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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1403 lines
48 KiB
C
1403 lines
48 KiB
C
/*
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* GStreamer
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* Copyright (C) 2016 Vivia Nikolaidou <vivia@toolsonair.com>
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*
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* Based on gstvideoframe-audiolevel.c:
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* Copyright (C) 2015 Vivia Nikolaidou <vivia@toolsonair.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-avwait
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* @title: avwait
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*
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* This element will drop all buffers until a specific timecode or running
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* time has been reached. It will then pass-through both audio and video,
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* starting from that specific timecode or running time, making sure that
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* audio starts as early as possible after the video (or at the same time as
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* the video). In the "video-first" mode, it only drops audio buffers until
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* video has started.
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*
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* The "recording" property acts essentially like a valve connected before
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* everything else. If recording is FALSE, all buffers are dropped regardless
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* of settings. If recording is TRUE, the other settings (mode,
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* target-timecode, target-running-time, etc) are taken into account. Audio
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* will always start and end together with the video, as long as the stream
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* itself doesn't start too late or end too early.
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*
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* ## Example launch line
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* |[
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* 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
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstavwait.h"
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#define GST_CAT_DEFAULT gst_avwait_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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static GstStaticPadTemplate audio_sink_template =
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GST_STATIC_PAD_TEMPLATE ("asink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw")
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);
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static GstStaticPadTemplate audio_src_template =
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GST_STATIC_PAD_TEMPLATE ("asrc",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw")
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);
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static GstStaticPadTemplate video_sink_template =
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GST_STATIC_PAD_TEMPLATE ("vsink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("video/x-raw")
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);
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static GstStaticPadTemplate video_src_template =
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GST_STATIC_PAD_TEMPLATE ("vsrc",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("video/x-raw")
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);
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#define parent_class gst_avwait_parent_class
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G_DEFINE_TYPE (GstAvWait, gst_avwait, GST_TYPE_ELEMENT);
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GST_ELEMENT_REGISTER_DEFINE (avwait, "avwait", GST_RANK_NONE, GST_TYPE_AVWAIT);
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enum
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{
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PROP_0,
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PROP_TARGET_TIME_CODE,
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PROP_TARGET_TIME_CODE_STRING,
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PROP_TARGET_RUNNING_TIME,
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PROP_END_TIME_CODE,
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PROP_END_RUNNING_TIME,
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PROP_RECORDING,
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PROP_MODE
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};
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#define DEFAULT_TARGET_TIMECODE_STR "00:00:00:00"
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#define DEFAULT_TARGET_RUNNING_TIME GST_CLOCK_TIME_NONE
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#define DEFAULT_END_RUNNING_TIME GST_CLOCK_TIME_NONE
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#define DEFAULT_MODE MODE_TIMECODE
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/* flags for self->must_send_end_message */
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enum
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{
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END_MESSAGE_NORMAL = 0,
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END_MESSAGE_STREAM_ENDED = 1,
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END_MESSAGE_VIDEO_PUSHED = 2,
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END_MESSAGE_AUDIO_PUSHED = 4
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};
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static void gst_avwait_set_property (GObject * object,
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guint prop_id, const GValue * value, GParamSpec * pspec);
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static void gst_avwait_get_property (GObject * object,
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guint prop_id, GValue * value, GParamSpec * pspec);
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static GstFlowReturn gst_avwait_asink_chain (GstPad * pad,
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GstObject * parent, GstBuffer * inbuf);
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static GstFlowReturn gst_avwait_vsink_chain (GstPad * pad,
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GstObject * parent, GstBuffer * inbuf);
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static gboolean gst_avwait_asink_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static gboolean gst_avwait_vsink_event (GstPad * pad,
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GstObject * parent, GstEvent * event);
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static GstIterator *gst_avwait_iterate_internal_links (GstPad *
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pad, GstObject * parent);
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static void gst_avwait_finalize (GObject * gobject);
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static GstStateChangeReturn gst_avwait_change_state (GstElement *
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element, GstStateChange transition);
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static GType
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gst_avwait_mode_get_type (void)
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{
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static GType gtype = 0;
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if (gtype == 0) {
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static const GEnumValue values[] = {
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{MODE_TIMECODE, "time code (default)", "timecode"},
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{MODE_RUNNING_TIME, "running time", "running-time"},
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{MODE_VIDEO_FIRST, "video first", "video-first"},
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{0, NULL, NULL}
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};
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gtype = g_enum_register_static ("GstAvWaitMode", values);
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}
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return gtype;
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}
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static void
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gst_avwait_class_init (GstAvWaitClass * klass)
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{
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GstElementClass *gstelement_class;
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GST_DEBUG_CATEGORY_INIT (gst_avwait_debug, "avwait", 0, "avwait");
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gstelement_class = (GstElementClass *) klass;
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gst_element_class_set_static_metadata (gstelement_class,
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"Timecode Wait", "Filter/Audio/Video",
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"Drops all audio/video until a specific timecode or running time has been reached",
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"Vivia Nikolaidou <vivia@toolsonair.com>");
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gobject_class->set_property = gst_avwait_set_property;
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gobject_class->get_property = gst_avwait_get_property;
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g_object_class_install_property (gobject_class, PROP_TARGET_TIME_CODE_STRING,
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g_param_spec_string ("target-timecode-string", "Target timecode (string)",
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"Timecode to wait for in timecode mode (string). Must take the form 00:00:00:00",
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DEFAULT_TARGET_TIMECODE_STR,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TARGET_TIME_CODE,
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g_param_spec_boxed ("target-timecode", "Target timecode (object)",
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"Timecode to wait for in timecode mode (object)",
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GST_TYPE_VIDEO_TIME_CODE,
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GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_TARGET_RUNNING_TIME,
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g_param_spec_uint64 ("target-running-time", "Target running time",
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"Running time to wait for in running-time mode",
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0, G_MAXUINT64,
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DEFAULT_TARGET_RUNNING_TIME,
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GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_MODE,
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g_param_spec_enum ("mode", "Mode",
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"Operation mode: What to wait for",
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GST_TYPE_AVWAIT_MODE,
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DEFAULT_MODE,
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GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_END_TIME_CODE,
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g_param_spec_boxed ("end-timecode", "End timecode (object)",
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"Timecode to end at in timecode mode (object)",
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GST_TYPE_VIDEO_TIME_CODE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_END_RUNNING_TIME,
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g_param_spec_uint64 ("end-running-time", "End running time",
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"Running time to end at in running-time mode",
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0, G_MAXUINT64,
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DEFAULT_END_RUNNING_TIME,
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GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RECORDING,
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g_param_spec_boolean ("recording",
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"Recording state",
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"Whether the element is stopped or recording. "
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"If set to FALSE, all buffers will be dropped regardless of settings.",
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TRUE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gobject_class->finalize = gst_avwait_finalize;
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gstelement_class->change_state = gst_avwait_change_state;
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gst_element_class_add_static_pad_template (gstelement_class,
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&audio_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&audio_sink_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&video_src_template);
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gst_element_class_add_static_pad_template (gstelement_class,
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&video_sink_template);
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gst_type_mark_as_plugin_api (GST_TYPE_AVWAIT_MODE, 0);
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}
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static void
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gst_avwait_init (GstAvWait * self)
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{
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self->asinkpad =
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gst_pad_new_from_static_template (&audio_sink_template, "asink");
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gst_pad_set_chain_function (self->asinkpad,
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GST_DEBUG_FUNCPTR (gst_avwait_asink_chain));
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gst_pad_set_event_function (self->asinkpad,
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GST_DEBUG_FUNCPTR (gst_avwait_asink_event));
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gst_pad_set_iterate_internal_links_function (self->asinkpad,
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GST_DEBUG_FUNCPTR (gst_avwait_iterate_internal_links));
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gst_element_add_pad (GST_ELEMENT (self), self->asinkpad);
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self->vsinkpad =
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gst_pad_new_from_static_template (&video_sink_template, "vsink");
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gst_pad_set_chain_function (self->vsinkpad,
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GST_DEBUG_FUNCPTR (gst_avwait_vsink_chain));
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gst_pad_set_event_function (self->vsinkpad,
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GST_DEBUG_FUNCPTR (gst_avwait_vsink_event));
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gst_pad_set_iterate_internal_links_function (self->vsinkpad,
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GST_DEBUG_FUNCPTR (gst_avwait_iterate_internal_links));
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gst_element_add_pad (GST_ELEMENT (self), self->vsinkpad);
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self->asrcpad =
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gst_pad_new_from_static_template (&audio_src_template, "asrc");
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gst_pad_set_iterate_internal_links_function (self->asrcpad,
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GST_DEBUG_FUNCPTR (gst_avwait_iterate_internal_links));
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gst_element_add_pad (GST_ELEMENT (self), self->asrcpad);
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self->vsrcpad =
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gst_pad_new_from_static_template (&video_src_template, "vsrc");
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gst_pad_set_iterate_internal_links_function (self->vsrcpad,
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GST_DEBUG_FUNCPTR (gst_avwait_iterate_internal_links));
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gst_element_add_pad (GST_ELEMENT (self), self->vsrcpad);
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GST_PAD_SET_PROXY_CAPS (self->asinkpad);
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GST_PAD_SET_PROXY_ALLOCATION (self->asinkpad);
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GST_PAD_SET_PROXY_CAPS (self->asrcpad);
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GST_PAD_SET_PROXY_SCHEDULING (self->asrcpad);
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GST_PAD_SET_PROXY_CAPS (self->vsinkpad);
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GST_PAD_SET_PROXY_ALLOCATION (self->vsinkpad);
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GST_PAD_SET_PROXY_CAPS (self->vsrcpad);
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GST_PAD_SET_PROXY_SCHEDULING (self->vsrcpad);
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self->running_time_to_wait_for = GST_CLOCK_TIME_NONE;
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self->last_seen_video_running_time = GST_CLOCK_TIME_NONE;
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self->first_audio_running_time = GST_CLOCK_TIME_NONE;
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self->last_seen_tc = NULL;
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self->video_eos_flag = FALSE;
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self->audio_eos_flag = FALSE;
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self->video_flush_flag = FALSE;
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self->audio_flush_flag = FALSE;
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self->shutdown_flag = FALSE;
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self->dropping = TRUE;
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self->tc = gst_video_time_code_new_empty ();
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self->end_tc = NULL;
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self->running_time_to_end_at = GST_CLOCK_TIME_NONE;
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self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE;
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self->audio_running_time_to_end_at = GST_CLOCK_TIME_NONE;
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self->recording = TRUE;
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self->target_running_time = DEFAULT_TARGET_RUNNING_TIME;
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self->end_running_time = DEFAULT_TARGET_RUNNING_TIME;
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self->mode = DEFAULT_MODE;
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gst_video_info_init (&self->vinfo);
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g_mutex_init (&self->mutex);
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g_cond_init (&self->cond);
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g_cond_init (&self->audio_cond);
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}
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static void
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gst_avwait_send_element_message (GstAvWait * self, gboolean dropping,
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GstClockTime running_time)
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{
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if (!gst_element_post_message (GST_ELEMENT (self),
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gst_message_new_element (GST_OBJECT (self),
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gst_structure_new ("avwait-status",
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"dropping", G_TYPE_BOOLEAN, dropping,
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"running-time", GST_TYPE_CLOCK_TIME, running_time, NULL)))) {
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GST_ERROR_OBJECT (self, "Unable to send element message!");
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g_assert_not_reached ();
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}
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}
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static GstStateChangeReturn
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gst_avwait_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstAvWait *self = GST_AVWAIT (element);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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g_mutex_lock (&self->mutex);
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self->shutdown_flag = TRUE;
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g_cond_signal (&self->cond);
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g_cond_signal (&self->audio_cond);
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g_mutex_unlock (&self->mutex);
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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g_mutex_lock (&self->mutex);
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self->shutdown_flag = FALSE;
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self->video_eos_flag = FALSE;
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self->audio_eos_flag = FALSE;
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self->video_flush_flag = FALSE;
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self->audio_flush_flag = FALSE;
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self->must_send_end_message = END_MESSAGE_NORMAL;
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g_mutex_unlock (&self->mutex);
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:{
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gboolean send_message = FALSE;
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g_mutex_lock (&self->mutex);
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if (self->mode != MODE_RUNNING_TIME) {
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GST_DEBUG_OBJECT (self, "First time reset in paused to ready");
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self->running_time_to_wait_for = GST_CLOCK_TIME_NONE;
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self->running_time_to_end_at = GST_CLOCK_TIME_NONE;
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self->audio_running_time_to_wait_for = GST_CLOCK_TIME_NONE;
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self->audio_running_time_to_end_at = GST_CLOCK_TIME_NONE;
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}
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if (!self->dropping) {
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self->dropping = TRUE;
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send_message = TRUE;
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}
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gst_segment_init (&self->asegment, GST_FORMAT_UNDEFINED);
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self->asegment.position = GST_CLOCK_TIME_NONE;
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gst_segment_init (&self->vsegment, GST_FORMAT_UNDEFINED);
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self->vsegment.position = GST_CLOCK_TIME_NONE;
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gst_video_info_init (&self->vinfo);
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self->last_seen_video_running_time = GST_CLOCK_TIME_NONE;
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self->first_audio_running_time = GST_CLOCK_TIME_NONE;
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if (self->last_seen_tc)
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gst_video_time_code_free (self->last_seen_tc);
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self->last_seen_tc = NULL;
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g_mutex_unlock (&self->mutex);
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if (send_message)
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gst_avwait_send_element_message (self, TRUE, GST_CLOCK_TIME_NONE);
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break;
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}
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default:
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break;
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}
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return ret;
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}
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static void
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gst_avwait_finalize (GObject * object)
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{
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GstAvWait *self = GST_AVWAIT (object);
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if (self->tc) {
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gst_video_time_code_free (self->tc);
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self->tc = NULL;
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}
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if (self->end_tc) {
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gst_video_time_code_free (self->end_tc);
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self->end_tc = NULL;
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}
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g_mutex_clear (&self->mutex);
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g_cond_clear (&self->cond);
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g_cond_clear (&self->audio_cond);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_avwait_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAvWait *self = GST_AVWAIT (object);
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switch (prop_id) {
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case PROP_TARGET_TIME_CODE_STRING:{
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g_mutex_lock (&self->mutex);
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if (self->tc)
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g_value_take_string (value, gst_video_time_code_to_string (self->tc));
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else
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g_value_set_string (value, DEFAULT_TARGET_TIMECODE_STR);
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g_mutex_unlock (&self->mutex);
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break;
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}
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case PROP_TARGET_TIME_CODE:{
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g_mutex_lock (&self->mutex);
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g_value_set_boxed (value, self->tc);
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g_mutex_unlock (&self->mutex);
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break;
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}
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case PROP_END_TIME_CODE:{
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g_mutex_lock (&self->mutex);
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g_value_set_boxed (value, self->end_tc);
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g_mutex_unlock (&self->mutex);
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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);
|
|
if (self->mode == MODE_RUNNING_TIME) {
|
|
if (self->target_running_time > self->last_seen_video_running_time) {
|
|
self->dropping = TRUE;
|
|
}
|
|
}
|
|
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);
|
|
if (self->mode == MODE_RUNNING_TIME) {
|
|
if (self->end_running_time >= self->last_seen_video_running_time) {
|
|
self->dropping = TRUE;
|
|
}
|
|
}
|
|
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) {
|
|
switch (self->mode) {
|
|
case MODE_TIMECODE:
|
|
if (self->last_seen_tc && self->tc &&
|
|
gst_video_time_code_compare (self->last_seen_tc,
|
|
self->tc) < 0) {
|
|
self->running_time_to_wait_for = GST_CLOCK_TIME_NONE;
|
|
self->running_time_to_end_at = GST_CLOCK_TIME_NONE;
|
|
self->dropping = TRUE;
|
|
}
|
|
break;
|
|
case MODE_RUNNING_TIME:
|
|
self->running_time_to_wait_for = GST_CLOCK_TIME_NONE;
|
|
self->running_time_to_end_at = GST_CLOCK_TIME_NONE;
|
|
if (self->target_running_time > self->last_seen_video_running_time
|
|
|| self->end_running_time >=
|
|
self->last_seen_video_running_time) {
|
|
self->dropping = TRUE;
|
|
}
|
|
break;
|
|
/* Let the chain functions handle the rest */
|
|
case MODE_VIDEO_FIRST:
|
|
/* pass-through */
|
|
default:
|
|
break;
|
|
}
|
|
}
|
|
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:{
|
|
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) {
|
|
gboolean emit_passthrough_signal = FALSE;
|
|
|
|
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) {
|
|
GST_INFO_OBJECT (self, "Target timecode reached at %" GST_TIME_FORMAT,
|
|
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 =
|
|
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) {
|
|
GST_INFO_OBJECT (self, "End timecode reached at %" GST_TIME_FORMAT,
|
|
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 = self->running_time_to_end_at;
|
|
self->must_send_end_message |= END_MESSAGE_STREAM_ENDED;
|
|
}
|
|
}
|
|
|
|
if (inbuf) {
|
|
gst_buffer_unref (inbuf);
|
|
inbuf = NULL;
|
|
}
|
|
} else 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
|
|
&& 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) {
|
|
/* 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;
|
|
}
|
|
if (self->recording) {
|
|
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_ARGS (self->end_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;
|
|
}
|
|
} else 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:{
|
|
if (self->running_time_to_wait_for == GST_CLOCK_TIME_NONE) {
|
|
self->running_time_to_wait_for = running_time;
|
|
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;
|
|
if (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);
|
|
}
|
|
|
|
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;
|
|
}
|