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gstreamer/gst/timecode/gstavwait.c
Stéphane Cerveau 891be51105 gst-plugins: allow per feature registration 
Split plugin into features including
dynamic types which can be indiviually
registered during a static build.

More details here:

https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2110>
2021-04-11 16:16:55 +00:00

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/*
* GStreamer
* Copyright (C) 2016 Vivia Nikolaidou <vivia@toolsonair.com>
*
* Based on gstvideoframe-audiolevel.c:
* Copyright (C) 2015 Vivia Nikolaidou <vivia@toolsonair.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* 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 <vivia@toolsonair.com>");
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);
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:{
const GstSegment *segment;
gboolean send_message = FALSE;
gboolean segment_changed;
g_mutex_lock (&self->mutex);
gst_event_parse_segment (event, &segment);
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:{
const GstSegment *segment;
gboolean segment_changed;
g_mutex_lock (&self->mutex);
gst_event_parse_segment (event, &segment);
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, &current_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;
}
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