock (&self->mutex); self->video_flush_flag = FALSE; 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; send_message = TRUE; } gst_segment_init (&self->vsegment, GST_FORMAT_UNDEFINED); self->vsegment.position = GST_CLOCK_TIME_NONE; g_mutex_unlock (&self->mutex); if (send_message) gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE); break; } case GST_EVENT_CAPS:{ GstCaps *caps; gst_event_parse_caps (event, &caps); GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps); g_mutex_lock (&self->mutex); if (!gst_video_info_from_caps (&self->vinfo, caps)) { gst_event_unref (event); g_mutex_unlock (&self->mutex); return FALSE; } if (self->tc && self->tc->config.fps_n == 0 && self->vinfo.fps_n != 0) { self->tc->config.fps_n = self->vinfo.fps_n; self->tc->config.fps_d = self->vinfo.fps_d; } if (self->end_tc && self->end_tc->config.fps_n == 0 && self->vinfo.fps_n != 0) { self->end_tc->config.fps_n = self->vinfo.fps_n; self->end_tc->config.fps_d = self->vinfo.fps_d; } g_mutex_unlock (&self->mutex); break; } default: break; } return gst_pad_event_default (pad, parent, event); } static gboolean gst_avwait_asink_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstAvWait *self = GST_AVWAIT (parent); GST_LOG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEGMENT:{ GstSegment segment; gboolean segment_changed; g_mutex_lock (&self->mutex); gst_event_copy_segment (event, &segment); segment.position = self->asegment.position; segment_changed = !gst_segment_is_equal (&segment, &self->asegment); self->asegment = segment; if (self->asegment.format != GST_FORMAT_TIME) { GST_ERROR_OBJECT (self, "Invalid segment format"); g_mutex_unlock (&self->mutex); gst_event_unref (event); return FALSE; } if (segment_changed) { self->asegment.position = GST_CLOCK_TIME_NONE; } g_mutex_unlock (&self->mutex); break; } case GST_EVENT_FLUSH_START: g_mutex_lock (&self->mutex); self->audio_flush_flag = TRUE; g_cond_signal (&self->cond); g_mutex_unlock (&self->mutex); break; case GST_EVENT_EOS:{ gboolean send_message = FALSE; GstClockTime audio_running_time_to_end_at; g_mutex_lock (&self->mutex); self->audio_eos_flag = TRUE; g_cond_signal (&self->audio_cond); if ((self->must_send_end_message & END_MESSAGE_VIDEO_PUSHED)) { self->must_send_end_message = END_MESSAGE_NORMAL; audio_running_time_to_end_at = self->audio_running_time_to_end_at; send_message = TRUE; } else if (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) { self->must_send_end_message |= END_MESSAGE_AUDIO_PUSHED; } else { self->must_send_end_message = END_MESSAGE_NORMAL; } g_mutex_unlock (&self->mutex); if (send_message) gst_avwait_send_element_message (self, TRUE, audio_running_time_to_end_at); break; } case GST_EVENT_FLUSH_STOP: g_mutex_lock (&self->mutex); self->audio_flush_flag = FALSE; gst_segment_init (&self->asegment, GST_FORMAT_UNDEFINED); self->asegment.position = GST_CLOCK_TIME_NONE; g_mutex_unlock (&self->mutex); break; case GST_EVENT_CAPS:{ GstCaps *caps; gst_event_parse_caps (event, &caps); GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps); g_mutex_lock (&self->mutex); if (!gst_audio_info_from_caps (&self->ainfo, caps)) { g_mutex_unlock (&self->mutex); gst_event_unref (event); return FALSE; } g_mutex_unlock (&self->mutex); break; } default: break; } return gst_pad_event_default (pad, parent, event); } static GstFlowReturn gst_avwait_vsink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf) { GstClockTime timestamp; GstAvWait *self = GST_AVWAIT (parent); GstClockTime running_time; GstVideoTimeCode *tc = NULL; GstVideoTimeCodeMeta *tc_meta; gboolean retry = FALSE; gboolean ret = GST_FLOW_OK; gboolean send_message = FALSE; GstClockTime message_running_time; gboolean message_dropping; timestamp = GST_BUFFER_TIMESTAMP (inbuf); if (timestamp == GST_CLOCK_TIME_NONE) { gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } g_mutex_lock (&self->mutex); self->vsegment.position = timestamp; running_time = gst_segment_to_running_time (&self->vsegment, GST_FORMAT_TIME, self->vsegment.position); self->last_seen_video_running_time = running_time; tc_meta = gst_buffer_get_video_time_code_meta (inbuf); if (tc_meta) { 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; } while (self->mode == MODE_VIDEO_FIRST && self->first_audio_running_time == GST_CLOCK_TIME_NONE && !self->audio_eos_flag && !self->shutdown_flag && !self->video_flush_flag) { GST_DEBUG_OBJECT (self, "Waiting for first audio buffer"); g_cond_wait (&self->audio_cond, &self->mutex); } if (self->video_flush_flag || self->shutdown_flag) { GST_DEBUG_OBJECT (self, "Shutting down, ignoring buffer"); gst_buffer_unref (inbuf); g_mutex_unlock (&self->mutex); return GST_FLOW_FLUSHING; } switch (self->mode) { case MODE_TIMECODE:{ gboolean emit_passthrough_signal = FALSE; if (self->tc && self->end_tc && gst_video_time_code_compare (self->tc, self->end_tc) != -1) { gchar *tc_str, *end_tc; tc_str = gst_video_time_code_to_string (self->tc); end_tc = gst_video_time_code_to_string (self->end_tc); GST_ELEMENT_ERROR (self, LIBRARY, SETTINGS, (NULL), ("End timecode %s must be after start timecode %s. Start timecode rejected", end_tc, tc_str)); g_free (end_tc); g_free (tc_str); gst_buffer_unref (inbuf); g_mutex_unlock (&self->mutex); return GST_FLOW_ERROR; } if (self->tc != NULL && tc != NULL) { if (gst_video_time_code_compare (tc, self->tc) < 0 && self->running_time_to_wait_for == GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (self, "Timecode not yet reached, ignoring frame"); gst_buffer_unref (inbuf); inbuf = NULL; } else if (self->running_time_to_wait_for == GST_CLOCK_TIME_NONE) { gchar *target_tc_str, *tc_str; target_tc_str = gst_video_time_code_to_string (self->tc); tc_str = gst_video_time_code_to_string (tc); GST_INFO_OBJECT (self, "Target timecode %s reached at %s (%" GST_TIME_FORMAT ")", target_tc_str, tc_str, GST_TIME_ARGS (self->vsegment.position)); g_free (target_tc_str); g_free (tc_str); /* Don't emit a signal if we weren't dropping (e.g. settings changed * mid-flight) */ emit_passthrough_signal = self->dropping; self->dropping = FALSE; self->running_time_to_wait_for = running_time; if (self->recording) { self->audio_running_time_to_wait_for = running_time; send_message = TRUE; message_running_time = running_time; message_dropping = FALSE; } } } if (tc != NULL && self->end_tc && gst_video_time_code_compare (tc, self->end_tc) >= 0) { if (self->running_time_to_end_at == GST_CLOCK_TIME_NONE) { gchar *end_tc_str, *tc_str; end_tc_str = gst_video_time_code_to_string (self->end_tc); tc_str = gst_video_time_code_to_string (tc); GST_INFO_OBJECT (self, "End timecode %s reached at %s (%" GST_TIME_FORMAT ")", end_tc_str, tc_str, GST_TIME_ARGS (self->vsegment.position)); g_free (end_tc_str); g_free (tc_str); self->dropping = TRUE; self->running_time_to_end_at = running_time; if (self->recording) { self->audio_running_time_to_end_at = self->running_time_to_end_at; self->must_send_end_message |= END_MESSAGE_STREAM_ENDED; } } if (inbuf) { gst_buffer_unref (inbuf); inbuf = NULL; } emit_passthrough_signal = FALSE; } if (emit_passthrough_signal && self->recording) { send_message = TRUE; message_running_time = self->running_time_to_wait_for; message_dropping = FALSE; } break; } case MODE_RUNNING_TIME:{ gboolean emit_passthrough_signal = FALSE; if (self->target_running_time != GST_CLOCK_TIME_NONE) { if (running_time < self->target_running_time) { GST_DEBUG_OBJECT (self, "Have %" GST_TIME_FORMAT ", waiting for %" GST_TIME_FORMAT, GST_TIME_ARGS (running_time), GST_TIME_ARGS (self->target_running_time)); gst_buffer_unref (inbuf); inbuf = NULL; } else if (self->running_time_to_wait_for == GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (self, "Target running time %" GST_TIME_FORMAT " reached at %" GST_TIME_FORMAT "(%" GST_TIME_FORMAT ")", GST_TIME_ARGS (self->target_running_time), GST_TIME_ARGS (running_time), GST_TIME_ARGS (self->vsegment.position)); /* Don't emit a signal if we weren't dropping (e.g. settings changed * mid-flight) */ emit_passthrough_signal = self->dropping; self->dropping = FALSE; self->running_time_to_wait_for = running_time; if (self->recording) { self->audio_running_time_to_wait_for = running_time; send_message = TRUE; message_running_time = running_time; message_dropping = FALSE; } } } if (GST_CLOCK_TIME_IS_VALID (self->end_running_time) && running_time >= self->end_running_time) { if (self->running_time_to_end_at == GST_CLOCK_TIME_NONE) { GST_INFO_OBJECT (self, "End running time %" GST_TIME_FORMAT " reached at %" GST_TIME_FORMAT "(%" GST_TIME_FORMAT ")", GST_TIME_ARGS (self->end_running_time), GST_TIME_ARGS (running_time), GST_TIME_ARGS (self->vsegment.position)); self->dropping = TRUE; self->running_time_to_end_at = running_time; if (self->recording) { self->audio_running_time_to_end_at = running_time; self->must_send_end_message |= END_MESSAGE_STREAM_ENDED; } } if (inbuf) { gst_buffer_unref (inbuf); inbuf = NULL; } emit_passthrough_signal = FALSE; } if (emit_passthrough_signal && self->recording) { send_message = TRUE; message_running_time = self->running_time_to_wait_for; message_dropping = FALSE; } break; } case MODE_VIDEO_FIRST:{ gboolean emit_passthrough_signal = FALSE; if (self->running_time_to_wait_for == GST_CLOCK_TIME_NONE) { GST_DEBUG_OBJECT (self, "First video running time is %" GST_TIME_FORMAT, GST_TIME_ARGS (running_time)); emit_passthrough_signal = self->dropping; self->dropping = FALSE; self->running_time_to_wait_for = running_time; if (self->recording) { self->audio_running_time_to_wait_for = running_time; send_message = TRUE; message_running_time = running_time; message_dropping = FALSE; } } if (emit_passthrough_signal && self->recording) { send_message = TRUE; message_running_time = self->running_time_to_wait_for; message_dropping = FALSE; } 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 || self->running_time_to_end_at == GST_CLOCK_TIME_NONE)) { /* We just stopped recording: synchronise the audio */ if (self->running_time_to_end_at == GST_CLOCK_TIME_NONE) self->running_time_to_end_at = running_time; self->audio_running_time_to_end_at = running_time; self->must_send_end_message |= END_MESSAGE_STREAM_ENDED; } else if (running_time < self->running_time_to_wait_for && self->running_time_to_wait_for != GST_CLOCK_TIME_NONE) { self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE; } } /* 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 (self->mode != MODE_VIDEO_FIRST || self->first_audio_running_time <= running_time || self->audio_eos_flag) { 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 (self->running_time_to_wait_for != GST_CLOCK_TIME_NONE && running_time > self->running_time_to_wait_for) { /* We just started recording: synchronise the audio */ self->audio_running_time_to_wait_for = running_time; send_message = TRUE; message_running_time = running_time; message_dropping = FALSE; } 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; } } else { /* We are in video-first mode and behind the first audio timestamp. We * should drop all video buffers until the first audio timestamp, so * we can catch up with it. (In timecode mode and running-time mode, we * don't care about when the audio starts, we start as soon as the * target timecode or running time has been reached) */ if (inbuf) { gst_buffer_unref (inbuf); inbuf = NULL; } retry = TRUE; } } } if (!retry) self->was_recording = self->recording; g_cond_signal (&self->cond); g_mutex_unlock (&self->mutex); if (send_message) gst_avwait_send_element_message (self, message_dropping, message_running_time); send_message = FALSE; if (inbuf) { GST_DEBUG_OBJECT (self, "Pass video buffer %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT, GST_TIME_ARGS (gst_segment_to_running_time (&self->vsegment, GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (inbuf))), GST_TIME_ARGS (gst_segment_to_running_time (&self->vsegment, GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (inbuf) + GST_BUFFER_DURATION (inbuf)))); ret = gst_pad_push (self->vsrcpad, inbuf); } else { GST_DEBUG_OBJECT (self, "Dropped video buffer %" GST_TIME_FORMAT, GST_TIME_ARGS (running_time)); } g_mutex_lock (&self->mutex); if (self->must_send_end_message & END_MESSAGE_AUDIO_PUSHED) { self->must_send_end_message = END_MESSAGE_NORMAL; send_message = TRUE; message_dropping = TRUE; message_running_time = self->audio_running_time_to_end_at; } else if (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) { if (self->audio_eos_flag) { self->must_send_end_message = END_MESSAGE_NORMAL; send_message = TRUE; message_dropping = TRUE; message_running_time = self->audio_running_time_to_end_at; } else { self->must_send_end_message |= END_MESSAGE_VIDEO_PUSHED; } } g_mutex_unlock (&self->mutex); if (send_message) gst_avwait_send_element_message (self, message_dropping, message_running_time); return ret; } /* * assumes sign1 and sign2 are either 1 or -1 * returns 0 if sign1*num1 == sign2*num2 * -1 if sign1*num1 < sign2*num2 * 1 if sign1*num1 > sign2*num2 */ static gint gst_avwait_compare_guint64_with_signs (gint sign1, guint64 num1, gint sign2, guint64 num2) { if (sign1 != sign2) return sign1; else if (num1 == num2) return 0; else return num1 > num2 ? sign1 : -sign1; } static GstFlowReturn gst_avwait_asink_chain (GstPad * pad, GstObject * parent, GstBuffer * inbuf) { GstClockTime timestamp; GstAvWait *self = GST_AVWAIT (parent); GstClockTime current_running_time; GstClockTime video_running_time = GST_CLOCK_TIME_NONE; GstClockTime duration; GstClockTime running_time_at_end = GST_CLOCK_TIME_NONE; gint asign, vsign = 1, esign = 1; GstFlowReturn ret = GST_FLOW_OK; /* Make sure the video thread doesn't send the element message before we * actually call gst_pad_push */ gboolean send_element_message = FALSE; timestamp = GST_BUFFER_TIMESTAMP (inbuf); if (timestamp == GST_CLOCK_TIME_NONE) { gst_buffer_unref (inbuf); return GST_FLOW_ERROR; } g_mutex_lock (&self->mutex); self->asegment.position = timestamp; asign = gst_segment_to_running_time_full (&self->asegment, GST_FORMAT_TIME, self->asegment.position, ¤t_running_time); if (asign == 0) { g_mutex_unlock (&self->mutex); gst_buffer_unref (inbuf); GST_ERROR_OBJECT (self, "Could not get current running time"); return GST_FLOW_ERROR; } if (self->first_audio_running_time == GST_CLOCK_TIME_NONE) { self->first_audio_running_time = current_running_time; } g_cond_signal (&self->audio_cond); if (self->vsegment.format == GST_FORMAT_TIME) { vsign = gst_segment_to_running_time_full (&self->vsegment, GST_FORMAT_TIME, self->vsegment.position, &video_running_time); if (vsign == 0) { 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 */ /* Wait if we haven't received video yet */ (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, running_time_at_end, vsign, video_running_time) == 1)) { GST_DEBUG_OBJECT (self, "Waiting for video: audio at %s%" GST_TIME_FORMAT ", video at %s%" GST_TIME_FORMAT, asign < 0 ? "-" : "+", GST_TIME_ARGS (running_time_at_end), vsign < 0 ? "-" : "+", GST_TIME_ARGS (video_running_time)); g_cond_wait (&self->cond, &self->mutex); vsign = gst_segment_to_running_time_full (&self->vsegment, GST_FORMAT_TIME, self->vsegment.position, &video_running_time); if (vsign == 0) { video_running_time = GST_CLOCK_TIME_NONE; } } if (self->audio_flush_flag || self->shutdown_flag) { GST_DEBUG_OBJECT (self, "Shutting down, ignoring frame"); gst_buffer_unref (inbuf); g_mutex_unlock (&self->mutex); return GST_FLOW_FLUSHING; } 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->audio_running_time_to_wait_for) == -1 /* Audio starts after end: drop */ || 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->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; if (current_running_time >= self->audio_running_time_to_end_at && (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) && !(self->must_send_end_message & END_MESSAGE_AUDIO_PUSHED)) { send_element_message = TRUE; } } else if (gst_avwait_compare_guint64_with_signs (esign, running_time_at_end, 1, self->audio_running_time_to_wait_for) >= 0 && gst_avwait_compare_guint64_with_signs (esign, running_time_at_end, 1, self->audio_running_time_to_end_at) == -1) { /* Audio ends after start, but before end: clip */ GstSegment asegment2 = self->asegment; guint64 start; gint ssign; ssign = gst_segment_position_from_running_time_full (&asegment2, GST_FORMAT_TIME, self->audio_running_time_to_wait_for, &start); if (ssign > 0) { asegment2.start = start; } else { /* Starting before the start of the audio segment?! */ /* This shouldn't happen: we already know that the current audio is * inside the segment, and that the end is after the current audio * position */ GST_ELEMENT_ERROR (self, CORE, FAILED, ("Failed to clip audio: it should have started before the current segment"), NULL); } 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->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; gint ssign; ssign = gst_segment_position_from_running_time_full (&asegment2, GST_FORMAT_TIME, self->audio_running_time_to_end_at, &stop); if (ssign > 0) { asegment2.stop = stop; } else { /* Stopping before the start of the audio segment?! */ /* This shouldn't happen: we already know that the current audio is * inside the segment, and that the end is after the current audio * position */ GST_ELEMENT_ERROR (self, CORE, FAILED, ("Failed to clip audio: it should have ended before the current segment"), NULL); } inbuf = gst_audio_buffer_clip (inbuf, &asegment2, self->ainfo.rate, self->ainfo.bpf); if (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) { send_element_message = TRUE; } } else { /* Programming error? Shouldn't happen */ g_assert_not_reached (); } g_mutex_unlock (&self->mutex); if (inbuf) { GstClockTime new_duration = gst_util_uint64_scale (gst_buffer_get_size (inbuf) / self->ainfo.bpf, GST_SECOND, self->ainfo.rate); GstClockTime new_running_time_at_end = gst_segment_to_running_time (&self->asegment, GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (inbuf) + new_duration); GST_DEBUG_OBJECT (self, "Pass audio buffer %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT, GST_TIME_ARGS (gst_segment_to_running_time (&self->asegment, GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (inbuf))), GST_TIME_ARGS (new_running_time_at_end)); ret = gst_pad_push (self->asrcpad, inbuf); } if (send_element_message) { gboolean send_message = FALSE; GstClockTime audio_running_time_to_end_at; g_mutex_lock (&self->mutex); if ((self->must_send_end_message & END_MESSAGE_VIDEO_PUSHED) || self->video_eos_flag) { self->must_send_end_message = END_MESSAGE_NORMAL; send_message = TRUE; audio_running_time_to_end_at = self->audio_running_time_to_end_at; } else if (self->must_send_end_message & END_MESSAGE_STREAM_ENDED) { self->must_send_end_message |= END_MESSAGE_AUDIO_PUSHED; } else { g_assert_not_reached (); } g_mutex_unlock (&self->mutex); if (send_message) gst_avwait_send_element_message (self, TRUE, audio_running_time_to_end_at); } send_element_message = FALSE; return ret; } static GstIterator * gst_avwait_iterate_internal_links (GstPad * pad, GstObject * parent) { GstIterator *it = NULL; GstPad *opad; GValue val = G_VALUE_INIT; GstAvWait *self = GST_AVWAIT (parent); if (self->asinkpad == pad) opad = gst_object_ref (self->asrcpad); else if (self->asrcpad == pad) opad = gst_object_ref (self->asinkpad); else if (self->vsinkpad == pad) opad = gst_object_ref (self->vsrcpad); else if (self->vsrcpad == pad) opad = gst_object_ref (self->vsinkpad); else goto out; g_value_init (&val, GST_TYPE_PAD); g_value_set_object (&val, opad); it = gst_iterator_new_single (GST_TYPE_PAD, &val); g_value_unset (&val); gst_object_unref (opad); out: return it; }