| /* GStreamer |
| * Copyright (C) 2005-2007 Wim Taymans <wim.taymans@gmail.com> |
| * |
| * gstbasesink.c: Base class for sink elements |
| * |
| * 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., 59 Temple Place - Suite 330, |
| * Boston, MA 02111-1307, USA. |
| */ |
| |
| /** |
| * SECTION:gstbasesink |
| * @short_description: Base class for sink elements |
| * @see_also: #GstBaseTransform, #GstBaseSource |
| * |
| * #GstBaseSink is the base class for sink elements in GStreamer, such as |
| * xvimagesink or filesink. It is a layer on top of #GstElement that provides a |
| * simplified interface to plugin writers. #GstBaseSink handles many details |
| * for you, for example: preroll, clock synchronization, state changes, |
| * activation in push or pull mode, and queries. |
| * |
| * In most cases, when writing sink elements, there is no need to implement |
| * class methods from #GstElement or to set functions on pads, because the |
| * #GstBaseSink infrastructure should be sufficient. |
| * |
| * #GstBaseSink provides support for exactly one sink pad, which should be |
| * named "sink". A sink implementation (subclass of #GstBaseSink) should |
| * install a pad template in its base_init function, like so: |
| * <programlisting> |
| * static void |
| * my_element_base_init (gpointer g_class) |
| * { |
| * GstElementClass *gstelement_class = GST_ELEMENT_CLASS (g_class); |
| * |
| * // sinktemplate should be a #GstStaticPadTemplate with direction |
| * // #GST_PAD_SINK and name "sink" |
| * gst_element_class_add_pad_template (gstelement_class, |
| * gst_static_pad_template_get (&sinktemplate)); |
| * // see #GstElementDetails |
| * gst_element_class_set_details (gstelement_class, &details); |
| * } |
| * </programlisting> |
| * |
| * #GstBaseSink will handle the prerolling correctly. This means that it will |
| * return #GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first |
| * buffer arrives in this element. The base class will call the |
| * #GstBaseSink::preroll vmethod with this preroll buffer and will then commit |
| * the state change to the next asynchronously pending state. |
| * |
| * When the element is set to PLAYING, #GstBaseSink will synchronise on the |
| * clock using the times returned from ::get_times. If this function returns |
| * #GST_CLOCK_TIME_NONE for the start time, no synchronisation will be done. |
| * Synchronisation can be disabled entirely by setting the object "sync" |
| * property to %FALSE. |
| * |
| * After synchronisation the virtual method #GstBaseSink::render will be called. |
| * Subclasses should minimally implement this method. |
| * |
| * Since 0.10.3 subclasses that synchronise on the clock in the ::render method |
| * are supported as well. These classes typically receive a buffer in the render |
| * method and can then potentially block on the clock while rendering. A typical |
| * example is an audiosink. Since 0.10.11 these subclasses can use |
| * gst_base_sink_wait_preroll() to perform the blocking wait. |
| * |
| * Upon receiving the EOS event in the PLAYING state, #GstBaseSink will wait |
| * for the clock to reach the time indicated by the stop time of the last |
| * ::get_times call before posting an EOS message. When the element receives |
| * EOS in PAUSED, preroll completes, the event is queued and an EOS message is |
| * posted when going to PLAYING. |
| * |
| * #GstBaseSink will internally use the #GST_EVENT_NEWSEGMENT events to schedule |
| * synchronisation and clipping of buffers. Buffers that fall completely outside |
| * of the current segment are dropped. Buffers that fall partially in the |
| * segment are rendered (and prerolled). Subclasses should do any subbuffer |
| * clipping themselves when needed. |
| * |
| * #GstBaseSink will by default report the current playback position in |
| * #GST_FORMAT_TIME based on the current clock time and segment information. |
| * If no clock has been set on the element, the query will be forwarded |
| * upstream. |
| * |
| * The ::set_caps function will be called when the subclass should configure |
| * itself to process a specific media type. |
| * |
| * The ::start and ::stop virtual methods will be called when resources should |
| * be allocated. Any ::preroll, ::render and ::set_caps function will be |
| * called between the ::start and ::stop calls. |
| * |
| * The ::event virtual method will be called when an event is received by |
| * #GstBaseSink. Normally this method should only be overriden by very specific |
| * elements (such as file sinks) which need to handle the newsegment event |
| * specially. |
| * |
| * #GstBaseSink provides an overridable ::buffer_alloc function that can be |
| * used by sinks that want to do reverse negotiation or to provide |
| * custom buffers (hardware buffers for example) to upstream elements. |
| * |
| * The ::unlock method is called when the elements should unblock any blocking |
| * operations they perform in the ::render method. This is mostly useful when |
| * the ::render method performs a blocking write on a file descriptor, for |
| * example. |
| * |
| * The max-lateness property affects how the sink deals with buffers that |
| * arrive too late in the sink. A buffer arrives too late in the sink when |
| * the presentation time (as a combination of the last segment, buffer |
| * timestamp and element base_time) plus the duration is before the current |
| * time of the clock. |
| * If the frame is later than max-lateness, the sink will drop the buffer |
| * without calling the render method. |
| * This feature is disabled if sync is disabled, the ::get-times method does |
| * not return a valid start time or max-lateness is set to -1 (the default). |
| * Subclasses can use gst_base_sink_set_max_lateness() to configure the |
| * max-lateness value. |
| * |
| * The qos property will enable the quality-of-service features of the basesink |
| * which gather statistics about the real-time performance of the clock |
| * synchronisation. For each buffer received in the sink, statistics are |
| * gathered and a QOS event is sent upstream with these numbers. This |
| * information can then be used by upstream elements to reduce their processing |
| * rate, for example. |
| * |
| * Since 0.10.15 the async property can be used to instruct the sink to never |
| * perform an ASYNC state change. This feature is mostly usable when dealing |
| * with non-synchronized streams or sparse streams. |
| * |
| * Last reviewed on 2007-08-29 (0.10.15) |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include "config.h" |
| #endif |
| |
| #include "gstbasesink.h" |
| #include <gst/gstmarshal.h> |
| #include <gst/gst_private.h> |
| #include <gst/gst-i18n-lib.h> |
| |
| GST_DEBUG_CATEGORY_STATIC (gst_base_sink_debug); |
| #define GST_CAT_DEFAULT gst_base_sink_debug |
| |
| #define GST_BASE_SINK_GET_PRIVATE(obj) \ |
| (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_SINK, GstBaseSinkPrivate)) |
| |
| #define GST_FLOW_STEP GST_FLOW_CUSTOM_ERROR |
| |
| typedef struct |
| { |
| gboolean valid; /* if this info is valid */ |
| guint32 seqnum; /* the seqnum of the STEP event */ |
| GstFormat format; /* the format of the amount */ |
| guint64 amount; /* the total amount of data to skip */ |
| guint64 position; /* the position in the stepped data */ |
| guint64 duration; /* the duration in time of the skipped data */ |
| guint64 start; /* running_time of the start */ |
| gdouble rate; /* rate of skipping */ |
| gdouble start_rate; /* rate before skipping */ |
| guint64 start_start; /* start position skipping */ |
| guint64 start_stop; /* stop position skipping */ |
| gboolean flush; /* if this was a flushing step */ |
| gboolean intermediate; /* if this is an intermediate step */ |
| gboolean need_preroll; /* if we need preroll after this step */ |
| } GstStepInfo; |
| |
| /* FIXME, some stuff in ABI.data and other in Private... |
| * Make up your mind please. |
| */ |
| struct _GstBaseSinkPrivate |
| { |
| gint qos_enabled; /* ATOMIC */ |
| gboolean async_enabled; |
| GstClockTimeDiff ts_offset; |
| GstClockTime render_delay; |
| |
| /* start, stop of current buffer, stream time, used to report position */ |
| GstClockTime current_sstart; |
| GstClockTime current_sstop; |
| |
| /* start, stop and jitter of current buffer, running time */ |
| GstClockTime current_rstart; |
| GstClockTime current_rstop; |
| GstClockTimeDiff current_jitter; |
| |
| /* EOS sync time in running time */ |
| GstClockTime eos_rtime; |
| |
| /* last buffer that arrived in time, running time */ |
| GstClockTime last_in_time; |
| /* when the last buffer left the sink, running time */ |
| GstClockTime last_left; |
| |
| /* running averages go here these are done on running time */ |
| GstClockTime avg_pt; |
| GstClockTime avg_duration; |
| gdouble avg_rate; |
| |
| /* these are done on system time. avg_jitter and avg_render are |
| * compared to eachother to see if the rendering time takes a |
| * huge amount of the processing, If so we are flooded with |
| * buffers. */ |
| GstClockTime last_left_systime; |
| GstClockTime avg_jitter; |
| GstClockTime start, stop; |
| GstClockTime avg_render; |
| |
| /* number of rendered and dropped frames */ |
| guint64 rendered; |
| guint64 dropped; |
| |
| /* latency stuff */ |
| GstClockTime latency; |
| |
| /* if we already commited the state */ |
| gboolean commited; |
| |
| /* when we received EOS */ |
| gboolean received_eos; |
| |
| /* when we are prerolled and able to report latency */ |
| gboolean have_latency; |
| |
| /* the last buffer we prerolled or rendered. Useful for making snapshots */ |
| GstBuffer *last_buffer; |
| |
| /* caps for pull based scheduling */ |
| GstCaps *pull_caps; |
| |
| /* blocksize for pulling */ |
| guint blocksize; |
| |
| gboolean discont; |
| |
| /* seqnum of the stream */ |
| guint32 seqnum; |
| |
| gboolean call_preroll; |
| gboolean step_unlock; |
| |
| /* we have a pending and a current step operation */ |
| GstStepInfo current_step; |
| GstStepInfo pending_step; |
| }; |
| |
| #define DO_RUNNING_AVG(avg,val,size) (((val) + ((size)-1) * (avg)) / (size)) |
| |
| /* generic running average, this has a neutral window size */ |
| #define UPDATE_RUNNING_AVG(avg,val) DO_RUNNING_AVG(avg,val,8) |
| |
| /* the windows for these running averages are experimentally obtained. |
| * possitive values get averaged more while negative values use a small |
| * window so we can react faster to badness. */ |
| #define UPDATE_RUNNING_AVG_P(avg,val) DO_RUNNING_AVG(avg,val,16) |
| #define UPDATE_RUNNING_AVG_N(avg,val) DO_RUNNING_AVG(avg,val,4) |
| |
| /* BaseSink properties */ |
| |
| #define DEFAULT_CAN_ACTIVATE_PULL FALSE /* fixme: enable me */ |
| #define DEFAULT_CAN_ACTIVATE_PUSH TRUE |
| |
| #define DEFAULT_PREROLL_QUEUE_LEN 0 |
| #define DEFAULT_SYNC TRUE |
| #define DEFAULT_MAX_LATENESS -1 |
| #define DEFAULT_QOS FALSE |
| #define DEFAULT_ASYNC TRUE |
| #define DEFAULT_TS_OFFSET 0 |
| #define DEFAULT_BLOCKSIZE 4096 |
| #define DEFAULT_RENDER_DELAY 0 |
| |
| enum |
| { |
| PROP_0, |
| PROP_PREROLL_QUEUE_LEN, |
| PROP_SYNC, |
| PROP_MAX_LATENESS, |
| PROP_QOS, |
| PROP_ASYNC, |
| PROP_TS_OFFSET, |
| PROP_LAST_BUFFER, |
| PROP_BLOCKSIZE, |
| PROP_RENDER_DELAY, |
| PROP_LAST |
| }; |
| |
| static GstElementClass *parent_class = NULL; |
| |
| static void gst_base_sink_class_init (GstBaseSinkClass * klass); |
| static void gst_base_sink_init (GstBaseSink * trans, gpointer g_class); |
| static void gst_base_sink_finalize (GObject * object); |
| |
| GType |
| gst_base_sink_get_type (void) |
| { |
| static volatile gsize base_sink_type = 0; |
| |
| if (g_once_init_enter (&base_sink_type)) { |
| GType _type; |
| static const GTypeInfo base_sink_info = { |
| sizeof (GstBaseSinkClass), |
| NULL, |
| NULL, |
| (GClassInitFunc) gst_base_sink_class_init, |
| NULL, |
| NULL, |
| sizeof (GstBaseSink), |
| 0, |
| (GInstanceInitFunc) gst_base_sink_init, |
| }; |
| |
| _type = g_type_register_static (GST_TYPE_ELEMENT, |
| "GstBaseSink", &base_sink_info, G_TYPE_FLAG_ABSTRACT); |
| g_once_init_leave (&base_sink_type, _type); |
| } |
| return base_sink_type; |
| } |
| |
| static void gst_base_sink_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec); |
| static void gst_base_sink_get_property (GObject * object, guint prop_id, |
| GValue * value, GParamSpec * pspec); |
| |
| static gboolean gst_base_sink_send_event (GstElement * element, |
| GstEvent * event); |
| static gboolean gst_base_sink_query (GstElement * element, GstQuery * query); |
| |
| static GstCaps *gst_base_sink_get_caps (GstBaseSink * sink); |
| static gboolean gst_base_sink_set_caps (GstBaseSink * sink, GstCaps * caps); |
| static GstFlowReturn gst_base_sink_buffer_alloc (GstBaseSink * sink, |
| guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf); |
| static void gst_base_sink_get_times (GstBaseSink * basesink, GstBuffer * buffer, |
| GstClockTime * start, GstClockTime * end); |
| static gboolean gst_base_sink_set_flushing (GstBaseSink * basesink, |
| GstPad * pad, gboolean flushing); |
| static gboolean gst_base_sink_default_activate_pull (GstBaseSink * basesink, |
| gboolean active); |
| static gboolean gst_base_sink_default_do_seek (GstBaseSink * sink, |
| GstSegment * segment); |
| static gboolean gst_base_sink_default_prepare_seek_segment (GstBaseSink * sink, |
| GstEvent * event, GstSegment * segment); |
| |
| static GstStateChangeReturn gst_base_sink_change_state (GstElement * element, |
| GstStateChange transition); |
| |
| static GstFlowReturn gst_base_sink_chain (GstPad * pad, GstBuffer * buffer); |
| static GstFlowReturn gst_base_sink_chain_list (GstPad * pad, |
| GstBufferList * list); |
| |
| static void gst_base_sink_loop (GstPad * pad); |
| static gboolean gst_base_sink_pad_activate (GstPad * pad); |
| static gboolean gst_base_sink_pad_activate_push (GstPad * pad, gboolean active); |
| static gboolean gst_base_sink_pad_activate_pull (GstPad * pad, gboolean active); |
| static gboolean gst_base_sink_event (GstPad * pad, GstEvent * event); |
| static gboolean gst_base_sink_peer_query (GstBaseSink * sink, GstQuery * query); |
| |
| static gboolean gst_base_sink_negotiate_pull (GstBaseSink * basesink); |
| |
| /* check if an object was too late */ |
| static gboolean gst_base_sink_is_too_late (GstBaseSink * basesink, |
| GstMiniObject * obj, GstClockTime start, GstClockTime stop, |
| GstClockReturn status, GstClockTimeDiff jitter); |
| static GstFlowReturn gst_base_sink_preroll_object (GstBaseSink * basesink, |
| gboolean is_list, GstMiniObject * obj); |
| |
| static void |
| gst_base_sink_class_init (GstBaseSinkClass * klass) |
| { |
| GObjectClass *gobject_class; |
| GstElementClass *gstelement_class; |
| |
| gobject_class = G_OBJECT_CLASS (klass); |
| gstelement_class = GST_ELEMENT_CLASS (klass); |
| |
| GST_DEBUG_CATEGORY_INIT (gst_base_sink_debug, "basesink", 0, |
| "basesink element"); |
| |
| g_type_class_add_private (klass, sizeof (GstBaseSinkPrivate)); |
| |
| parent_class = g_type_class_peek_parent (klass); |
| |
| gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_base_sink_finalize); |
| gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_base_sink_set_property); |
| gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_base_sink_get_property); |
| |
| /* FIXME, this next value should be configured using an event from the |
| * upstream element, ie, the BUFFER_SIZE event. */ |
| g_object_class_install_property (gobject_class, PROP_PREROLL_QUEUE_LEN, |
| g_param_spec_uint ("preroll-queue-len", "Preroll queue length", |
| "Number of buffers to queue during preroll", 0, G_MAXUINT, |
| DEFAULT_PREROLL_QUEUE_LEN, |
| G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS)); |
| |
| g_object_class_install_property (gobject_class, PROP_SYNC, |
| g_param_spec_boolean ("sync", "Sync", "Sync on the clock", DEFAULT_SYNC, |
| G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| g_object_class_install_property (gobject_class, PROP_MAX_LATENESS, |
| g_param_spec_int64 ("max-lateness", "Max Lateness", |
| "Maximum number of nanoseconds that a buffer can be late before it " |
| "is dropped (-1 unlimited)", -1, G_MAXINT64, DEFAULT_MAX_LATENESS, |
| G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| g_object_class_install_property (gobject_class, PROP_QOS, |
| g_param_spec_boolean ("qos", "Qos", |
| "Generate Quality-of-Service events upstream", DEFAULT_QOS, |
| G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| /** |
| * GstBaseSink:async |
| * |
| * If set to #TRUE, the basesink will perform asynchronous state changes. |
| * When set to #FALSE, the sink will not signal the parent when it prerolls. |
| * Use this option when dealing with sparse streams or when synchronisation is |
| * not required. |
| * |
| * Since: 0.10.15 |
| */ |
| g_object_class_install_property (gobject_class, PROP_ASYNC, |
| g_param_spec_boolean ("async", "Async", |
| "Go asynchronously to PAUSED", DEFAULT_ASYNC, |
| G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| /** |
| * GstBaseSink:ts-offset |
| * |
| * Controls the final synchronisation, a negative value will render the buffer |
| * earlier while a positive value delays playback. This property can be |
| * used to fix synchronisation in bad files. |
| * |
| * Since: 0.10.15 |
| */ |
| g_object_class_install_property (gobject_class, PROP_TS_OFFSET, |
| g_param_spec_int64 ("ts-offset", "TS Offset", |
| "Timestamp offset in nanoseconds", G_MININT64, G_MAXINT64, |
| DEFAULT_TS_OFFSET, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| /** |
| * GstBaseSink:last-buffer |
| * |
| * The last buffer that arrived in the sink and was used for preroll or for |
| * rendering. This property can be used to generate thumbnails. This property |
| * can be NULL when the sink has not yet received a bufer. |
| * |
| * Since: 0.10.15 |
| */ |
| g_object_class_install_property (gobject_class, PROP_LAST_BUFFER, |
| gst_param_spec_mini_object ("last-buffer", "Last Buffer", |
| "The last buffer received in the sink", GST_TYPE_BUFFER, |
| G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); |
| /** |
| * GstBaseSink:blocksize |
| * |
| * The amount of bytes to pull when operating in pull mode. |
| * |
| * Since: 0.10.22 |
| */ |
| g_object_class_install_property (gobject_class, PROP_BLOCKSIZE, |
| g_param_spec_uint ("blocksize", "Block size", |
| "Size in bytes to pull per buffer (0 = default)", 0, G_MAXUINT, |
| DEFAULT_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| /** |
| * GstBaseSink:render-delay |
| * |
| * The additional delay between synchronisation and actual rendering of the |
| * media. This property will add additional latency to the device in order to |
| * make other sinks compensate for the delay. |
| * |
| * Since: 0.10.22 |
| */ |
| g_object_class_install_property (gobject_class, PROP_RENDER_DELAY, |
| g_param_spec_uint64 ("render-delay", "Render Delay", |
| "Additional render delay of the sink in nanoseconds", 0, G_MAXUINT64, |
| DEFAULT_RENDER_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); |
| |
| gstelement_class->change_state = |
| GST_DEBUG_FUNCPTR (gst_base_sink_change_state); |
| gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_base_sink_send_event); |
| gstelement_class->query = GST_DEBUG_FUNCPTR (gst_base_sink_query); |
| |
| klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_sink_get_caps); |
| klass->set_caps = GST_DEBUG_FUNCPTR (gst_base_sink_set_caps); |
| klass->buffer_alloc = GST_DEBUG_FUNCPTR (gst_base_sink_buffer_alloc); |
| klass->get_times = GST_DEBUG_FUNCPTR (gst_base_sink_get_times); |
| klass->activate_pull = |
| GST_DEBUG_FUNCPTR (gst_base_sink_default_activate_pull); |
| } |
| |
| static GstCaps * |
| gst_base_sink_pad_getcaps (GstPad * pad) |
| { |
| GstBaseSinkClass *bclass; |
| GstBaseSink *bsink; |
| GstCaps *caps = NULL; |
| |
| bsink = GST_BASE_SINK (gst_pad_get_parent (pad)); |
| bclass = GST_BASE_SINK_GET_CLASS (bsink); |
| |
| if (bsink->pad_mode == GST_ACTIVATE_PULL) { |
| /* if we are operating in pull mode we only accept the negotiated caps */ |
| GST_OBJECT_LOCK (pad); |
| if ((caps = GST_PAD_CAPS (pad))) |
| gst_caps_ref (caps); |
| GST_OBJECT_UNLOCK (pad); |
| } |
| if (caps == NULL) { |
| if (bclass->get_caps) |
| caps = bclass->get_caps (bsink); |
| |
| if (caps == NULL) { |
| GstPadTemplate *pad_template; |
| |
| pad_template = |
| gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), |
| "sink"); |
| if (pad_template != NULL) { |
| caps = gst_caps_ref (gst_pad_template_get_caps (pad_template)); |
| } |
| } |
| } |
| gst_object_unref (bsink); |
| |
| return caps; |
| } |
| |
| static gboolean |
| gst_base_sink_pad_setcaps (GstPad * pad, GstCaps * caps) |
| { |
| GstBaseSinkClass *bclass; |
| GstBaseSink *bsink; |
| gboolean res = TRUE; |
| |
| bsink = GST_BASE_SINK (gst_pad_get_parent (pad)); |
| bclass = GST_BASE_SINK_GET_CLASS (bsink); |
| |
| if (res && bclass->set_caps) |
| res = bclass->set_caps (bsink, caps); |
| |
| gst_object_unref (bsink); |
| |
| return res; |
| } |
| |
| static void |
| gst_base_sink_pad_fixate (GstPad * pad, GstCaps * caps) |
| { |
| GstBaseSinkClass *bclass; |
| GstBaseSink *bsink; |
| |
| bsink = GST_BASE_SINK (gst_pad_get_parent (pad)); |
| bclass = GST_BASE_SINK_GET_CLASS (bsink); |
| |
| if (bclass->fixate) |
| bclass->fixate (bsink, caps); |
| |
| gst_object_unref (bsink); |
| } |
| |
| static GstFlowReturn |
| gst_base_sink_pad_buffer_alloc (GstPad * pad, guint64 offset, guint size, |
| GstCaps * caps, GstBuffer ** buf) |
| { |
| GstBaseSinkClass *bclass; |
| GstBaseSink *bsink; |
| GstFlowReturn result = GST_FLOW_OK; |
| |
| bsink = GST_BASE_SINK (gst_pad_get_parent (pad)); |
| bclass = GST_BASE_SINK_GET_CLASS (bsink); |
| |
| if (bclass->buffer_alloc) |
| result = bclass->buffer_alloc (bsink, offset, size, caps, buf); |
| else |
| *buf = NULL; /* fallback in gstpad.c will allocate generic buffer */ |
| |
| gst_object_unref (bsink); |
| |
| return result; |
| } |
| |
| static void |
| gst_base_sink_init (GstBaseSink * basesink, gpointer g_class) |
| { |
| GstPadTemplate *pad_template; |
| GstBaseSinkPrivate *priv; |
| |
| basesink->priv = priv = GST_BASE_SINK_GET_PRIVATE (basesink); |
| |
| pad_template = |
| gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "sink"); |
| g_return_if_fail (pad_template != NULL); |
| |
| basesink->sinkpad = gst_pad_new_from_template (pad_template, "sink"); |
| |
| gst_pad_set_getcaps_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_pad_getcaps)); |
| gst_pad_set_setcaps_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_pad_setcaps)); |
| gst_pad_set_fixatecaps_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_pad_fixate)); |
| gst_pad_set_bufferalloc_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_pad_buffer_alloc)); |
| gst_pad_set_activate_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate)); |
| gst_pad_set_activatepush_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate_push)); |
| gst_pad_set_activatepull_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_pad_activate_pull)); |
| gst_pad_set_event_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_event)); |
| gst_pad_set_chain_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_chain)); |
| gst_pad_set_chain_list_function (basesink->sinkpad, |
| GST_DEBUG_FUNCPTR (gst_base_sink_chain_list)); |
| gst_element_add_pad (GST_ELEMENT_CAST (basesink), basesink->sinkpad); |
| |
| basesink->pad_mode = GST_ACTIVATE_NONE; |
| basesink->preroll_queue = g_queue_new (); |
| basesink->abidata.ABI.clip_segment = gst_segment_new (); |
| priv->have_latency = FALSE; |
| |
| basesink->can_activate_push = DEFAULT_CAN_ACTIVATE_PUSH; |
| basesink->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL; |
| |
| basesink->sync = DEFAULT_SYNC; |
| basesink->abidata.ABI.max_lateness = DEFAULT_MAX_LATENESS; |
| g_atomic_int_set (&priv->qos_enabled, DEFAULT_QOS); |
| priv->async_enabled = DEFAULT_ASYNC; |
| priv->ts_offset = DEFAULT_TS_OFFSET; |
| priv->render_delay = DEFAULT_RENDER_DELAY; |
| priv->blocksize = DEFAULT_BLOCKSIZE; |
| |
| GST_OBJECT_FLAG_SET (basesink, GST_ELEMENT_IS_SINK); |
| } |
| |
| static void |
| gst_base_sink_finalize (GObject * object) |
| { |
| GstBaseSink *basesink; |
| |
| basesink = GST_BASE_SINK (object); |
| |
| g_queue_free (basesink->preroll_queue); |
| gst_segment_free (basesink->abidata.ABI.clip_segment); |
| |
| G_OBJECT_CLASS (parent_class)->finalize (object); |
| } |
| |
| /** |
| * gst_base_sink_set_sync: |
| * @sink: the sink |
| * @sync: the new sync value. |
| * |
| * Configures @sink to synchronize on the clock or not. When |
| * @sync is FALSE, incomming samples will be played as fast as |
| * possible. If @sync is TRUE, the timestamps of the incomming |
| * buffers will be used to schedule the exact render time of its |
| * contents. |
| * |
| * Since: 0.10.4 |
| */ |
| void |
| gst_base_sink_set_sync (GstBaseSink * sink, gboolean sync) |
| { |
| g_return_if_fail (GST_IS_BASE_SINK (sink)); |
| |
| GST_OBJECT_LOCK (sink); |
| sink->sync = sync; |
| GST_OBJECT_UNLOCK (sink); |
| } |
| |
| /** |
| * gst_base_sink_get_sync: |
| * @sink: the sink |
| * |
| * Checks if @sink is currently configured to synchronize against the |
| * clock. |
| * |
| * Returns: TRUE if the sink is configured to synchronize against the clock. |
| * |
| * Since: 0.10.4 |
| */ |
| gboolean |
| gst_base_sink_get_sync (GstBaseSink * sink) |
| { |
| gboolean res; |
| |
| g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE); |
| |
| GST_OBJECT_LOCK (sink); |
| res = sink->sync; |
| GST_OBJECT_UNLOCK (sink); |
| |
| return res; |
| } |
| |
| /** |
| * gst_base_sink_set_max_lateness: |
| * @sink: the sink |
| * @max_lateness: the new max lateness value. |
| * |
| * Sets the new max lateness value to @max_lateness. This value is |
| * used to decide if a buffer should be dropped or not based on the |
| * buffer timestamp and the current clock time. A value of -1 means |
| * an unlimited time. |
| * |
| * Since: 0.10.4 |
| */ |
| void |
| gst_base_sink_set_max_lateness (GstBaseSink * sink, gint64 max_lateness) |
| { |
| g_return_if_fail (GST_IS_BASE_SINK (sink)); |
| |
| GST_OBJECT_LOCK (sink); |
| sink->abidata.ABI.max_lateness = max_lateness; |
| GST_OBJECT_UNLOCK (sink); |
| } |
| |
| /** |
| * gst_base_sink_get_max_lateness: |
| * @sink: the sink |
| * |
| * Gets the max lateness value. See gst_base_sink_set_max_lateness for |
| * more details. |
| * |
| * Returns: The maximum time in nanoseconds that a buffer can be late |
| * before it is dropped and not rendered. A value of -1 means an |
| * unlimited time. |
| * |
| * Since: 0.10.4 |
| */ |
| gint64 |
| gst_base_sink_get_max_lateness (GstBaseSink * sink) |
| { |
| gint64 res; |
| |
| g_return_val_if_fail (GST_IS_BASE_SINK (sink), -1); |
| |
| GST_OBJECT_LOCK (sink); |
| res = sink->abidata.ABI.max_lateness; |
| GST_OBJECT_UNLOCK (sink); |
| |
| return res; |
| } |
| |
| /** |
| * gst_base_sink_set_qos_enabled: |
| * @sink: the sink |
| * @enabled: the new qos value. |
| * |
| * Configures @sink to send Quality-of-Service events upstream. |
| * |
| * Since: 0.10.5 |
| */ |
| void |
| gst_base_sink_set_qos_enabled (GstBaseSink * sink, gboolean enabled) |
| { |
| g_return_if_fail (GST_IS_BASE_SINK (sink)); |
| |
| g_atomic_int_set (&sink->priv->qos_enabled, enabled); |
| } |
| |
| /** |
| * gst_base_sink_is_qos_enabled: |
| * @sink: the sink |
| * |
| * Checks if @sink is currently configured to send Quality-of-Service events |
| * upstream. |
| * |
| * Returns: TRUE if the sink is configured to perform Quality-of-Service. |
| * |
| * Since: 0.10.5 |
| */ |
| gboolean |
| gst_base_sink_is_qos_enabled (GstBaseSink * sink) |
| { |
| gboolean res; |
| |
| g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE); |
| |
| res = g_atomic_int_get (&sink->priv->qos_enabled); |
| |
| return res; |
| } |
| |
| /** |
| * gst_base_sink_set_async_enabled: |
| * @sink: the sink |
| * @enabled: the new async value. |
| * |
| * Configures @sink to perform all state changes asynchronusly. When async is |
| * disabled, the sink will immediatly go to PAUSED instead of waiting for a |
| * preroll buffer. This feature is usefull if the sink does not synchronize |
| * against the clock or when it is dealing with sparse streams. |
| * |
| * Since: 0.10.15 |
| */ |
| void |
| gst_base_sink_set_async_enabled (GstBaseSink * sink, gboolean enabled) |
| { |
| g_return_if_fail (GST_IS_BASE_SINK (sink)); |
| |
| GST_PAD_PREROLL_LOCK (sink->sinkpad); |
| sink->priv->async_enabled = enabled; |
| GST_LOG_OBJECT (sink, "set async enabled to %d", enabled); |
| GST_PAD_PREROLL_UNLOCK (sink->sinkpad); |
| } |
| |
| /** |
| * gst_base_sink_is_async_enabled: |
| * @sink: the sink |
| * |
| * Checks if @sink is currently configured to perform asynchronous state |
| * changes to PAUSED. |
| * |
| * Returns: TRUE if the sink is configured to perform asynchronous state |
| * changes. |
| * |
| * Since: 0.10.15 |
| */ |
| gboolean |
| gst_base_sink_is_async_enabled (GstBaseSink * sink) |
| { |
| gboolean res; |
| |
| g_return_val_if_fail (GST_IS_BASE_SINK (sink), FALSE); |
| |
| GST_PAD_PREROLL_LOCK (sink->sinkpad); |
| res = sink->priv->async_enabled; |
| GST_PAD_PREROLL_UNLOCK (sink->sinkpad); |
| |
| return res; |
| } |
| |
| /** |
| * gst_base_sink_set_ts_offset: |
| * @sink: the sink |
| * @offset: the new offset |
| * |
| * Adjust the synchronisation of @sink with @offset. A negative value will |
| * render buffers earlier than their timestamp. A positive value will delay |
| * rendering. This function can be used to fix playback of badly timestamped |
| * buffers. |
| * |
| * Since: 0.10.15 |
| */ |
| void |
| gst_base_sink_set_ts_offset (GstBaseSink * sink, GstClockTimeDiff offset) |
| { |
| g_return_if_fail (GST_IS_BASE_SINK (sink)); |
| |
| GST_OBJECT_LOCK (sink); |
| sink->priv->ts_offset = offset; |
| GST_LOG_OBJECT (sink, "set time offset to %" G_GINT64_FORMAT, offset); |
| GST_OBJECT_UNLOCK (sink); |
| } |
| |
| /** |
| * gst_base_sink_get_ts_offset: |
| * @sink: the sink |
| * |
| * Get the synchronisation offset of @sink. |
| * |
| * Returns: The synchronisation offset. |
| * |
| * Since: 0.10.15 |
| */ |
| GstClockTimeDiff |
| gst_base_sink_get_ts_offset (GstBaseSink * sink) |
| { |
| GstClockTimeDiff res; |
| |
| g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0); |
| |
| GST_OBJECT_LOCK (sink); |
| res = sink->priv->ts_offset; |
| GST_OBJECT_UNLOCK (sink); |
| |
| return res; |
| } |
| |
| /** |
| * gst_base_sink_get_last_buffer: |
| * @sink: the sink |
| * |
| * Get the last buffer that arrived in the sink and was used for preroll or for |
| * rendering. This property can be used to generate thumbnails. |
| * |
| * The #GstCaps on the buffer can be used to determine the type of the buffer. |
| * |
| * Returns: a #GstBuffer. gst_buffer_unref() after usage. This function returns |
| * NULL when no buffer has arrived in the sink yet or when the sink is not in |
| * PAUSED or PLAYING. |
| * |
| * Since: 0.10.15 |
| */ |
| GstBuffer * |
| gst_base_sink_get_last_buffer (GstBaseSink * sink) |
| { |
| GstBuffer *res; |
| |
| g_return_val_if_fail (GST_IS_BASE_SINK (sink), NULL); |
| |
| GST_OBJECT_LOCK (sink); |
| if ((res = sink->priv->last_buffer)) |
| gst_buffer_ref (res); |
| GST_OBJECT_UNLOCK (sink); |
| |
| return res; |
| } |
| |
| static void |
| gst_base_sink_set_last_buffer (GstBaseSink * sink, GstBuffer * buffer) |
| { |
| GstBuffer *old; |
| |
| GST_OBJECT_LOCK (sink); |
| old = sink->priv->last_buffer; |
| if (G_LIKELY (old != buffer)) { |
| GST_DEBUG_OBJECT (sink, "setting last buffer to %p", buffer); |
| if (G_LIKELY (buffer)) |
| gst_buffer_ref (buffer); |
| sink->priv->last_buffer = buffer; |
| } else { |
| old = NULL; |
| } |
| GST_OBJECT_UNLOCK (sink); |
| |
| /* avoid unreffing with the lock because cleanup code might want to take the |
| * lock too */ |
| if (G_LIKELY (old)) |
| gst_buffer_unref (old); |
| } |
| |
| /** |
| * gst_base_sink_get_latency: |
| * @sink: the sink |
| * |
| * Get the currently configured latency. |
| * |
| * Returns: The configured latency. |
| * |
| * Since: 0.10.12 |
| */ |
| GstClockTime |
| gst_base_sink_get_latency (GstBaseSink * sink) |
| { |
| GstClockTime res; |
| |
| GST_OBJECT_LOCK (sink); |
| res = sink->priv->latency; |
| GST_OBJECT_UNLOCK (sink); |
| |
| return res; |
| } |
| |
| /** |
| * gst_base_sink_query_latency: |
| * @sink: the sink |
| * @live: if the sink is live |
| * @upstream_live: if an upstream element is live |
| * @min_latency: the min latency of the upstream elements |
| * @max_latency: the max latency of the upstream elements |
| * |
| * Query the sink for the latency parameters. The latency will be queried from |
| * the upstream elements. @live will be TRUE if @sink is configured to |
| * synchronize against the clock. @upstream_live will be TRUE if an upstream |
| * element is live. |
| * |
| * If both @live and @upstream_live are TRUE, the sink will want to compensate |
| * for the latency introduced by the upstream elements by setting the |
| * @min_latency to a strictly possitive value. |
| * |
| * This function is mostly used by subclasses. |
| * |
| * Returns: TRUE if the query succeeded. |
| * |
| * Since: 0.10.12 |
| */ |
| gboolean |
| gst_base_sink_query_latency (GstBaseSink * sink, gboolean * live, |
| gboolean * upstream_live, GstClockTime * min_latency, |
| GstClockTime * max_latency) |
| { |
| gboolean l, us_live, res, have_latency; |
| GstClockTime min, max, render_delay; |
| GstQuery *query; |
| GstClockTime us_min, us_max; |
| |
| /* we are live when we sync to the clock */ |
| GST_OBJECT_LOCK (sink); |
| l = sink->sync; |
| have_latency = sink->priv->have_latency; |
| render_delay = sink->priv->render_delay; |
| GST_OBJECT_UNLOCK (sink); |
| |
| /* assume no latency */ |
| min = 0; |
| max = -1; |
| us_live = FALSE; |
| |
| if (have_latency) { |
| GST_DEBUG_OBJECT (sink, "we are ready for LATENCY query"); |
| /* we are ready for a latency query this is when we preroll or when we are |
| * not async. */ |
| query = gst_query_new_latency (); |
| |
| /* ask the peer for the latency */ |
| if ((res = gst_base_sink_peer_query (sink, query))) { |
| /* get upstream min and max latency */ |
| gst_query_parse_latency (query, &us_live, &us_min, &us_max); |
| |
| if (us_live) { |
| /* upstream live, use its latency, subclasses should use these |
| * values to create the complete latency. */ |
| min = us_min; |
| max = us_max; |
| } |
| if (l) { |
| /* we need to add the render delay if we are live */ |
| if (min != -1) |
| min += render_delay; |
| if (max != -1) |
| max += render_delay; |
| } |
| } |
| gst_query_unref (query); |
| } else { |
| GST_DEBUG_OBJECT (sink, "we are not yet ready for LATENCY query"); |
| res = FALSE; |
| } |
| |
| /* not live, we tried to do the query, if it failed we return TRUE anyway */ |
| if (!res) { |
| if (!l) { |
| res = TRUE; |
| GST_DEBUG_OBJECT (sink, "latency query failed but we are not live"); |
| } else { |
| GST_DEBUG_OBJECT (sink, "latency query failed and we are live"); |
| } |
| } |
| |
| if (res) { |
| GST_DEBUG_OBJECT (sink, "latency query: live: %d, have_latency %d," |
| " upstream: %d, min %" GST_TIME_FORMAT ", max %" GST_TIME_FORMAT, l, |
| have_latency, us_live, GST_TIME_ARGS (min), GST_TIME_ARGS (max)); |
| |
| if (live) |
| *live = l; |
| if (upstream_live) |
| *upstream_live = us_live; |
| if (min_latency) |
| *min_latency = min; |
| if (max_latency) |
| *max_latency = max; |
| } |
| return res; |
| } |
| |
| /** |
| * gst_base_sink_set_render_delay: |
| * @sink: a #GstBaseSink |
| * @delay: the new delay |
| * |
| * Set the render delay in @sink to @delay. The render delay is the time |
| * between actual rendering of a buffer and its synchronisation time. Some |
| * devices might delay media rendering which can be compensated for with this |
| * function. |
| * |
| * After calling this function, this sink will report additional latency and |
| * other sinks will adjust their latency to delay the rendering of their media. |
| * |
| * This function is usually called by subclasses. |
| * |
| * Since: 0.10.21 |
| */ |
| void |
| gst_base_sink_set_render_delay (GstBaseSink * sink, GstClockTime delay) |
| { |
| GstClockTime old_render_delay; |
| |
| g_return_if_fail (GST_IS_BASE_SINK (sink)); |
| |
| GST_OBJECT_LOCK (sink); |
| old_render_delay = sink->priv->render_delay; |
| sink->priv->render_delay = delay; |
| GST_LOG_OBJECT (sink, "set render delay to %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (delay)); |
| GST_OBJECT_UNLOCK (sink); |
| |
| if (delay != old_render_delay) { |
| GST_DEBUG_OBJECT (sink, "posting latency changed"); |
| gst_element_post_message (GST_ELEMENT_CAST (sink), |
| gst_message_new_latency (GST_OBJECT_CAST (sink))); |
| } |
| } |
| |
| /** |
| * gst_base_sink_get_render_delay: |
| * @sink: a #GstBaseSink |
| * |
| * Get the render delay of @sink. see gst_base_sink_set_render_delay() for more |
| * information about the render delay. |
| * |
| * Returns: the render delay of @sink. |
| * |
| * Since: 0.10.21 |
| */ |
| GstClockTime |
| gst_base_sink_get_render_delay (GstBaseSink * sink) |
| { |
| GstClockTimeDiff res; |
| |
| g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0); |
| |
| GST_OBJECT_LOCK (sink); |
| res = sink->priv->render_delay; |
| GST_OBJECT_UNLOCK (sink); |
| |
| return res; |
| } |
| |
| /** |
| * gst_base_sink_set_blocksize: |
| * @sink: a #GstBaseSink |
| * @blocksize: the blocksize in bytes |
| * |
| * Set the number of bytes that the sink will pull when it is operating in pull |
| * mode. |
| * |
| * Since: 0.10.22 |
| */ |
| void |
| gst_base_sink_set_blocksize (GstBaseSink * sink, guint blocksize) |
| { |
| g_return_if_fail (GST_IS_BASE_SINK (sink)); |
| |
| GST_OBJECT_LOCK (sink); |
| sink->priv->blocksize = blocksize; |
| GST_LOG_OBJECT (sink, "set blocksize to %u", blocksize); |
| GST_OBJECT_UNLOCK (sink); |
| } |
| |
| /** |
| * gst_base_sink_get_blocksize: |
| * @sink: a #GstBaseSink |
| * |
| * Get the number of bytes that the sink will pull when it is operating in pull |
| * mode. |
| * |
| * Returns: the number of bytes @sink will pull in pull mode. |
| * |
| * Since: 0.10.22 |
| */ |
| guint |
| gst_base_sink_get_blocksize (GstBaseSink * sink) |
| { |
| guint res; |
| |
| g_return_val_if_fail (GST_IS_BASE_SINK (sink), 0); |
| |
| GST_OBJECT_LOCK (sink); |
| res = sink->priv->blocksize; |
| GST_OBJECT_UNLOCK (sink); |
| |
| return res; |
| } |
| |
| static void |
| gst_base_sink_set_property (GObject * object, guint prop_id, |
| const GValue * value, GParamSpec * pspec) |
| { |
| GstBaseSink *sink = GST_BASE_SINK (object); |
| |
| switch (prop_id) { |
| case PROP_PREROLL_QUEUE_LEN: |
| /* preroll lock necessary to serialize with finish_preroll */ |
| GST_PAD_PREROLL_LOCK (sink->sinkpad); |
| sink->preroll_queue_max_len = g_value_get_uint (value); |
| GST_PAD_PREROLL_UNLOCK (sink->sinkpad); |
| break; |
| case PROP_SYNC: |
| gst_base_sink_set_sync (sink, g_value_get_boolean (value)); |
| break; |
| case PROP_MAX_LATENESS: |
| gst_base_sink_set_max_lateness (sink, g_value_get_int64 (value)); |
| break; |
| case PROP_QOS: |
| gst_base_sink_set_qos_enabled (sink, g_value_get_boolean (value)); |
| break; |
| case PROP_ASYNC: |
| gst_base_sink_set_async_enabled (sink, g_value_get_boolean (value)); |
| break; |
| case PROP_TS_OFFSET: |
| gst_base_sink_set_ts_offset (sink, g_value_get_int64 (value)); |
| break; |
| case PROP_BLOCKSIZE: |
| gst_base_sink_set_blocksize (sink, g_value_get_uint (value)); |
| break; |
| case PROP_RENDER_DELAY: |
| gst_base_sink_set_render_delay (sink, g_value_get_uint64 (value)); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| static void |
| gst_base_sink_get_property (GObject * object, guint prop_id, GValue * value, |
| GParamSpec * pspec) |
| { |
| GstBaseSink *sink = GST_BASE_SINK (object); |
| |
| switch (prop_id) { |
| case PROP_PREROLL_QUEUE_LEN: |
| GST_PAD_PREROLL_LOCK (sink->sinkpad); |
| g_value_set_uint (value, sink->preroll_queue_max_len); |
| GST_PAD_PREROLL_UNLOCK (sink->sinkpad); |
| break; |
| case PROP_SYNC: |
| g_value_set_boolean (value, gst_base_sink_get_sync (sink)); |
| break; |
| case PROP_MAX_LATENESS: |
| g_value_set_int64 (value, gst_base_sink_get_max_lateness (sink)); |
| break; |
| case PROP_QOS: |
| g_value_set_boolean (value, gst_base_sink_is_qos_enabled (sink)); |
| break; |
| case PROP_ASYNC: |
| g_value_set_boolean (value, gst_base_sink_is_async_enabled (sink)); |
| break; |
| case PROP_TS_OFFSET: |
| g_value_set_int64 (value, gst_base_sink_get_ts_offset (sink)); |
| break; |
| case PROP_LAST_BUFFER: |
| gst_value_take_buffer (value, gst_base_sink_get_last_buffer (sink)); |
| break; |
| case PROP_BLOCKSIZE: |
| g_value_set_uint (value, gst_base_sink_get_blocksize (sink)); |
| break; |
| case PROP_RENDER_DELAY: |
| g_value_set_uint64 (value, gst_base_sink_get_render_delay (sink)); |
| break; |
| default: |
| G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
| break; |
| } |
| } |
| |
| |
| static GstCaps * |
| gst_base_sink_get_caps (GstBaseSink * sink) |
| { |
| return NULL; |
| } |
| |
| static gboolean |
| gst_base_sink_set_caps (GstBaseSink * sink, GstCaps * caps) |
| { |
| return TRUE; |
| } |
| |
| static GstFlowReturn |
| gst_base_sink_buffer_alloc (GstBaseSink * sink, guint64 offset, guint size, |
| GstCaps * caps, GstBuffer ** buf) |
| { |
| *buf = NULL; |
| return GST_FLOW_OK; |
| } |
| |
| /* with PREROLL_LOCK, STREAM_LOCK */ |
| static void |
| gst_base_sink_preroll_queue_flush (GstBaseSink * basesink, GstPad * pad) |
| { |
| GstMiniObject *obj; |
| |
| GST_DEBUG_OBJECT (basesink, "flushing queue %p", basesink); |
| while ((obj = g_queue_pop_head (basesink->preroll_queue))) { |
| GST_DEBUG_OBJECT (basesink, "popped %p", obj); |
| gst_mini_object_unref (obj); |
| } |
| /* we can't have EOS anymore now */ |
| basesink->eos = FALSE; |
| basesink->priv->received_eos = FALSE; |
| basesink->have_preroll = FALSE; |
| basesink->priv->step_unlock = FALSE; |
| basesink->eos_queued = FALSE; |
| basesink->preroll_queued = 0; |
| basesink->buffers_queued = 0; |
| basesink->events_queued = 0; |
| /* can't report latency anymore until we preroll again */ |
| if (basesink->priv->async_enabled) { |
| GST_OBJECT_LOCK (basesink); |
| basesink->priv->have_latency = FALSE; |
| GST_OBJECT_UNLOCK (basesink); |
| } |
| /* and signal any waiters now */ |
| GST_PAD_PREROLL_SIGNAL (pad); |
| } |
| |
| /* with STREAM_LOCK, configures given segment with the event information. */ |
| static void |
| gst_base_sink_configure_segment (GstBaseSink * basesink, GstPad * pad, |
| GstEvent * event, GstSegment * segment) |
| { |
| gboolean update; |
| gdouble rate, arate; |
| GstFormat format; |
| gint64 start; |
| gint64 stop; |
| gint64 time; |
| |
| /* the newsegment event is needed to bring the buffer timestamps to the |
| * stream time and to drop samples outside of the playback segment. */ |
| gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, |
| &start, &stop, &time); |
| |
| /* The segment is protected with both the STREAM_LOCK and the OBJECT_LOCK. |
| * We protect with the OBJECT_LOCK so that we can use the values to |
| * safely answer a POSITION query. */ |
| GST_OBJECT_LOCK (basesink); |
| gst_segment_set_newsegment_full (segment, update, rate, arate, format, start, |
| stop, time); |
| |
| if (format == GST_FORMAT_TIME) { |
| GST_DEBUG_OBJECT (basesink, |
| "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " |
| "format GST_FORMAT_TIME, " |
| "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT |
| ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT, |
| update, rate, arate, GST_TIME_ARGS (segment->start), |
| GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time), |
| GST_TIME_ARGS (segment->accum)); |
| } else { |
| GST_DEBUG_OBJECT (basesink, |
| "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " |
| "format %d, " |
| "%" G_GINT64_FORMAT " -- %" G_GINT64_FORMAT ", time %" |
| G_GINT64_FORMAT ", accum %" G_GINT64_FORMAT, update, rate, arate, |
| segment->format, segment->start, segment->stop, segment->time, |
| segment->accum); |
| } |
| GST_OBJECT_UNLOCK (basesink); |
| } |
| |
| /* with PREROLL_LOCK, STREAM_LOCK */ |
| static gboolean |
| gst_base_sink_commit_state (GstBaseSink * basesink) |
| { |
| /* commit state and proceed to next pending state */ |
| GstState current, next, pending, post_pending; |
| gboolean post_paused = FALSE; |
| gboolean post_async_done = FALSE; |
| gboolean post_playing = FALSE; |
| |
| /* we are certainly not playing async anymore now */ |
| basesink->playing_async = FALSE; |
| |
| GST_OBJECT_LOCK (basesink); |
| current = GST_STATE (basesink); |
| next = GST_STATE_NEXT (basesink); |
| pending = GST_STATE_PENDING (basesink); |
| post_pending = pending; |
| |
| switch (pending) { |
| case GST_STATE_PLAYING: |
| { |
| GstBaseSinkClass *bclass; |
| GstStateChangeReturn ret; |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| GST_DEBUG_OBJECT (basesink, "commiting state to PLAYING"); |
| |
| basesink->need_preroll = FALSE; |
| post_async_done = TRUE; |
| basesink->priv->commited = TRUE; |
| post_playing = TRUE; |
| /* post PAUSED too when we were READY */ |
| if (current == GST_STATE_READY) { |
| post_paused = TRUE; |
| } |
| |
| /* make sure we notify the subclass of async playing */ |
| if (bclass->async_play) { |
| GST_WARNING_OBJECT (basesink, "deprecated async_play"); |
| ret = bclass->async_play (basesink); |
| if (ret == GST_STATE_CHANGE_FAILURE) |
| goto async_failed; |
| } |
| break; |
| } |
| case GST_STATE_PAUSED: |
| GST_DEBUG_OBJECT (basesink, "commiting state to PAUSED"); |
| post_paused = TRUE; |
| post_async_done = TRUE; |
| basesink->priv->commited = TRUE; |
| post_pending = GST_STATE_VOID_PENDING; |
| break; |
| case GST_STATE_READY: |
| case GST_STATE_NULL: |
| goto stopping; |
| case GST_STATE_VOID_PENDING: |
| goto nothing_pending; |
| default: |
| break; |
| } |
| |
| /* we can report latency queries now */ |
| basesink->priv->have_latency = TRUE; |
| |
| GST_STATE (basesink) = pending; |
| GST_STATE_NEXT (basesink) = GST_STATE_VOID_PENDING; |
| GST_STATE_PENDING (basesink) = GST_STATE_VOID_PENDING; |
| GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_SUCCESS; |
| GST_OBJECT_UNLOCK (basesink); |
| |
| if (post_paused) { |
| GST_DEBUG_OBJECT (basesink, "posting PAUSED state change message"); |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_state_changed (GST_OBJECT_CAST (basesink), |
| current, next, post_pending)); |
| } |
| if (post_async_done) { |
| GST_DEBUG_OBJECT (basesink, "posting async-done message"); |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_async_done (GST_OBJECT_CAST (basesink))); |
| } |
| if (post_playing) { |
| GST_DEBUG_OBJECT (basesink, "posting PLAYING state change message"); |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_state_changed (GST_OBJECT_CAST (basesink), |
| next, pending, GST_STATE_VOID_PENDING)); |
| } |
| |
| GST_STATE_BROADCAST (basesink); |
| |
| return TRUE; |
| |
| nothing_pending: |
| { |
| /* Depending on the state, set our vars. We get in this situation when the |
| * state change function got a change to update the state vars before the |
| * streaming thread did. This is fine but we need to make sure that we |
| * update the need_preroll var since it was TRUE when we got here and might |
| * become FALSE if we got to PLAYING. */ |
| GST_DEBUG_OBJECT (basesink, "nothing to commit, now in %s", |
| gst_element_state_get_name (current)); |
| switch (current) { |
| case GST_STATE_PLAYING: |
| basesink->need_preroll = FALSE; |
| break; |
| case GST_STATE_PAUSED: |
| basesink->need_preroll = TRUE; |
| break; |
| default: |
| basesink->need_preroll = FALSE; |
| basesink->flushing = TRUE; |
| break; |
| } |
| /* we can report latency queries now */ |
| basesink->priv->have_latency = TRUE; |
| GST_OBJECT_UNLOCK (basesink); |
| return TRUE; |
| } |
| stopping: |
| { |
| /* app is going to READY */ |
| GST_DEBUG_OBJECT (basesink, "stopping"); |
| basesink->need_preroll = FALSE; |
| basesink->flushing = TRUE; |
| GST_OBJECT_UNLOCK (basesink); |
| return FALSE; |
| } |
| async_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, "async commit failed"); |
| GST_STATE_RETURN (basesink) = GST_STATE_CHANGE_FAILURE; |
| GST_OBJECT_UNLOCK (basesink); |
| return FALSE; |
| } |
| } |
| |
| static void |
| start_stepping (GstBaseSink * sink, GstSegment * segment, |
| GstStepInfo * pending, GstStepInfo * current) |
| { |
| gint64 end; |
| GstMessage *message; |
| |
| GST_DEBUG_OBJECT (sink, "update pending step"); |
| |
| GST_OBJECT_LOCK (sink); |
| memcpy (current, pending, sizeof (GstStepInfo)); |
| pending->valid = FALSE; |
| GST_OBJECT_UNLOCK (sink); |
| |
| /* post message first */ |
| message = |
| gst_message_new_step_start (GST_OBJECT (sink), TRUE, current->format, |
| current->amount, current->rate, current->flush, current->intermediate); |
| gst_message_set_seqnum (message, current->seqnum); |
| gst_element_post_message (GST_ELEMENT (sink), message); |
| |
| /* get the running time of where we paused and remember it */ |
| current->start = gst_element_get_start_time (GST_ELEMENT_CAST (sink)); |
| gst_segment_set_running_time (segment, GST_FORMAT_TIME, current->start); |
| |
| /* set the new rate for the remainder of the segment */ |
| current->start_rate = segment->rate; |
| segment->rate *= current->rate; |
| segment->abs_rate = ABS (segment->rate); |
| |
| /* save values */ |
| if (segment->rate > 0.0) |
| current->start_stop = segment->stop; |
| else |
| current->start_start = segment->start; |
| |
| if (current->format == GST_FORMAT_TIME) { |
| end = current->start + current->amount; |
| if (!current->flush) { |
| /* update the segment clipping regions for non-flushing seeks */ |
| if (segment->rate > 0.0) { |
| segment->stop = gst_segment_to_position (segment, GST_FORMAT_TIME, end); |
| segment->last_stop = segment->stop; |
| } else { |
| gint64 position; |
| |
| position = gst_segment_to_position (segment, GST_FORMAT_TIME, end); |
| segment->time = position; |
| segment->start = position; |
| segment->last_stop = position; |
| } |
| } |
| } |
| |
| GST_DEBUG_OBJECT (sink, |
| "segment now rate %lf, applied rate %lf, " |
| "format GST_FORMAT_TIME, " |
| "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT |
| ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT, |
| segment->rate, segment->applied_rate, GST_TIME_ARGS (segment->start), |
| GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time), |
| GST_TIME_ARGS (segment->accum)); |
| |
| GST_DEBUG_OBJECT (sink, "step started at running_time %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (current->start)); |
| |
| if (current->amount == -1) { |
| GST_DEBUG_OBJECT (sink, "step amount == -1, stop stepping"); |
| current->valid = FALSE; |
| } else { |
| GST_DEBUG_OBJECT (sink, "step amount: %" G_GUINT64_FORMAT ", format: %s, " |
| "rate: %f", current->amount, gst_format_get_name (current->format), |
| current->rate); |
| } |
| } |
| |
| static void |
| stop_stepping (GstBaseSink * sink, GstSegment * segment, |
| GstStepInfo * current, gint64 rstart, gint64 rstop, gboolean eos) |
| { |
| gint64 stop, position; |
| GstMessage *message; |
| |
| GST_DEBUG_OBJECT (sink, "step complete"); |
| |
| if (segment->rate > 0.0) |
| stop = rstart; |
| else |
| stop = rstop; |
| |
| GST_DEBUG_OBJECT (sink, |
| "step stop at running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (stop)); |
| |
| if (stop == -1) |
| current->duration = current->position; |
| else |
| current->duration = stop - current->start; |
| |
| GST_DEBUG_OBJECT (sink, "step elapsed running_time %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (current->duration)); |
| |
| position = current->start + current->duration; |
| |
| /* now move the segment to the new running time */ |
| gst_segment_set_running_time (segment, GST_FORMAT_TIME, position); |
| |
| if (current->flush) { |
| /* and remove the accumulated time we flushed, start time did not change */ |
| segment->accum = current->start; |
| } else { |
| /* start time is now the stepped position */ |
| gst_element_set_start_time (GST_ELEMENT_CAST (sink), position); |
| } |
| |
| /* restore the previous rate */ |
| segment->rate = current->start_rate; |
| segment->abs_rate = ABS (segment->rate); |
| |
| if (segment->rate > 0.0) |
| segment->stop = current->start_stop; |
| else |
| segment->start = current->start_start; |
| |
| /* the clip segment is used for position report in paused... */ |
| memcpy (sink->abidata.ABI.clip_segment, segment, sizeof (GstSegment)); |
| |
| /* post the step done when we know the stepped duration in TIME */ |
| message = |
| gst_message_new_step_done (GST_OBJECT_CAST (sink), current->format, |
| current->amount, current->rate, current->flush, current->intermediate, |
| current->duration, eos); |
| gst_message_set_seqnum (message, current->seqnum); |
| gst_element_post_message (GST_ELEMENT_CAST (sink), message); |
| |
| if (!current->intermediate) |
| sink->need_preroll = current->need_preroll; |
| |
| /* and the current step info finished and becomes invalid */ |
| current->valid = FALSE; |
| } |
| |
| static gboolean |
| handle_stepping (GstBaseSink * sink, GstSegment * segment, |
| GstStepInfo * current, gint64 * cstart, gint64 * cstop, gint64 * rstart, |
| gint64 * rstop) |
| { |
| gboolean step_end = FALSE; |
| |
| /* see if we need to skip this buffer because of stepping */ |
| switch (current->format) { |
| case GST_FORMAT_TIME: |
| { |
| guint64 end; |
| gint64 first, last; |
| |
| if (segment->rate > 0.0) { |
| first = *rstart; |
| last = *rstop; |
| } else { |
| first = *rstop; |
| last = *rstart; |
| } |
| |
| end = current->start + current->amount; |
| current->position = first - current->start; |
| |
| if (G_UNLIKELY (segment->abs_rate != 1.0)) |
| current->position /= segment->abs_rate; |
| |
| GST_DEBUG_OBJECT (sink, |
| "buffer: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT, |
| GST_TIME_ARGS (first), GST_TIME_ARGS (last)); |
| GST_DEBUG_OBJECT (sink, |
| "got time step %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT "/%" |
| GST_TIME_FORMAT, GST_TIME_ARGS (current->position), |
| GST_TIME_ARGS (last - current->start), |
| GST_TIME_ARGS (current->amount)); |
| |
| if ((current->flush && current->position >= current->amount) |
| || last >= end) { |
| GST_DEBUG_OBJECT (sink, "step ended, we need clipping"); |
| step_end = TRUE; |
| if (segment->rate > 0.0) { |
| *rstart = end; |
| *cstart = gst_segment_to_position (segment, GST_FORMAT_TIME, end); |
| } else { |
| *rstop = end; |
| *cstop = gst_segment_to_position (segment, GST_FORMAT_TIME, end); |
| } |
| } |
| GST_DEBUG_OBJECT (sink, |
| "cstart %" GST_TIME_FORMAT ", rstart %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (*cstart), GST_TIME_ARGS (*rstart)); |
| GST_DEBUG_OBJECT (sink, |
| "cstop %" GST_TIME_FORMAT ", rstop %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (*cstop), GST_TIME_ARGS (*rstop)); |
| break; |
| } |
| case GST_FORMAT_BUFFERS: |
| GST_DEBUG_OBJECT (sink, |
| "got default step %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT, |
| current->position, current->amount); |
| |
| if (current->position < current->amount) { |
| current->position++; |
| } else { |
| step_end = TRUE; |
| } |
| break; |
| case GST_FORMAT_DEFAULT: |
| default: |
| GST_DEBUG_OBJECT (sink, |
| "got unknown step %" G_GUINT64_FORMAT "/%" G_GUINT64_FORMAT, |
| current->position, current->amount); |
| break; |
| } |
| return step_end; |
| } |
| |
| /* with STREAM_LOCK, PREROLL_LOCK |
| * |
| * Returns TRUE if the object needs synchronisation and takes therefore |
| * part in prerolling. |
| * |
| * rsstart/rsstop contain the start/stop in stream time. |
| * rrstart/rrstop contain the start/stop in running time. |
| */ |
| static gboolean |
| gst_base_sink_get_sync_times (GstBaseSink * basesink, GstMiniObject * obj, |
| GstClockTime * rsstart, GstClockTime * rsstop, |
| GstClockTime * rrstart, GstClockTime * rrstop, gboolean * do_sync, |
| gboolean * stepped, GstSegment * segment, GstStepInfo * step, |
| gboolean * step_end) |
| { |
| GstBaseSinkClass *bclass; |
| GstBuffer *buffer; |
| GstClockTime start, stop; /* raw start/stop timestamps */ |
| gint64 cstart, cstop; /* clipped raw timestamps */ |
| gint64 rstart, rstop; /* clipped timestamps converted to running time */ |
| GstClockTime sstart, sstop; /* clipped timestamps converted to stream time */ |
| GstFormat format; |
| GstBaseSinkPrivate *priv; |
| gboolean eos; |
| |
| priv = basesink->priv; |
| |
| /* start with nothing */ |
| start = stop = -1; |
| |
| if (G_UNLIKELY (GST_IS_EVENT (obj))) { |
| GstEvent *event = GST_EVENT_CAST (obj); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| /* EOS event needs syncing */ |
| case GST_EVENT_EOS: |
| { |
| if (basesink->segment.rate >= 0.0) { |
| sstart = sstop = priv->current_sstop; |
| if (sstart == -1) { |
| /* we have not seen a buffer yet, use the segment values */ |
| sstart = sstop = gst_segment_to_stream_time (&basesink->segment, |
| basesink->segment.format, basesink->segment.stop); |
| } |
| } else { |
| sstart = sstop = priv->current_sstart; |
| if (sstart == -1) { |
| /* we have not seen a buffer yet, use the segment values */ |
| sstart = sstop = gst_segment_to_stream_time (&basesink->segment, |
| basesink->segment.format, basesink->segment.start); |
| } |
| } |
| |
| rstart = rstop = priv->eos_rtime; |
| *do_sync = rstart != -1; |
| GST_DEBUG_OBJECT (basesink, "sync times for EOS %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (rstart)); |
| /* if we are stepping, we end now */ |
| *step_end = step->valid; |
| eos = TRUE; |
| goto eos_done; |
| } |
| default: |
| /* other events do not need syncing */ |
| /* FIXME, maybe NEWSEGMENT might need synchronisation |
| * since the POSITION query depends on accumulated times and |
| * we cannot accumulate the current segment before the previous |
| * one completed. |
| */ |
| return FALSE; |
| } |
| } |
| |
| eos = FALSE; |
| |
| /* else do buffer sync code */ |
| buffer = GST_BUFFER_CAST (obj); |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| /* just get the times to see if we need syncing, if the start returns -1 we |
| * don't sync. */ |
| if (bclass->get_times) |
| bclass->get_times (basesink, buffer, &start, &stop); |
| |
| if (start == -1) { |
| /* we don't need to sync but we still want to get the timestamps for |
| * tracking the position */ |
| gst_base_sink_get_times (basesink, buffer, &start, &stop); |
| *do_sync = FALSE; |
| } else { |
| *do_sync = TRUE; |
| } |
| |
| GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT |
| ", stop: %" GST_TIME_FORMAT ", do_sync %d", GST_TIME_ARGS (start), |
| GST_TIME_ARGS (stop), *do_sync); |
| |
| /* collect segment and format for code clarity */ |
| format = segment->format; |
| |
| /* no timestamp clipping if we did not get a TIME segment format */ |
| if (G_UNLIKELY (format != GST_FORMAT_TIME)) { |
| cstart = start; |
| cstop = stop; |
| /* do running and stream time in TIME format */ |
| format = GST_FORMAT_TIME; |
| GST_LOG_OBJECT (basesink, "not time format, don't clip"); |
| goto do_times; |
| } |
| |
| /* clip, only when we know about time */ |
| if (G_UNLIKELY (!gst_segment_clip (segment, GST_FORMAT_TIME, |
| (gint64) start, (gint64) stop, &cstart, &cstop))) { |
| if (step->valid) { |
| GST_DEBUG_OBJECT (basesink, "step out of segment"); |
| /* when we are stepping, pretend we're at the end of the segment */ |
| if (segment->rate > 0.0) { |
| cstart = segment->stop; |
| cstop = segment->stop; |
| } else { |
| cstart = segment->start; |
| cstop = segment->start; |
| } |
| goto do_times; |
| } |
| goto out_of_segment; |
| } |
| |
| if (G_UNLIKELY (start != cstart || stop != cstop)) { |
| GST_DEBUG_OBJECT (basesink, "clipped to: start %" GST_TIME_FORMAT |
| ", stop: %" GST_TIME_FORMAT, GST_TIME_ARGS (cstart), |
| GST_TIME_ARGS (cstop)); |
| } |
| |
| /* set last stop position */ |
| if (G_LIKELY (cstop != GST_CLOCK_TIME_NONE)) |
| gst_segment_set_last_stop (segment, GST_FORMAT_TIME, cstop); |
| else |
| gst_segment_set_last_stop (segment, GST_FORMAT_TIME, cstart); |
| |
| do_times: |
| rstart = gst_segment_to_running_time (segment, format, cstart); |
| rstop = gst_segment_to_running_time (segment, format, cstop); |
| |
| if (G_UNLIKELY (step->valid)) { |
| if (!(*step_end = handle_stepping (basesink, segment, step, &cstart, &cstop, |
| &rstart, &rstop))) { |
| /* step is still busy, we discard data when we are flushing */ |
| *stepped = step->flush; |
| } |
| } |
| /* this can produce wrong values if we accumulated non-TIME segments. If this happens, |
| * upstream is behaving very badly */ |
| sstart = gst_segment_to_stream_time (segment, format, cstart); |
| sstop = gst_segment_to_stream_time (segment, format, cstop); |
| |
| eos_done: |
| /* eos_done label only called when doing EOS, we also stop stepping then */ |
| if (*step_end && step->flush) { |
| GST_DEBUG_OBJECT (basesink, "flushing step ended"); |
| stop_stepping (basesink, segment, step, rstart, rstop, eos); |
| *step_end = FALSE; |
| } |
| |
| /* save times */ |
| *rsstart = sstart; |
| *rsstop = sstop; |
| *rrstart = rstart; |
| *rrstop = rstop; |
| |
| /* buffers and EOS always need syncing and preroll */ |
| return TRUE; |
| |
| /* special cases */ |
| out_of_segment: |
| { |
| /* we usually clip in the chain function already but stepping could cause |
| * the segment to be updated later. we return FALSE so that we don't try |
| * to sync on it. */ |
| GST_LOG_OBJECT (basesink, "buffer skipped, not in segment"); |
| return FALSE; |
| } |
| } |
| |
| /* with STREAM_LOCK, PREROLL_LOCK, LOCK |
| * adjust a timestamp with the latency and timestamp offset */ |
| static GstClockTime |
| gst_base_sink_adjust_time (GstBaseSink * basesink, GstClockTime time) |
| { |
| GstClockTimeDiff ts_offset; |
| |
| /* don't do anything funny with invalid timestamps */ |
| if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) |
| return time; |
| |
| time += basesink->priv->latency; |
| |
| /* apply offset, be carefull for underflows */ |
| ts_offset = basesink->priv->ts_offset; |
| if (ts_offset < 0) { |
| ts_offset = -ts_offset; |
| if (ts_offset < time) |
| time -= ts_offset; |
| else |
| time = 0; |
| } else |
| time += ts_offset; |
| |
| return time; |
| } |
| |
| /** |
| * gst_base_sink_wait_clock: |
| * @sink: the sink |
| * @time: the running_time to be reached |
| * @jitter: the jitter to be filled with time diff (can be NULL) |
| * |
| * This function will block until @time is reached. It is usually called by |
| * subclasses that use their own internal synchronisation. |
| * |
| * If @time is not valid, no sycnhronisation is done and #GST_CLOCK_BADTIME is |
| * returned. Likewise, if synchronisation is disabled in the element or there |
| * is no clock, no synchronisation is done and #GST_CLOCK_BADTIME is returned. |
| * |
| * This function should only be called with the PREROLL_LOCK held, like when |
| * receiving an EOS event in the ::event vmethod or when receiving a buffer in |
| * the ::render vmethod. |
| * |
| * The @time argument should be the running_time of when this method should |
| * return and is not adjusted with any latency or offset configured in the |
| * sink. |
| * |
| * Since 0.10.20 |
| * |
| * Returns: #GstClockReturn |
| */ |
| GstClockReturn |
| gst_base_sink_wait_clock (GstBaseSink * sink, GstClockTime time, |
| GstClockTimeDiff * jitter) |
| { |
| GstClockID id; |
| GstClockReturn ret; |
| GstClock *clock; |
| |
| if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) |
| goto invalid_time; |
| |
| GST_OBJECT_LOCK (sink); |
| if (G_UNLIKELY (!sink->sync)) |
| goto no_sync; |
| |
| if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (sink)) == NULL)) |
| goto no_clock; |
| |
| /* add base_time to running_time to get the time against the clock */ |
| time += GST_ELEMENT_CAST (sink)->base_time; |
| |
| id = gst_clock_new_single_shot_id (clock, time); |
| GST_OBJECT_UNLOCK (sink); |
| |
| /* A blocking wait is performed on the clock. We save the ClockID |
| * so we can unlock the entry at any time. While we are blocking, we |
| * release the PREROLL_LOCK so that other threads can interrupt the |
| * entry. */ |
| sink->clock_id = id; |
| /* release the preroll lock while waiting */ |
| GST_PAD_PREROLL_UNLOCK (sink->sinkpad); |
| |
| ret = gst_clock_id_wait (id, jitter); |
| |
| GST_PAD_PREROLL_LOCK (sink->sinkpad); |
| gst_clock_id_unref (id); |
| sink->clock_id = NULL; |
| |
| return ret; |
| |
| /* no syncing needed */ |
| invalid_time: |
| { |
| GST_DEBUG_OBJECT (sink, "time not valid, no sync needed"); |
| return GST_CLOCK_BADTIME; |
| } |
| no_sync: |
| { |
| GST_DEBUG_OBJECT (sink, "sync disabled"); |
| GST_OBJECT_UNLOCK (sink); |
| return GST_CLOCK_BADTIME; |
| } |
| no_clock: |
| { |
| GST_DEBUG_OBJECT (sink, "no clock, can't sync"); |
| GST_OBJECT_UNLOCK (sink); |
| return GST_CLOCK_BADTIME; |
| } |
| } |
| |
| /** |
| * gst_base_sink_wait_preroll: |
| * @sink: the sink |
| * |
| * If the #GstBaseSinkClass::render method performs its own synchronisation against |
| * the clock it must unblock when going from PLAYING to the PAUSED state and call |
| * this method before continuing to render the remaining data. |
| * |
| * This function will block until a state change to PLAYING happens (in which |
| * case this function returns #GST_FLOW_OK) or the processing must be stopped due |
| * to a state change to READY or a FLUSH event (in which case this function |
| * returns #GST_FLOW_WRONG_STATE). |
| * |
| * This function should only be called with the PREROLL_LOCK held, like in the |
| * render function. |
| * |
| * Since: 0.10.11 |
| * |
| * Returns: #GST_FLOW_OK if the preroll completed and processing can |
| * continue. Any other return value should be returned from the render vmethod. |
| */ |
| GstFlowReturn |
| gst_base_sink_wait_preroll (GstBaseSink * sink) |
| { |
| sink->have_preroll = TRUE; |
| GST_DEBUG_OBJECT (sink, "waiting in preroll for flush or PLAYING"); |
| /* block until the state changes, or we get a flush, or something */ |
| GST_PAD_PREROLL_WAIT (sink->sinkpad); |
| sink->have_preroll = FALSE; |
| if (G_UNLIKELY (sink->flushing)) |
| goto stopping; |
| if (G_UNLIKELY (sink->priv->step_unlock)) |
| goto step_unlocked; |
| GST_DEBUG_OBJECT (sink, "continue after preroll"); |
| |
| return GST_FLOW_OK; |
| |
| /* ERRORS */ |
| stopping: |
| { |
| GST_DEBUG_OBJECT (sink, "preroll interrupted because of flush"); |
| return GST_FLOW_WRONG_STATE; |
| } |
| step_unlocked: |
| { |
| sink->priv->step_unlock = FALSE; |
| GST_DEBUG_OBJECT (sink, "preroll interrupted because of step"); |
| return GST_FLOW_STEP; |
| } |
| } |
| |
| /** |
| * gst_base_sink_do_preroll: |
| * @sink: the sink |
| * @obj: the object that caused the preroll |
| * |
| * If the @sink spawns its own thread for pulling buffers from upstream it |
| * should call this method after it has pulled a buffer. If the element needed |
| * to preroll, this function will perform the preroll and will then block |
| * until the element state is changed. |
| * |
| * This function should be called with the PREROLL_LOCK held. |
| * |
| * Since 0.10.22 |
| * |
| * Returns: #GST_FLOW_OK if the preroll completed and processing can |
| * continue. Any other return value should be returned from the render vmethod. |
| */ |
| GstFlowReturn |
| gst_base_sink_do_preroll (GstBaseSink * sink, GstMiniObject * obj) |
| { |
| GstFlowReturn ret; |
| |
| while (G_UNLIKELY (sink->need_preroll)) { |
| GST_DEBUG_OBJECT (sink, "prerolling object %p", obj); |
| |
| ret = gst_base_sink_preroll_object (sink, FALSE, obj); |
| if (ret != GST_FLOW_OK) |
| goto preroll_failed; |
| |
| /* need to recheck here because the commit state could have |
| * made us not need the preroll anymore */ |
| if (G_LIKELY (sink->need_preroll)) { |
| /* block until the state changes, or we get a flush, or something */ |
| ret = gst_base_sink_wait_preroll (sink); |
| if (ret != GST_FLOW_OK) { |
| if (ret == GST_FLOW_STEP) |
| ret = GST_FLOW_OK; |
| else |
| goto preroll_failed; |
| } |
| } |
| } |
| return GST_FLOW_OK; |
| |
| /* ERRORS */ |
| preroll_failed: |
| { |
| GST_DEBUG_OBJECT (sink, "preroll failed %d", ret); |
| return ret; |
| } |
| } |
| |
| /** |
| * gst_base_sink_wait_eos: |
| * @sink: the sink |
| * @time: the running_time to be reached |
| * @jitter: the jitter to be filled with time diff (can be NULL) |
| * |
| * This function will block until @time is reached. It is usually called by |
| * subclasses that use their own internal synchronisation but want to let the |
| * EOS be handled by the base class. |
| * |
| * This function should only be called with the PREROLL_LOCK held, like when |
| * receiving an EOS event in the ::event vmethod. |
| * |
| * The @time argument should be the running_time of when the EOS should happen |
| * and will be adjusted with any latency and offset configured in the sink. |
| * |
| * Since 0.10.15 |
| * |
| * Returns: #GstFlowReturn |
| */ |
| GstFlowReturn |
| gst_base_sink_wait_eos (GstBaseSink * sink, GstClockTime time, |
| GstClockTimeDiff * jitter) |
| { |
| GstClockReturn status; |
| GstFlowReturn ret; |
| |
| do { |
| GstClockTime stime; |
| |
| GST_DEBUG_OBJECT (sink, "checking preroll"); |
| |
| /* first wait for the playing state before we can continue */ |
| if (G_UNLIKELY (sink->need_preroll)) { |
| ret = gst_base_sink_wait_preroll (sink); |
| if (ret != GST_FLOW_OK) { |
| if (ret == GST_FLOW_STEP) |
| ret = GST_FLOW_OK; |
| else |
| goto flushing; |
| } |
| } |
| |
| /* preroll done, we can sync since we are in PLAYING now. */ |
| GST_DEBUG_OBJECT (sink, "possibly waiting for clock to reach %" |
| GST_TIME_FORMAT, GST_TIME_ARGS (time)); |
| |
| /* compensate for latency and ts_offset. We don't adjust for render delay |
| * because we don't interact with the device on EOS normally. */ |
| stime = gst_base_sink_adjust_time (sink, time); |
| |
| /* wait for the clock, this can be interrupted because we got shut down or |
| * we PAUSED. */ |
| status = gst_base_sink_wait_clock (sink, stime, jitter); |
| |
| GST_DEBUG_OBJECT (sink, "clock returned %d", status); |
| |
| /* invalid time, no clock or sync disabled, just continue then */ |
| if (status == GST_CLOCK_BADTIME) |
| break; |
| |
| /* waiting could have been interrupted and we can be flushing now */ |
| if (G_UNLIKELY (sink->flushing)) |
| goto flushing; |
| |
| /* retry if we got unscheduled, which means we did not reach the timeout |
| * yet. if some other error occures, we continue. */ |
| } while (status == GST_CLOCK_UNSCHEDULED); |
| |
| GST_DEBUG_OBJECT (sink, "end of stream"); |
| |
| return GST_FLOW_OK; |
| |
| /* ERRORS */ |
| flushing: |
| { |
| GST_DEBUG_OBJECT (sink, "we are flushing"); |
| return GST_FLOW_WRONG_STATE; |
| } |
| } |
| |
| /* with STREAM_LOCK, PREROLL_LOCK |
| * |
| * Make sure we are in PLAYING and synchronize an object to the clock. |
| * |
| * If we need preroll, we are not in PLAYING. We try to commit the state |
| * if needed and then block if we still are not PLAYING. |
| * |
| * We start waiting on the clock in PLAYING. If we got interrupted, we |
| * immediatly try to re-preroll. |
| * |
| * Some objects do not need synchronisation (most events) and so this function |
| * immediatly returns GST_FLOW_OK. |
| * |
| * for objects that arrive later than max-lateness to be synchronized to the |
| * clock have the @late boolean set to TRUE. |
| * |
| * This function keeps a running average of the jitter (the diff between the |
| * clock time and the requested sync time). The jitter is negative for |
| * objects that arrive in time and positive for late buffers. |
| * |
| * does not take ownership of obj. |
| */ |
| static GstFlowReturn |
| gst_base_sink_do_sync (GstBaseSink * basesink, GstPad * pad, |
| GstMiniObject * obj, gboolean * late, gboolean * step_end) |
| { |
| GstClockTimeDiff jitter; |
| gboolean syncable; |
| GstClockReturn status = GST_CLOCK_OK; |
| GstClockTime rstart, rstop, sstart, sstop, stime; |
| gboolean do_sync; |
| GstBaseSinkPrivate *priv; |
| GstFlowReturn ret; |
| GstStepInfo *current, *pending; |
| gboolean stepped; |
| |
| priv = basesink->priv; |
| |
| do_step: |
| sstart = sstop = rstart = rstop = -1; |
| do_sync = TRUE; |
| stepped = FALSE; |
| |
| priv->current_rstart = -1; |
| |
| /* get stepping info */ |
| current = &priv->current_step; |
| pending = &priv->pending_step; |
| |
| /* get timing information for this object against the render segment */ |
| syncable = gst_base_sink_get_sync_times (basesink, obj, |
| &sstart, &sstop, &rstart, &rstop, &do_sync, &stepped, &basesink->segment, |
| current, step_end); |
| |
| if (G_UNLIKELY (stepped)) |
| goto step_skipped; |
| |
| /* a syncable object needs to participate in preroll and |
| * clocking. All buffers and EOS are syncable. */ |
| if (G_UNLIKELY (!syncable)) |
| goto not_syncable; |
| |
| /* store timing info for current object */ |
| priv->current_rstart = rstart; |
| priv->current_rstop = (rstop != -1 ? rstop : rstart); |
| |
| /* save sync time for eos when the previous object needed sync */ |
| priv->eos_rtime = (do_sync ? priv->current_rstop : -1); |
| |
| again: |
| /* first do preroll, this makes sure we commit our state |
| * to PAUSED and can continue to PLAYING. We cannot perform |
| * any clock sync in PAUSED because there is no clock. */ |
| ret = gst_base_sink_do_preroll (basesink, obj); |
| if (G_UNLIKELY (ret != GST_FLOW_OK)) |
| goto preroll_failed; |
| |
| /* After rendering we store the position of the last buffer so that we can use |
| * it to report the position. We need to take the lock here. */ |
| GST_OBJECT_LOCK (basesink); |
| priv->current_sstart = sstart; |
| priv->current_sstop = (sstop != -1 ? sstop : sstart); |
| GST_OBJECT_UNLOCK (basesink); |
| |
| /* update the segment with a pending step if the current one is invalid and we |
| * have a new pending one. We only accept new step updates after a preroll */ |
| if (G_UNLIKELY (pending->valid && !current->valid)) { |
| start_stepping (basesink, &basesink->segment, pending, current); |
| goto do_step; |
| } |
| |
| if (!do_sync) |
| goto done; |
| |
| /* adjust for latency */ |
| stime = gst_base_sink_adjust_time (basesink, rstart); |
| |
| /* adjust for render-delay, avoid underflows */ |
| if (stime != -1) { |
| if (stime > priv->render_delay) |
| stime -= priv->render_delay; |
| else |
| stime = 0; |
| } |
| |
| /* preroll done, we can sync since we are in PLAYING now. */ |
| GST_DEBUG_OBJECT (basesink, "possibly waiting for clock to reach %" |
| GST_TIME_FORMAT ", adjusted %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (rstart), GST_TIME_ARGS (stime)); |
| |
| /* This function will return immediatly if start == -1, no clock |
| * or sync is disabled with GST_CLOCK_BADTIME. */ |
| status = gst_base_sink_wait_clock (basesink, stime, &jitter); |
| |
| GST_DEBUG_OBJECT (basesink, "clock returned %d", status); |
| |
| /* invalid time, no clock or sync disabled, just render */ |
| if (status == GST_CLOCK_BADTIME) |
| goto done; |
| |
| /* waiting could have been interrupted and we can be flushing now */ |
| if (G_UNLIKELY (basesink->flushing)) |
| goto flushing; |
| |
| /* check for unlocked by a state change, we are not flushing so |
| * we can try to preroll on the current buffer. */ |
| if (G_UNLIKELY (status == GST_CLOCK_UNSCHEDULED)) { |
| GST_DEBUG_OBJECT (basesink, "unscheduled, waiting some more"); |
| priv->call_preroll = TRUE; |
| goto again; |
| } |
| |
| /* successful syncing done, record observation */ |
| priv->current_jitter = jitter; |
| |
| /* check if the object should be dropped */ |
| *late = gst_base_sink_is_too_late (basesink, obj, rstart, rstop, |
| status, jitter); |
| |
| done: |
| return GST_FLOW_OK; |
| |
| /* ERRORS */ |
| step_skipped: |
| { |
| GST_DEBUG_OBJECT (basesink, "skipped stepped object %p", obj); |
| *late = TRUE; |
| return GST_FLOW_OK; |
| } |
| not_syncable: |
| { |
| GST_DEBUG_OBJECT (basesink, "non syncable object %p", obj); |
| return GST_FLOW_OK; |
| } |
| flushing: |
| { |
| GST_DEBUG_OBJECT (basesink, "we are flushing"); |
| return GST_FLOW_WRONG_STATE; |
| } |
| preroll_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, "preroll failed"); |
| *step_end = FALSE; |
| return ret; |
| } |
| } |
| |
| static gboolean |
| gst_base_sink_send_qos (GstBaseSink * basesink, |
| gdouble proportion, GstClockTime time, GstClockTimeDiff diff) |
| { |
| GstEvent *event; |
| gboolean res; |
| |
| /* generate Quality-of-Service event */ |
| GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink, |
| "qos: proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %" |
| GST_TIME_FORMAT, proportion, diff, GST_TIME_ARGS (time)); |
| |
| event = gst_event_new_qos (proportion, diff, time); |
| |
| /* send upstream */ |
| res = gst_pad_push_event (basesink->sinkpad, event); |
| |
| return res; |
| } |
| |
| static void |
| gst_base_sink_perform_qos (GstBaseSink * sink, gboolean dropped) |
| { |
| GstBaseSinkPrivate *priv; |
| GstClockTime start, stop; |
| GstClockTimeDiff jitter; |
| GstClockTime pt, entered, left; |
| GstClockTime duration; |
| gdouble rate; |
| |
| priv = sink->priv; |
| |
| start = priv->current_rstart; |
| |
| if (priv->current_step.valid) |
| return; |
| |
| /* if Quality-of-Service disabled, do nothing */ |
| if (!g_atomic_int_get (&priv->qos_enabled) || start == -1) |
| return; |
| |
| stop = priv->current_rstop; |
| jitter = priv->current_jitter; |
| |
| if (jitter < 0) { |
| /* this is the time the buffer entered the sink */ |
| if (start < -jitter) |
| entered = 0; |
| else |
| entered = start + jitter; |
| left = start; |
| } else { |
| /* this is the time the buffer entered the sink */ |
| entered = start + jitter; |
| /* this is the time the buffer left the sink */ |
| left = start + jitter; |
| } |
| |
| /* calculate duration of the buffer */ |
| if (stop != -1) |
| duration = stop - start; |
| else |
| duration = -1; |
| |
| /* if we have the time when the last buffer left us, calculate |
| * processing time */ |
| if (priv->last_left != -1) { |
| if (entered > priv->last_left) { |
| pt = entered - priv->last_left; |
| } else { |
| pt = 0; |
| } |
| } else { |
| pt = priv->avg_pt; |
| } |
| |
| GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "start: %" GST_TIME_FORMAT |
| ", entered %" GST_TIME_FORMAT ", left %" GST_TIME_FORMAT ", pt: %" |
| GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ",jitter %" |
| G_GINT64_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (entered), |
| GST_TIME_ARGS (left), GST_TIME_ARGS (pt), GST_TIME_ARGS (duration), |
| jitter); |
| |
| GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, "avg_duration: %" GST_TIME_FORMAT |
| ", avg_pt: %" GST_TIME_FORMAT ", avg_rate: %g", |
| GST_TIME_ARGS (priv->avg_duration), GST_TIME_ARGS (priv->avg_pt), |
| priv->avg_rate); |
| |
| /* collect running averages. for first observations, we copy the |
| * values */ |
| if (priv->avg_duration == -1) |
| priv->avg_duration = duration; |
| else |
| priv->avg_duration = UPDATE_RUNNING_AVG (priv->avg_duration, duration); |
| |
| if (priv->avg_pt == -1) |
| priv->avg_pt = pt; |
| else |
| priv->avg_pt = UPDATE_RUNNING_AVG (priv->avg_pt, pt); |
| |
| if (priv->avg_duration != 0) |
| rate = |
| gst_guint64_to_gdouble (priv->avg_pt) / |
| gst_guint64_to_gdouble (priv->avg_duration); |
| else |
| rate = 0.0; |
| |
| if (priv->last_left != -1) { |
| if (dropped || priv->avg_rate < 0.0) { |
| priv->avg_rate = rate; |
| } else { |
| if (rate > 1.0) |
| priv->avg_rate = UPDATE_RUNNING_AVG_N (priv->avg_rate, rate); |
| else |
| priv->avg_rate = UPDATE_RUNNING_AVG_P (priv->avg_rate, rate); |
| } |
| } |
| |
| GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, sink, |
| "updated: avg_duration: %" GST_TIME_FORMAT ", avg_pt: %" GST_TIME_FORMAT |
| ", avg_rate: %g", GST_TIME_ARGS (priv->avg_duration), |
| GST_TIME_ARGS (priv->avg_pt), priv->avg_rate); |
| |
| |
| if (priv->avg_rate >= 0.0) { |
| /* if we have a valid rate, start sending QoS messages */ |
| if (priv->current_jitter < 0) { |
| /* make sure we never go below 0 when adding the jitter to the |
| * timestamp. */ |
| if (priv->current_rstart < -priv->current_jitter) |
| priv->current_jitter = -priv->current_rstart; |
| } |
| gst_base_sink_send_qos (sink, priv->avg_rate, priv->current_rstart, |
| priv->current_jitter); |
| } |
| |
| /* record when this buffer will leave us */ |
| priv->last_left = left; |
| } |
| |
| /* reset all qos measuring */ |
| static void |
| gst_base_sink_reset_qos (GstBaseSink * sink) |
| { |
| GstBaseSinkPrivate *priv; |
| |
| priv = sink->priv; |
| |
| priv->last_in_time = -1; |
| priv->last_left = -1; |
| priv->avg_duration = -1; |
| priv->avg_pt = -1; |
| priv->avg_rate = -1.0; |
| priv->avg_render = -1; |
| priv->rendered = 0; |
| priv->dropped = 0; |
| |
| } |
| |
| /* Checks if the object was scheduled too late. |
| * |
| * start/stop contain the raw timestamp start and stop values |
| * of the object. |
| * |
| * status and jitter contain the return values from the clock wait. |
| * |
| * returns TRUE if the buffer was too late. |
| */ |
| static gboolean |
| gst_base_sink_is_too_late (GstBaseSink * basesink, GstMiniObject * obj, |
| GstClockTime start, GstClockTime stop, |
| GstClockReturn status, GstClockTimeDiff jitter) |
| { |
| gboolean late; |
| gint64 max_lateness; |
| GstBaseSinkPrivate *priv; |
| |
| priv = basesink->priv; |
| |
| late = FALSE; |
| |
| /* only for objects that were too late */ |
| if (G_LIKELY (status != GST_CLOCK_EARLY)) |
| goto in_time; |
| |
| max_lateness = basesink->abidata.ABI.max_lateness; |
| |
| /* check if frame dropping is enabled */ |
| if (max_lateness == -1) |
| goto no_drop; |
| |
| /* only check for buffers */ |
| if (G_UNLIKELY (!GST_IS_BUFFER (obj))) |
| goto not_buffer; |
| |
| /* can't do check if we don't have a timestamp */ |
| if (G_UNLIKELY (start == -1)) |
| goto no_timestamp; |
| |
| /* we can add a valid stop time */ |
| if (stop != -1) |
| max_lateness += stop; |
| else |
| max_lateness += start; |
| |
| /* if the jitter bigger than duration and lateness we are too late */ |
| if ((late = start + jitter > max_lateness)) { |
| GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, basesink, |
| "buffer is too late %" GST_TIME_FORMAT |
| " > %" GST_TIME_FORMAT, GST_TIME_ARGS (start + jitter), |
| GST_TIME_ARGS (max_lateness)); |
| /* !!emergency!!, if we did not receive anything valid for more than a |
| * second, render it anyway so the user sees something */ |
| if (priv->last_in_time != -1 && start - priv->last_in_time > GST_SECOND) { |
| late = FALSE; |
| GST_ELEMENT_WARNING (basesink, CORE, CLOCK, |
| (_("A lot of buffers are being dropped.")), |
| ("There may be a timestamping problem, or this computer is too slow.")); |
| GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, basesink, |
| "**emergency** last buffer at %" GST_TIME_FORMAT " > GST_SECOND", |
| GST_TIME_ARGS (priv->last_in_time)); |
| } |
| } |
| |
| done: |
| if (!late) { |
| priv->last_in_time = start; |
| } |
| return late; |
| |
| /* all is fine */ |
| in_time: |
| { |
| GST_DEBUG_OBJECT (basesink, "object was scheduled in time"); |
| goto done; |
| } |
| no_drop: |
| { |
| GST_DEBUG_OBJECT (basesink, "frame dropping disabled"); |
| goto done; |
| } |
| not_buffer: |
| { |
| GST_DEBUG_OBJECT (basesink, "object is not a buffer"); |
| return FALSE; |
| } |
| no_timestamp: |
| { |
| GST_DEBUG_OBJECT (basesink, "buffer has no timestamp"); |
| return FALSE; |
| } |
| } |
| |
| /* called before and after calling the render vmethod. It keeps track of how |
| * much time was spent in the render method and is used to check if we are |
| * flooded */ |
| static void |
| gst_base_sink_do_render_stats (GstBaseSink * basesink, gboolean start) |
| { |
| GstBaseSinkPrivate *priv; |
| |
| priv = basesink->priv; |
| |
| if (start) { |
| priv->start = gst_util_get_timestamp (); |
| } else { |
| GstClockTime elapsed; |
| |
| priv->stop = gst_util_get_timestamp (); |
| |
| elapsed = GST_CLOCK_DIFF (priv->start, priv->stop); |
| |
| if (priv->avg_render == -1) |
| priv->avg_render = elapsed; |
| else |
| priv->avg_render = UPDATE_RUNNING_AVG (priv->avg_render, elapsed); |
| |
| GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, basesink, |
| "avg_render: %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->avg_render)); |
| } |
| } |
| |
| /* with STREAM_LOCK, PREROLL_LOCK, |
| * |
| * Synchronize the object on the clock and then render it. |
| * |
| * takes ownership of obj. |
| */ |
| static GstFlowReturn |
| gst_base_sink_render_object (GstBaseSink * basesink, GstPad * pad, |
| gboolean is_list, gpointer obj) |
| { |
| GstFlowReturn ret; |
| GstBaseSinkClass *bclass; |
| gboolean late, step_end; |
| gpointer sync_obj; |
| |
| GstBaseSinkPrivate *priv; |
| |
| priv = basesink->priv; |
| |
| if (is_list) { |
| /* |
| * If buffer list, use the first group buffer within the list |
| * for syncing |
| */ |
| sync_obj = gst_buffer_list_get (GST_BUFFER_LIST_CAST (obj), 0, 0); |
| g_assert (NULL != sync_obj); |
| } else { |
| sync_obj = obj; |
| } |
| |
| again: |
| late = FALSE; |
| step_end = FALSE; |
| |
| /* synchronize this object, non syncable objects return OK |
| * immediatly. */ |
| ret = gst_base_sink_do_sync (basesink, pad, sync_obj, &late, &step_end); |
| if (G_UNLIKELY (ret != GST_FLOW_OK)) |
| goto sync_failed; |
| |
| /* and now render, event or buffer/buffer list. */ |
| if (G_LIKELY (is_list || GST_IS_BUFFER (obj))) { |
| /* drop late buffers unconditionally, let's hope it's unlikely */ |
| if (G_UNLIKELY (late)) |
| goto dropped; |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| if (G_LIKELY ((is_list && bclass->render_list) || |
| (!is_list && bclass->render))) { |
| gint do_qos; |
| |
| /* read once, to get same value before and after */ |
| do_qos = g_atomic_int_get (&priv->qos_enabled); |
| |
| GST_DEBUG_OBJECT (basesink, "rendering object %p", obj); |
| |
| /* record rendering time for QoS and stats */ |
| if (do_qos) |
| gst_base_sink_do_render_stats (basesink, TRUE); |
| |
| if (!is_list) { |
| GstBuffer *buf; |
| |
| /* For buffer lists do not set last buffer. Creating buffer |
| * with meaningful data can be done only with memcpy which will |
| * significantly affect performance */ |
| buf = GST_BUFFER_CAST (obj); |
| gst_base_sink_set_last_buffer (basesink, buf); |
| |
| ret = bclass->render (basesink, buf); |
| } else { |
| GstBufferList *buflist; |
| |
| buflist = GST_BUFFER_LIST_CAST (obj); |
| |
| ret = bclass->render_list (basesink, buflist); |
| } |
| |
| if (do_qos) |
| gst_base_sink_do_render_stats (basesink, FALSE); |
| |
| if (ret == GST_FLOW_STEP) |
| goto again; |
| |
| if (G_UNLIKELY (basesink->flushing)) |
| goto flushing; |
| |
| priv->rendered++; |
| } |
| } else { |
| GstEvent *event = GST_EVENT_CAST (obj); |
| gboolean event_res = TRUE; |
| GstEventType type; |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| type = GST_EVENT_TYPE (event); |
| |
| GST_DEBUG_OBJECT (basesink, "rendering event %p, type %s", obj, |
| gst_event_type_get_name (type)); |
| |
| if (bclass->event) |
| event_res = bclass->event (basesink, event); |
| |
| /* when we get here we could be flushing again when the event handler calls |
| * _wait_eos(). We have to ignore this object in that case. */ |
| if (G_UNLIKELY (basesink->flushing)) |
| goto flushing; |
| |
| if (G_LIKELY (event_res)) { |
| guint32 seqnum; |
| |
| seqnum = basesink->priv->seqnum = gst_event_get_seqnum (event); |
| GST_DEBUG_OBJECT (basesink, "Got seqnum #%" G_GUINT32_FORMAT, seqnum); |
| |
| switch (type) { |
| case GST_EVENT_EOS: |
| { |
| GstMessage *message; |
| |
| /* the EOS event is completely handled so we mark |
| * ourselves as being in the EOS state. eos is also |
| * protected by the object lock so we can read it when |
| * answering the POSITION query. */ |
| GST_OBJECT_LOCK (basesink); |
| basesink->eos = TRUE; |
| GST_OBJECT_UNLOCK (basesink); |
| |
| /* ok, now we can post the message */ |
| GST_DEBUG_OBJECT (basesink, "Now posting EOS"); |
| |
| message = gst_message_new_eos (GST_OBJECT_CAST (basesink)); |
| gst_message_set_seqnum (message, seqnum); |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), message); |
| break; |
| } |
| case GST_EVENT_NEWSEGMENT: |
| /* configure the segment */ |
| gst_base_sink_configure_segment (basesink, pad, event, |
| &basesink->segment); |
| break; |
| default: |
| break; |
| } |
| } |
| } |
| |
| done: |
| if (step_end) { |
| /* the step ended, check if we need to activate a new step */ |
| GST_DEBUG_OBJECT (basesink, "step ended"); |
| stop_stepping (basesink, &basesink->segment, &priv->current_step, |
| priv->current_rstart, priv->current_rstop, basesink->eos); |
| goto again; |
| } |
| |
| gst_base_sink_perform_qos (basesink, late); |
| |
| GST_DEBUG_OBJECT (basesink, "object unref after render %p", obj); |
| gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); |
| return ret; |
| |
| /* ERRORS */ |
| sync_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, "do_sync returned %s", gst_flow_get_name (ret)); |
| goto done; |
| } |
| dropped: |
| { |
| priv->dropped++; |
| GST_DEBUG_OBJECT (basesink, "buffer late, dropping"); |
| goto done; |
| } |
| flushing: |
| { |
| GST_DEBUG_OBJECT (basesink, "we are flushing, ignore object"); |
| gst_mini_object_unref (obj); |
| return GST_FLOW_WRONG_STATE; |
| } |
| } |
| |
| /* with STREAM_LOCK, PREROLL_LOCK |
| * |
| * Perform preroll on the given object. For buffers this means |
| * calling the preroll subclass method. |
| * If that succeeds, the state will be commited. |
| * |
| * function does not take ownership of obj. |
| */ |
| static GstFlowReturn |
| gst_base_sink_preroll_object (GstBaseSink * basesink, gboolean is_list, |
| GstMiniObject * obj) |
| { |
| GstFlowReturn ret; |
| |
| GST_DEBUG_OBJECT (basesink, "prerolling object %p", obj); |
| |
| /* if it's a buffer, we need to call the preroll method */ |
| if (G_LIKELY (is_list || GST_IS_BUFFER (obj)) && basesink->priv->call_preroll) { |
| GstBaseSinkClass *bclass; |
| GstBuffer *buf; |
| GstClockTime timestamp; |
| |
| if (is_list) { |
| buf = gst_buffer_list_get (GST_BUFFER_LIST_CAST (obj), 0, 0); |
| g_assert (NULL != buf); |
| } else { |
| buf = GST_BUFFER_CAST (obj); |
| } |
| |
| timestamp = GST_BUFFER_TIMESTAMP (buf); |
| |
| GST_DEBUG_OBJECT (basesink, "preroll buffer %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (timestamp)); |
| |
| /* |
| * For buffer lists do not set last buffer. Creating buffer |
| * with meaningful data can be done only with memcpy which will |
| * significantly affect performance |
| */ |
| if (!is_list) { |
| gst_base_sink_set_last_buffer (basesink, buf); |
| } |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| if (bclass->preroll) |
| if ((ret = bclass->preroll (basesink, buf)) != GST_FLOW_OK) |
| goto preroll_failed; |
| |
| basesink->priv->call_preroll = FALSE; |
| } |
| |
| /* commit state */ |
| if (G_LIKELY (basesink->playing_async)) { |
| if (G_UNLIKELY (!gst_base_sink_commit_state (basesink))) |
| goto stopping; |
| } |
| |
| return GST_FLOW_OK; |
| |
| /* ERRORS */ |
| preroll_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, "preroll failed, abort state"); |
| gst_element_abort_state (GST_ELEMENT_CAST (basesink)); |
| return ret; |
| } |
| stopping: |
| { |
| GST_DEBUG_OBJECT (basesink, "stopping while commiting state"); |
| return GST_FLOW_WRONG_STATE; |
| } |
| } |
| |
| /* with STREAM_LOCK, PREROLL_LOCK |
| * |
| * Queue an object for rendering. |
| * The first prerollable object queued will complete the preroll. If the |
| * preroll queue if filled, we render all the objects in the queue. |
| * |
| * This function takes ownership of the object. |
| */ |
| static GstFlowReturn |
| gst_base_sink_queue_object_unlocked (GstBaseSink * basesink, GstPad * pad, |
| gboolean is_list, gpointer obj, gboolean prerollable) |
| { |
| GstFlowReturn ret = GST_FLOW_OK; |
| gint length; |
| GQueue *q; |
| |
| if (G_UNLIKELY (basesink->need_preroll)) { |
| if (G_LIKELY (prerollable)) |
| basesink->preroll_queued++; |
| |
| length = basesink->preroll_queued; |
| |
| GST_DEBUG_OBJECT (basesink, "now %d prerolled items", length); |
| |
| /* first prerollable item needs to finish the preroll */ |
| if (length == 1) { |
| ret = gst_base_sink_preroll_object (basesink, is_list, obj); |
| if (G_UNLIKELY (ret != GST_FLOW_OK)) |
| goto preroll_failed; |
| } |
| /* need to recheck if we need preroll, commmit state during preroll |
| * could have made us not need more preroll. */ |
| if (G_UNLIKELY (basesink->need_preroll)) { |
| /* see if we can render now, if we can't add the object to the preroll |
| * queue. */ |
| if (G_UNLIKELY (length <= basesink->preroll_queue_max_len)) |
| goto more_preroll; |
| } |
| } |
| /* we can start rendering (or blocking) the queued object |
| * if any. */ |
| q = basesink->preroll_queue; |
| while (G_UNLIKELY (!g_queue_is_empty (q))) { |
| GstMiniObject *o; |
| |
| o = g_queue_pop_head (q); |
| GST_DEBUG_OBJECT (basesink, "rendering queued object %p", o); |
| |
| /* do something with the return value */ |
| ret = gst_base_sink_render_object (basesink, pad, FALSE, o); |
| if (ret != GST_FLOW_OK) |
| goto dequeue_failed; |
| } |
| |
| /* now render the object */ |
| ret = gst_base_sink_render_object (basesink, pad, is_list, obj); |
| basesink->preroll_queued = 0; |
| |
| return ret; |
| |
| /* special cases */ |
| preroll_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, "preroll failed, reason %s", |
| gst_flow_get_name (ret)); |
| gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); |
| return ret; |
| } |
| more_preroll: |
| { |
| /* add object to the queue and return */ |
| GST_DEBUG_OBJECT (basesink, "need more preroll data %d <= %d", |
| length, basesink->preroll_queue_max_len); |
| g_queue_push_tail (basesink->preroll_queue, obj); |
| return GST_FLOW_OK; |
| } |
| dequeue_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, "rendering queued objects failed, reason %s", |
| gst_flow_get_name (ret)); |
| gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); |
| return ret; |
| } |
| } |
| |
| /* with STREAM_LOCK |
| * |
| * This function grabs the PREROLL_LOCK and adds the object to |
| * the queue. |
| * |
| * This function takes ownership of obj. |
| */ |
| static GstFlowReturn |
| gst_base_sink_queue_object (GstBaseSink * basesink, GstPad * pad, |
| GstMiniObject * obj, gboolean prerollable) |
| { |
| GstFlowReturn ret; |
| |
| GST_PAD_PREROLL_LOCK (pad); |
| if (G_UNLIKELY (basesink->flushing)) |
| goto flushing; |
| |
| if (G_UNLIKELY (basesink->priv->received_eos)) |
| goto was_eos; |
| |
| ret = |
| gst_base_sink_queue_object_unlocked (basesink, pad, FALSE, obj, |
| prerollable); |
| GST_PAD_PREROLL_UNLOCK (pad); |
| |
| return ret; |
| |
| /* ERRORS */ |
| flushing: |
| { |
| GST_DEBUG_OBJECT (basesink, "sink is flushing"); |
| GST_PAD_PREROLL_UNLOCK (pad); |
| gst_mini_object_unref (obj); |
| return GST_FLOW_WRONG_STATE; |
| } |
| was_eos: |
| { |
| GST_DEBUG_OBJECT (basesink, |
| "we are EOS, dropping object, return UNEXPECTED"); |
| GST_PAD_PREROLL_UNLOCK (pad); |
| gst_mini_object_unref (obj); |
| return GST_FLOW_UNEXPECTED; |
| } |
| } |
| |
| static void |
| gst_base_sink_flush_start (GstBaseSink * basesink, GstPad * pad) |
| { |
| /* make sure we are not blocked on the clock also clear any pending |
| * eos state. */ |
| gst_base_sink_set_flushing (basesink, pad, TRUE); |
| |
| /* we grab the stream lock but that is not needed since setting the |
| * sink to flushing would make sure no state commit is being done |
| * anymore */ |
| GST_PAD_STREAM_LOCK (pad); |
| gst_base_sink_reset_qos (basesink); |
| if (basesink->priv->async_enabled) { |
| /* and we need to commit our state again on the next |
| * prerolled buffer */ |
| basesink->playing_async = TRUE; |
| gst_element_lost_state (GST_ELEMENT_CAST (basesink)); |
| } else { |
| basesink->priv->have_latency = TRUE; |
| basesink->need_preroll = FALSE; |
| } |
| gst_base_sink_set_last_buffer (basesink, NULL); |
| GST_PAD_STREAM_UNLOCK (pad); |
| } |
| |
| static void |
| gst_base_sink_flush_stop (GstBaseSink * basesink, GstPad * pad) |
| { |
| /* unset flushing so we can accept new data, this also flushes out any EOS |
| * event. */ |
| gst_base_sink_set_flushing (basesink, pad, FALSE); |
| |
| /* for position reporting */ |
| GST_OBJECT_LOCK (basesink); |
| basesink->priv->current_sstart = -1; |
| basesink->priv->current_sstop = -1; |
| basesink->priv->eos_rtime = -1; |
| basesink->priv->call_preroll = TRUE; |
| basesink->priv->current_step.valid = FALSE; |
| basesink->priv->pending_step.valid = FALSE; |
| if (basesink->pad_mode == GST_ACTIVATE_PUSH) { |
| /* we need new segment info after the flush. */ |
| basesink->have_newsegment = FALSE; |
| gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED); |
| gst_segment_init (basesink->abidata.ABI.clip_segment, GST_FORMAT_UNDEFINED); |
| } |
| GST_OBJECT_UNLOCK (basesink); |
| } |
| |
| static gboolean |
| gst_base_sink_event (GstPad * pad, GstEvent * event) |
| { |
| GstBaseSink *basesink; |
| gboolean result = TRUE; |
| GstBaseSinkClass *bclass; |
| |
| basesink = GST_BASE_SINK (gst_pad_get_parent (pad)); |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| GST_DEBUG_OBJECT (basesink, "event %p (%s)", event, |
| GST_EVENT_TYPE_NAME (event)); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| case GST_EVENT_EOS: |
| { |
| GstFlowReturn ret; |
| |
| GST_PAD_PREROLL_LOCK (pad); |
| if (G_UNLIKELY (basesink->flushing)) |
| goto flushing; |
| |
| if (G_UNLIKELY (basesink->priv->received_eos)) { |
| /* we can't accept anything when we are EOS */ |
| result = FALSE; |
| gst_event_unref (event); |
| } else { |
| /* we set the received EOS flag here so that we can use it when testing if |
| * we are prerolled and to refuse more buffers. */ |
| basesink->priv->received_eos = TRUE; |
| |
| /* EOS is a prerollable object, we call the unlocked version because it |
| * does not check the received_eos flag. */ |
| ret = gst_base_sink_queue_object_unlocked (basesink, pad, |
| FALSE, GST_MINI_OBJECT_CAST (event), TRUE); |
| if (G_UNLIKELY (ret != GST_FLOW_OK)) |
| result = FALSE; |
| } |
| GST_PAD_PREROLL_UNLOCK (pad); |
| break; |
| } |
| case GST_EVENT_NEWSEGMENT: |
| { |
| GstFlowReturn ret; |
| |
| GST_DEBUG_OBJECT (basesink, "newsegment %p", event); |
| |
| GST_PAD_PREROLL_LOCK (pad); |
| if (G_UNLIKELY (basesink->flushing)) |
| goto flushing; |
| |
| if (G_UNLIKELY (basesink->priv->received_eos)) { |
| /* we can't accept anything when we are EOS */ |
| result = FALSE; |
| gst_event_unref (event); |
| } else { |
| /* the new segment is a non prerollable item and does not block anything, |
| * we need to configure the current clipping segment and insert the event |
| * in the queue to serialize it with the buffers for rendering. */ |
| gst_base_sink_configure_segment (basesink, pad, event, |
| basesink->abidata.ABI.clip_segment); |
| |
| ret = |
| gst_base_sink_queue_object_unlocked (basesink, pad, |
| FALSE, GST_MINI_OBJECT_CAST (event), FALSE); |
| if (G_UNLIKELY (ret != GST_FLOW_OK)) |
| result = FALSE; |
| else { |
| GST_OBJECT_LOCK (basesink); |
| basesink->have_newsegment = TRUE; |
| GST_OBJECT_UNLOCK (basesink); |
| } |
| } |
| GST_PAD_PREROLL_UNLOCK (pad); |
| break; |
| } |
| case GST_EVENT_FLUSH_START: |
| if (bclass->event) |
| bclass->event (basesink, event); |
| |
| GST_DEBUG_OBJECT (basesink, "flush-start %p", event); |
| |
| gst_base_sink_flush_start (basesink, pad); |
| |
| gst_event_unref (event); |
| break; |
| case GST_EVENT_FLUSH_STOP: |
| if (bclass->event) |
| bclass->event (basesink, event); |
| |
| GST_DEBUG_OBJECT (basesink, "flush-stop %p", event); |
| |
| gst_base_sink_flush_stop (basesink, pad); |
| |
| gst_event_unref (event); |
| break; |
| default: |
| /* other events are sent to queue or subclass depending on if they |
| * are serialized. */ |
| if (GST_EVENT_IS_SERIALIZED (event)) { |
| gst_base_sink_queue_object (basesink, pad, |
| GST_MINI_OBJECT_CAST (event), FALSE); |
| } else { |
| if (bclass->event) |
| bclass->event (basesink, event); |
| gst_event_unref (event); |
| } |
| break; |
| } |
| done: |
| gst_object_unref (basesink); |
| |
| return result; |
| |
| /* ERRORS */ |
| flushing: |
| { |
| GST_DEBUG_OBJECT (basesink, "we are flushing"); |
| GST_PAD_PREROLL_UNLOCK (pad); |
| result = FALSE; |
| gst_event_unref (event); |
| goto done; |
| } |
| } |
| |
| /* default implementation to calculate the start and end |
| * timestamps on a buffer, subclasses can override |
| */ |
| static void |
| gst_base_sink_get_times (GstBaseSink * basesink, GstBuffer * buffer, |
| GstClockTime * start, GstClockTime * end) |
| { |
| GstClockTime timestamp, duration; |
| |
| timestamp = GST_BUFFER_TIMESTAMP (buffer); |
| if (GST_CLOCK_TIME_IS_VALID (timestamp)) { |
| |
| /* get duration to calculate end time */ |
| duration = GST_BUFFER_DURATION (buffer); |
| if (GST_CLOCK_TIME_IS_VALID (duration)) { |
| *end = timestamp + duration; |
| } |
| *start = timestamp; |
| } |
| } |
| |
| /* must be called with PREROLL_LOCK */ |
| static gboolean |
| gst_base_sink_needs_preroll (GstBaseSink * basesink) |
| { |
| gboolean is_prerolled, res; |
| |
| /* we have 2 cases where the PREROLL_LOCK is released: |
| * 1) we are blocking in the PREROLL_LOCK and thus are prerolled. |
| * 2) we are syncing on the clock |
| */ |
| is_prerolled = basesink->have_preroll || basesink->priv->received_eos; |
| res = !is_prerolled; |
| |
| GST_DEBUG_OBJECT (basesink, "have_preroll: %d, EOS: %d => needs preroll: %d", |
| basesink->have_preroll, basesink->priv->received_eos, res); |
| |
| return res; |
| } |
| |
| /* with STREAM_LOCK, PREROLL_LOCK |
| * |
| * Takes a buffer and compare the timestamps with the last segment. |
| * If the buffer falls outside of the segment boundaries, drop it. |
| * Else queue the buffer for preroll and rendering. |
| * |
| * This function takes ownership of the buffer. |
| */ |
| static GstFlowReturn |
| gst_base_sink_chain_unlocked (GstBaseSink * basesink, GstPad * pad, |
| gboolean is_list, gpointer obj) |
| { |
| GstBaseSinkClass *bclass; |
| GstFlowReturn result; |
| GstClockTime start = GST_CLOCK_TIME_NONE, end = GST_CLOCK_TIME_NONE; |
| GstSegment *clip_segment; |
| GstBuffer *time_buf; |
| |
| if (G_UNLIKELY (basesink->flushing)) |
| goto flushing; |
| |
| if (G_UNLIKELY (basesink->priv->received_eos)) |
| goto was_eos; |
| |
| if (is_list) { |
| time_buf = gst_buffer_list_get (GST_BUFFER_LIST_CAST (obj), 0, 0); |
| g_assert (NULL != time_buf); |
| } else { |
| time_buf = GST_BUFFER_CAST (obj); |
| } |
| |
| /* for code clarity */ |
| clip_segment = basesink->abidata.ABI.clip_segment; |
| |
| if (G_UNLIKELY (!basesink->have_newsegment)) { |
| gboolean sync; |
| |
| sync = gst_base_sink_get_sync (basesink); |
| if (sync) { |
| GST_ELEMENT_WARNING (basesink, STREAM, FAILED, |
| (_("Internal data flow problem.")), |
| ("Received buffer without a new-segment. Assuming timestamps start from 0.")); |
| } |
| |
| /* this means this sink will assume timestamps start from 0 */ |
| GST_OBJECT_LOCK (basesink); |
| clip_segment->start = 0; |
| clip_segment->stop = -1; |
| basesink->segment.start = 0; |
| basesink->segment.stop = -1; |
| basesink->have_newsegment = TRUE; |
| GST_OBJECT_UNLOCK (basesink); |
| } |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| /* check if the buffer needs to be dropped, we first ask the subclass for the |
| * start and end */ |
| if (bclass->get_times) |
| bclass->get_times (basesink, time_buf, &start, &end); |
| |
| if (start == -1) { |
| /* if the subclass does not want sync, we use our own values so that we at |
| * least clip the buffer to the segment */ |
| gst_base_sink_get_times (basesink, time_buf, &start, &end); |
| } |
| |
| GST_DEBUG_OBJECT (basesink, "got times start: %" GST_TIME_FORMAT |
| ", end: %" GST_TIME_FORMAT, GST_TIME_ARGS (start), GST_TIME_ARGS (end)); |
| |
| /* a dropped buffer does not participate in anything */ |
| if (GST_CLOCK_TIME_IS_VALID (start) && |
| (clip_segment->format == GST_FORMAT_TIME)) { |
| if (G_UNLIKELY (!gst_segment_clip (clip_segment, |
| GST_FORMAT_TIME, (gint64) start, (gint64) end, NULL, NULL))) |
| goto out_of_segment; |
| } |
| |
| /* now we can process the buffer in the queue, this function takes ownership |
| * of the buffer */ |
| result = gst_base_sink_queue_object_unlocked (basesink, pad, |
| is_list, obj, TRUE); |
| return result; |
| |
| /* ERRORS */ |
| flushing: |
| { |
| GST_DEBUG_OBJECT (basesink, "sink is flushing"); |
| gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); |
| return GST_FLOW_WRONG_STATE; |
| } |
| was_eos: |
| { |
| GST_DEBUG_OBJECT (basesink, |
| "we are EOS, dropping object, return UNEXPECTED"); |
| gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); |
| return GST_FLOW_UNEXPECTED; |
| } |
| out_of_segment: |
| { |
| GST_DEBUG_OBJECT (basesink, "dropping buffer, out of clipping segment"); |
| gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); |
| return GST_FLOW_OK; |
| } |
| } |
| |
| /* with STREAM_LOCK |
| */ |
| static GstFlowReturn |
| gst_base_sink_chain_main (GstBaseSink * basesink, GstPad * pad, |
| gboolean is_list, gpointer obj) |
| { |
| GstFlowReturn result; |
| |
| if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PUSH)) |
| goto wrong_mode; |
| |
| GST_PAD_PREROLL_LOCK (pad); |
| result = gst_base_sink_chain_unlocked (basesink, pad, is_list, obj); |
| GST_PAD_PREROLL_UNLOCK (pad); |
| |
| done: |
| return result; |
| |
| /* ERRORS */ |
| wrong_mode: |
| { |
| GST_OBJECT_LOCK (pad); |
| GST_WARNING_OBJECT (basesink, |
| "Push on pad %s:%s, but it was not activated in push mode", |
| GST_DEBUG_PAD_NAME (pad)); |
| GST_OBJECT_UNLOCK (pad); |
| gst_mini_object_unref (GST_MINI_OBJECT_CAST (obj)); |
| /* we don't post an error message this will signal to the peer |
| * pushing that EOS is reached. */ |
| result = GST_FLOW_UNEXPECTED; |
| goto done; |
| } |
| } |
| |
| static GstFlowReturn |
| gst_base_sink_chain (GstPad * pad, GstBuffer * buf) |
| { |
| GstBaseSink *basesink; |
| |
| basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad)); |
| |
| return gst_base_sink_chain_main (basesink, pad, FALSE, buf); |
| } |
| |
| static GstFlowReturn |
| gst_base_sink_chain_list (GstPad * pad, GstBufferList * list) |
| { |
| GstBaseSink *basesink; |
| GstBaseSinkClass *bclass; |
| GstFlowReturn result; |
| |
| basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad)); |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| if (G_LIKELY (bclass->render_list)) { |
| result = gst_base_sink_chain_main (basesink, pad, TRUE, list); |
| } else { |
| GstBufferListIterator *it; |
| GstBuffer *group; |
| |
| GST_INFO_OBJECT (pad, "chaining each group in list as a merged buffer"); |
| |
| it = gst_buffer_list_iterate (list); |
| |
| if (gst_buffer_list_iterator_next_group (it)) { |
| do { |
| group = gst_buffer_list_iterator_merge_group (it); |
| if (group == NULL) { |
| group = gst_buffer_new (); |
| GST_CAT_INFO_OBJECT (GST_CAT_SCHEDULING, pad, "chaining empty group"); |
| } else { |
| GST_CAT_INFO_OBJECT (GST_CAT_SCHEDULING, pad, "chaining group"); |
| } |
| result = gst_base_sink_chain_main (basesink, pad, FALSE, group); |
| } while (result == GST_FLOW_OK |
| && gst_buffer_list_iterator_next_group (it)); |
| } else { |
| GST_CAT_INFO_OBJECT (GST_CAT_SCHEDULING, pad, "chaining empty group"); |
| result = |
| gst_base_sink_chain_main (basesink, pad, FALSE, gst_buffer_new ()); |
| } |
| gst_buffer_list_iterator_free (it); |
| gst_buffer_list_unref (list); |
| } |
| return result; |
| } |
| |
| |
| static gboolean |
| gst_base_sink_default_do_seek (GstBaseSink * sink, GstSegment * segment) |
| { |
| gboolean res = TRUE; |
| |
| /* update our offset if the start/stop position was updated */ |
| if (segment->format == GST_FORMAT_BYTES) { |
| segment->time = segment->start; |
| } else if (segment->start == 0) { |
| /* seek to start, we can implement a default for this. */ |
| segment->time = 0; |
| } else { |
| res = FALSE; |
| GST_INFO_OBJECT (sink, "Can't do a default seek"); |
| } |
| |
| return res; |
| } |
| |
| #define SEEK_TYPE_IS_RELATIVE(t) (((t) != GST_SEEK_TYPE_NONE) && ((t) != GST_SEEK_TYPE_SET)) |
| |
| static gboolean |
| gst_base_sink_default_prepare_seek_segment (GstBaseSink * sink, |
| GstEvent * event, GstSegment * segment) |
| { |
| /* By default, we try one of 2 things: |
| * - For absolute seek positions, convert the requested position to our |
| * configured processing format and place it in the output segment \ |
| * - For relative seek positions, convert our current (input) values to the |
| * seek format, adjust by the relative seek offset and then convert back to |
| * the processing format |
| */ |
| GstSeekType cur_type, stop_type; |
| gint64 cur, stop; |
| GstSeekFlags flags; |
| GstFormat seek_format, dest_format; |
| gdouble rate; |
| gboolean update; |
| gboolean res = TRUE; |
| |
| gst_event_parse_seek (event, &rate, &seek_format, &flags, |
| &cur_type, &cur, &stop_type, &stop); |
| dest_format = segment->format; |
| |
| if (seek_format == dest_format) { |
| gst_segment_set_seek (segment, rate, seek_format, flags, |
| cur_type, cur, stop_type, stop, &update); |
| return TRUE; |
| } |
| |
| if (cur_type != GST_SEEK_TYPE_NONE) { |
| /* FIXME: Handle seek_cur & seek_end by converting the input segment vals */ |
| res = |
| gst_pad_query_convert (sink->sinkpad, seek_format, cur, &dest_format, |
| &cur); |
| cur_type = GST_SEEK_TYPE_SET; |
| } |
| |
| if (res && stop_type != GST_SEEK_TYPE_NONE) { |
| /* FIXME: Handle seek_cur & seek_end by converting the input segment vals */ |
| res = |
| gst_pad_query_convert (sink->sinkpad, seek_format, stop, &dest_format, |
| &stop); |
| stop_type = GST_SEEK_TYPE_SET; |
| } |
| |
| /* And finally, configure our output segment in the desired format */ |
| gst_segment_set_seek (segment, rate, dest_format, flags, cur_type, cur, |
| stop_type, stop, &update); |
| |
| if (!res) |
| goto no_format; |
| |
| return res; |
| |
| no_format: |
| { |
| GST_DEBUG_OBJECT (sink, "undefined format given, seek aborted."); |
| return FALSE; |
| } |
| } |
| |
| /* perform a seek, only executed in pull mode */ |
| static gboolean |
| gst_base_sink_perform_seek (GstBaseSink * sink, GstPad * pad, GstEvent * event) |
| { |
| gboolean flush; |
| gdouble rate; |
| GstFormat seek_format, dest_format; |
| GstSeekFlags flags; |
| GstSeekType cur_type, stop_type; |
| gboolean seekseg_configured = FALSE; |
| gint64 cur, stop; |
| gboolean update, res = TRUE; |
| GstSegment seeksegment; |
| |
| dest_format = sink->segment.format; |
| |
| if (event) { |
| GST_DEBUG_OBJECT (sink, "performing seek with event %p", event); |
| gst_event_parse_seek (event, &rate, &seek_format, &flags, |
| &cur_type, &cur, &stop_type, &stop); |
| |
| flush = flags & GST_SEEK_FLAG_FLUSH; |
| } else { |
| GST_DEBUG_OBJECT (sink, "performing seek without event"); |
| flush = FALSE; |
| } |
| |
| if (flush) { |
| GST_DEBUG_OBJECT (sink, "flushing upstream"); |
| gst_pad_push_event (pad, gst_event_new_flush_start ()); |
| gst_base_sink_flush_start (sink, pad); |
| } else { |
| GST_DEBUG_OBJECT (sink, "pausing pulling thread"); |
| } |
| |
| GST_PAD_STREAM_LOCK (pad); |
| |
| /* If we configured the seeksegment above, don't overwrite it now. Otherwise |
| * copy the current segment info into the temp segment that we can actually |
| * attempt the seek with. We only update the real segment if the seek suceeds. */ |
| if (!seekseg_configured) { |
| memcpy (&seeksegment, &sink->segment, sizeof (GstSegment)); |
| |
| /* now configure the final seek segment */ |
| if (event) { |
| if (sink->segment.format != seek_format) { |
| /* OK, here's where we give the subclass a chance to convert the relative |
| * seek into an absolute one in the processing format. We set up any |
| * absolute seek above, before taking the stream lock. */ |
| if (!gst_base_sink_default_prepare_seek_segment (sink, event, |
| &seeksegment)) { |
| GST_DEBUG_OBJECT (sink, |
| "Preparing the seek failed after flushing. " "Aborting seek"); |
| res = FALSE; |
| } |
| } else { |
| /* The seek format matches our processing format, no need to ask the |
| * the subclass to configure the segment. */ |
| gst_segment_set_seek (&seeksegment, rate, seek_format, flags, |
| cur_type, cur, stop_type, stop, &update); |
| } |
| } |
| /* Else, no seek event passed, so we're just (re)starting the |
| current segment. */ |
| } |
| |
| if (res) { |
| GST_DEBUG_OBJECT (sink, "segment configured from %" G_GINT64_FORMAT |
| " to %" G_GINT64_FORMAT ", position %" G_GINT64_FORMAT, |
| seeksegment.start, seeksegment.stop, seeksegment.last_stop); |
| |
| /* do the seek, segment.last_stop contains the new position. */ |
| res = gst_base_sink_default_do_seek (sink, &seeksegment); |
| } |
| |
| |
| if (flush) { |
| GST_DEBUG_OBJECT (sink, "stop flushing upstream"); |
| gst_pad_push_event (pad, gst_event_new_flush_stop ()); |
| gst_base_sink_flush_stop (sink, pad); |
| } else if (res && sink->abidata.ABI.running) { |
| /* we are running the current segment and doing a non-flushing seek, |
| * close the segment first based on the last_stop. */ |
| GST_DEBUG_OBJECT (sink, "closing running segment %" G_GINT64_FORMAT |
| " to %" G_GINT64_FORMAT, sink->segment.start, sink->segment.last_stop); |
| } |
| |
| /* The subclass must have converted the segment to the processing format |
| * by now */ |
| if (res && seeksegment.format != dest_format) { |
| GST_DEBUG_OBJECT (sink, "Subclass failed to prepare a seek segment " |
| "in the correct format. Aborting seek."); |
| res = FALSE; |
| } |
| |
| /* if successfull seek, we update our real segment and push |
| * out the new segment. */ |
| if (res) { |
| memcpy (&sink->segment, &seeksegment, sizeof (GstSegment)); |
| |
| if (sink->segment.flags & GST_SEEK_FLAG_SEGMENT) { |
| gst_element_post_message (GST_ELEMENT (sink), |
| gst_message_new_segment_start (GST_OBJECT (sink), |
| sink->segment.format, sink->segment.last_stop)); |
| } |
| } |
| |
| sink->priv->discont = TRUE; |
| sink->abidata.ABI.running = TRUE; |
| |
| GST_PAD_STREAM_UNLOCK (pad); |
| |
| return res; |
| } |
| |
| static void |
| set_step_info (GstBaseSink * sink, GstStepInfo * current, GstStepInfo * pending, |
| guint seqnum, GstFormat format, guint64 amount, gdouble rate, |
| gboolean flush, gboolean intermediate) |
| { |
| GST_OBJECT_LOCK (sink); |
| pending->seqnum = seqnum; |
| pending->format = format; |
| pending->amount = amount; |
| pending->position = 0; |
| pending->rate = rate; |
| pending->flush = flush; |
| pending->intermediate = intermediate; |
| pending->valid = TRUE; |
| /* flush invalidates the current stepping segment */ |
| if (flush) |
| current->valid = FALSE; |
| GST_OBJECT_UNLOCK (sink); |
| } |
| |
| static gboolean |
| gst_base_sink_perform_step (GstBaseSink * sink, GstPad * pad, GstEvent * event) |
| { |
| GstBaseSinkPrivate *priv; |
| GstBaseSinkClass *bclass; |
| gboolean flush, intermediate; |
| gdouble rate; |
| GstFormat format; |
| guint64 amount; |
| guint seqnum; |
| GstStepInfo *pending, *current; |
| GstMessage *message; |
| |
| bclass = GST_BASE_SINK_GET_CLASS (sink); |
| priv = sink->priv; |
| |
| GST_DEBUG_OBJECT (sink, "performing step with event %p", event); |
| |
| gst_event_parse_step (event, &format, &amount, &rate, &flush, &intermediate); |
| seqnum = gst_event_get_seqnum (event); |
| |
| pending = &priv->pending_step; |
| current = &priv->current_step; |
| |
| /* post message first */ |
| message = gst_message_new_step_start (GST_OBJECT (sink), FALSE, format, |
| amount, rate, flush, intermediate); |
| gst_message_set_seqnum (message, seqnum); |
| gst_element_post_message (GST_ELEMENT (sink), message); |
| |
| if (flush) { |
| /* we need to call ::unlock before locking PREROLL_LOCK |
| * since we lock it before going into ::render */ |
| if (bclass->unlock) |
| bclass->unlock (sink); |
| |
| GST_PAD_PREROLL_LOCK (sink->sinkpad); |
| /* now that we have the PREROLL lock, clear our unlock request */ |
| if (bclass->unlock_stop) |
| bclass->unlock_stop (sink); |
| |
| /* update the stepinfo and make it valid */ |
| set_step_info (sink, current, pending, seqnum, format, amount, rate, flush, |
| intermediate); |
| |
| if (sink->priv->async_enabled) { |
| /* and we need to commit our state again on the next |
| * prerolled buffer */ |
| sink->playing_async = TRUE; |
| priv->pending_step.need_preroll = TRUE; |
| sink->need_preroll = FALSE; |
| gst_element_lost_state_full (GST_ELEMENT_CAST (sink), FALSE); |
| } else { |
| sink->priv->have_latency = TRUE; |
| sink->need_preroll = FALSE; |
| } |
| priv->current_sstart = -1; |
| priv->current_sstop = -1; |
| priv->eos_rtime = -1; |
| priv->call_preroll = TRUE; |
| gst_base_sink_set_last_buffer (sink, NULL); |
| gst_base_sink_reset_qos (sink); |
| |
| if (sink->clock_id) { |
| gst_clock_id_unschedule (sink->clock_id); |
| } |
| |
| if (sink->have_preroll) { |
| GST_DEBUG_OBJECT (sink, "signal waiter"); |
| priv->step_unlock = TRUE; |
| GST_PAD_PREROLL_SIGNAL (sink->sinkpad); |
| } |
| GST_PAD_PREROLL_UNLOCK (sink->sinkpad); |
| } else { |
| /* update the stepinfo and make it valid */ |
| set_step_info (sink, current, pending, seqnum, format, amount, rate, flush, |
| intermediate); |
| } |
| |
| return TRUE; |
| } |
| |
| /* with STREAM_LOCK |
| */ |
| static void |
| gst_base_sink_loop (GstPad * pad) |
| { |
| GstBaseSink *basesink; |
| GstBuffer *buf = NULL; |
| GstFlowReturn result; |
| guint blocksize; |
| guint64 offset; |
| |
| basesink = GST_BASE_SINK (GST_OBJECT_PARENT (pad)); |
| |
| g_assert (basesink->pad_mode == GST_ACTIVATE_PULL); |
| |
| if ((blocksize = basesink->priv->blocksize) == 0) |
| blocksize = -1; |
| |
| offset = basesink->segment.last_stop; |
| |
| GST_DEBUG_OBJECT (basesink, "pulling %" G_GUINT64_FORMAT ", %u", |
| offset, blocksize); |
| |
| result = gst_pad_pull_range (pad, offset, blocksize, &buf); |
| if (G_UNLIKELY (result != GST_FLOW_OK)) |
| goto paused; |
| |
| if (G_UNLIKELY (buf == NULL)) |
| goto no_buffer; |
| |
| offset += GST_BUFFER_SIZE (buf); |
| |
| gst_segment_set_last_stop (&basesink->segment, GST_FORMAT_BYTES, offset); |
| |
| GST_PAD_PREROLL_LOCK (pad); |
| result = gst_base_sink_chain_unlocked (basesink, pad, FALSE, buf); |
| GST_PAD_PREROLL_UNLOCK (pad); |
| if (G_UNLIKELY (result != GST_FLOW_OK)) |
| goto paused; |
| |
| return; |
| |
| /* ERRORS */ |
| paused: |
| { |
| GST_LOG_OBJECT (basesink, "pausing task, reason %s", |
| gst_flow_get_name (result)); |
| gst_pad_pause_task (pad); |
| /* fatal errors and NOT_LINKED cause EOS */ |
| if (GST_FLOW_IS_FATAL (result) || result == GST_FLOW_NOT_LINKED) { |
| if (result == GST_FLOW_UNEXPECTED) { |
| /* perform EOS logic */ |
| if (basesink->segment.flags & GST_SEEK_FLAG_SEGMENT) { |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_segment_done (GST_OBJECT_CAST (basesink), |
| basesink->segment.format, basesink->segment.last_stop)); |
| } else { |
| gst_base_sink_event (pad, gst_event_new_eos ()); |
| } |
| } else { |
| /* for fatal errors we post an error message, post the error |
| * first so the app knows about the error first. */ |
| GST_ELEMENT_ERROR (basesink, STREAM, FAILED, |
| (_("Internal data stream error.")), |
| ("stream stopped, reason %s", gst_flow_get_name (result))); |
| gst_base_sink_event (pad, gst_event_new_eos ()); |
| } |
| } |
| return; |
| } |
| no_buffer: |
| { |
| GST_LOG_OBJECT (basesink, "no buffer, pausing"); |
| GST_ELEMENT_ERROR (basesink, STREAM, FAILED, |
| (_("Internal data flow error.")), ("element returned NULL buffer")); |
| result = GST_FLOW_ERROR; |
| goto paused; |
| } |
| } |
| |
| static gboolean |
| gst_base_sink_set_flushing (GstBaseSink * basesink, GstPad * pad, |
| gboolean flushing) |
| { |
| GstBaseSinkClass *bclass; |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| if (flushing) { |
| /* unlock any subclasses, we need to do this before grabbing the |
| * PREROLL_LOCK since we hold this lock before going into ::render. */ |
| if (bclass->unlock) |
| bclass->unlock (basesink); |
| } |
| |
| GST_PAD_PREROLL_LOCK (pad); |
| basesink->flushing = flushing; |
| if (flushing) { |
| /* step 1, now that we have the PREROLL lock, clear our unlock request */ |
| if (bclass->unlock_stop) |
| bclass->unlock_stop (basesink); |
| |
| /* set need_preroll before we unblock the clock. If the clock is unblocked |
| * before timing out, we can reuse the buffer for preroll. */ |
| basesink->need_preroll = TRUE; |
| |
| /* step 2, unblock clock sync (if any) or any other blocking thing */ |
| if (basesink->clock_id) { |
| gst_clock_id_unschedule (basesink->clock_id); |
| } |
| |
| /* flush out the data thread if it's locked in finish_preroll, this will |
| * also flush out the EOS state */ |
| GST_DEBUG_OBJECT (basesink, |
| "flushing out data thread, need preroll to TRUE"); |
| gst_base_sink_preroll_queue_flush (basesink, pad); |
| } |
| GST_PAD_PREROLL_UNLOCK (pad); |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_base_sink_default_activate_pull (GstBaseSink * basesink, gboolean active) |
| { |
| gboolean result; |
| |
| if (active) { |
| /* start task */ |
| result = gst_pad_start_task (basesink->sinkpad, |
| (GstTaskFunction) gst_base_sink_loop, basesink->sinkpad); |
| } else { |
| /* step 2, make sure streaming finishes */ |
| result = gst_pad_stop_task (basesink->sinkpad); |
| } |
| |
| return result; |
| } |
| |
| static gboolean |
| gst_base_sink_pad_activate (GstPad * pad) |
| { |
| gboolean result = FALSE; |
| GstBaseSink *basesink; |
| |
| basesink = GST_BASE_SINK (gst_pad_get_parent (pad)); |
| |
| GST_DEBUG_OBJECT (basesink, "Trying pull mode first"); |
| |
| gst_base_sink_set_flushing (basesink, pad, FALSE); |
| |
| /* we need to have the pull mode enabled */ |
| if (!basesink->can_activate_pull) { |
| GST_DEBUG_OBJECT (basesink, "pull mode disabled"); |
| goto fallback; |
| } |
| |
| /* check if downstreams supports pull mode at all */ |
| if (!gst_pad_check_pull_range (pad)) { |
| GST_DEBUG_OBJECT (basesink, "pull mode not supported"); |
| goto fallback; |
| } |
| |
| /* set the pad mode before starting the task so that it's in the |
| * correct state for the new thread. also the sink set_caps and get_caps |
| * function checks this */ |
| basesink->pad_mode = GST_ACTIVATE_PULL; |
| |
| /* we first try to negotiate a format so that when we try to activate |
| * downstream, it knows about our format */ |
| if (!gst_base_sink_negotiate_pull (basesink)) { |
| GST_DEBUG_OBJECT (basesink, "failed to negotiate in pull mode"); |
| goto fallback; |
| } |
| |
| /* ok activate now */ |
| if (!gst_pad_activate_pull (pad, TRUE)) { |
| /* clear any pending caps */ |
| GST_OBJECT_LOCK (basesink); |
| gst_caps_replace (&basesink->priv->pull_caps, NULL); |
| GST_OBJECT_UNLOCK (basesink); |
| GST_DEBUG_OBJECT (basesink, "failed to activate in pull mode"); |
| goto fallback; |
| } |
| |
| GST_DEBUG_OBJECT (basesink, "Success activating pull mode"); |
| result = TRUE; |
| goto done; |
| |
| /* push mode fallback */ |
| fallback: |
| GST_DEBUG_OBJECT (basesink, "Falling back to push mode"); |
| if ((result = gst_pad_activate_push (pad, TRUE))) { |
| GST_DEBUG_OBJECT (basesink, "Success activating push mode"); |
| } |
| |
| done: |
| if (!result) { |
| GST_WARNING_OBJECT (basesink, "Could not activate pad in either mode"); |
| gst_base_sink_set_flushing (basesink, pad, TRUE); |
| } |
| |
| gst_object_unref (basesink); |
| |
| return result; |
| } |
| |
| static gboolean |
| gst_base_sink_pad_activate_push (GstPad * pad, gboolean active) |
| { |
| gboolean result; |
| GstBaseSink *basesink; |
| |
| basesink = GST_BASE_SINK (gst_pad_get_parent (pad)); |
| |
| if (active) { |
| if (!basesink->can_activate_push) { |
| result = FALSE; |
| basesink->pad_mode = GST_ACTIVATE_NONE; |
| } else { |
| result = TRUE; |
| basesink->pad_mode = GST_ACTIVATE_PUSH; |
| } |
| } else { |
| if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PUSH)) { |
| g_warning ("Internal GStreamer activation error!!!"); |
| result = FALSE; |
| } else { |
| gst_base_sink_set_flushing (basesink, pad, TRUE); |
| result = TRUE; |
| basesink->pad_mode = GST_ACTIVATE_NONE; |
| } |
| } |
| |
| gst_object_unref (basesink); |
| |
| return result; |
| } |
| |
| static gboolean |
| gst_base_sink_negotiate_pull (GstBaseSink * basesink) |
| { |
| GstCaps *caps; |
| gboolean result; |
| |
| result = FALSE; |
| |
| /* this returns the intersection between our caps and the peer caps. If there |
| * is no peer, it returns NULL and we can't operate in pull mode so we can |
| * fail the negotiation. */ |
| caps = gst_pad_get_allowed_caps (GST_BASE_SINK_PAD (basesink)); |
| if (caps == NULL || gst_caps_is_empty (caps)) |
| goto no_caps_possible; |
| |
| GST_DEBUG_OBJECT (basesink, "allowed caps: %" GST_PTR_FORMAT, caps); |
| |
| caps = gst_caps_make_writable (caps); |
| /* get the first (prefered) format */ |
| gst_caps_truncate (caps); |
| /* try to fixate */ |
| gst_pad_fixate_caps (GST_BASE_SINK_PAD (basesink), caps); |
| |
| GST_DEBUG_OBJECT (basesink, "fixated to: %" GST_PTR_FORMAT, caps); |
| |
| if (gst_caps_is_any (caps)) { |
| GST_DEBUG_OBJECT (basesink, "caps were ANY after fixating, " |
| "allowing pull()"); |
| /* neither side has template caps in this case, so they are prepared for |
| pull() without setcaps() */ |
| result = TRUE; |
| } else if (gst_caps_is_fixed (caps)) { |
| if (!gst_pad_set_caps (GST_BASE_SINK_PAD (basesink), caps)) |
| goto could_not_set_caps; |
| |
| GST_OBJECT_LOCK (basesink); |
| gst_caps_replace (&basesink->priv->pull_caps, caps); |
| GST_OBJECT_UNLOCK (basesink); |
| |
| result = TRUE; |
| } |
| |
| gst_caps_unref (caps); |
| |
| return result; |
| |
| no_caps_possible: |
| { |
| GST_INFO_OBJECT (basesink, "Pipeline could not agree on caps"); |
| GST_DEBUG_OBJECT (basesink, "get_allowed_caps() returned EMPTY"); |
| if (caps) |
| gst_caps_unref (caps); |
| return FALSE; |
| } |
| could_not_set_caps: |
| { |
| GST_INFO_OBJECT (basesink, "Could not set caps: %" GST_PTR_FORMAT, caps); |
| gst_caps_unref (caps); |
| return FALSE; |
| } |
| } |
| |
| /* this won't get called until we implement an activate function */ |
| static gboolean |
| gst_base_sink_pad_activate_pull (GstPad * pad, gboolean active) |
| { |
| gboolean result = FALSE; |
| GstBaseSink *basesink; |
| GstBaseSinkClass *bclass; |
| |
| basesink = GST_BASE_SINK (gst_pad_get_parent (pad)); |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| if (active) { |
| GstFormat format; |
| gint64 duration; |
| |
| /* we mark we have a newsegment here because pull based |
| * mode works just fine without having a newsegment before the |
| * first buffer */ |
| format = GST_FORMAT_BYTES; |
| |
| gst_segment_init (&basesink->segment, format); |
| gst_segment_init (basesink->abidata.ABI.clip_segment, format); |
| GST_OBJECT_LOCK (basesink); |
| basesink->have_newsegment = TRUE; |
| GST_OBJECT_UNLOCK (basesink); |
| |
| /* get the peer duration in bytes */ |
| result = gst_pad_query_peer_duration (pad, &format, &duration); |
| if (result) { |
| GST_DEBUG_OBJECT (basesink, |
| "setting duration in bytes to %" G_GINT64_FORMAT, duration); |
| gst_segment_set_duration (basesink->abidata.ABI.clip_segment, format, |
| duration); |
| gst_segment_set_duration (&basesink->segment, format, duration); |
| } else { |
| GST_DEBUG_OBJECT (basesink, "unknown duration"); |
| } |
| |
| if (bclass->activate_pull) |
| result = bclass->activate_pull (basesink, TRUE); |
| else |
| result = FALSE; |
| |
| if (!result) |
| goto activate_failed; |
| |
| } else { |
| if (G_UNLIKELY (basesink->pad_mode != GST_ACTIVATE_PULL)) { |
| g_warning ("Internal GStreamer activation error!!!"); |
| result = FALSE; |
| } else { |
| result = gst_base_sink_set_flushing (basesink, pad, TRUE); |
| if (bclass->activate_pull) |
| result &= bclass->activate_pull (basesink, FALSE); |
| basesink->pad_mode = GST_ACTIVATE_NONE; |
| /* clear any pending caps */ |
| GST_OBJECT_LOCK (basesink); |
| gst_caps_replace (&basesink->priv->pull_caps, NULL); |
| GST_OBJECT_UNLOCK (basesink); |
| } |
| } |
| gst_object_unref (basesink); |
| |
| return result; |
| |
| /* ERRORS */ |
| activate_failed: |
| { |
| /* reset, as starting the thread failed */ |
| basesink->pad_mode = GST_ACTIVATE_NONE; |
| |
| GST_ERROR_OBJECT (basesink, "subclass failed to activate in pull mode"); |
| return FALSE; |
| } |
| } |
| |
| /* send an event to our sinkpad peer. */ |
| static gboolean |
| gst_base_sink_send_event (GstElement * element, GstEvent * event) |
| { |
| GstPad *pad; |
| GstBaseSink *basesink = GST_BASE_SINK (element); |
| gboolean forward, result = TRUE; |
| GstActivateMode mode; |
| |
| GST_OBJECT_LOCK (element); |
| /* get the pad and the scheduling mode */ |
| pad = gst_object_ref (basesink->sinkpad); |
| mode = basesink->pad_mode; |
| GST_OBJECT_UNLOCK (element); |
| |
| /* only push UPSTREAM events upstream */ |
| forward = GST_EVENT_IS_UPSTREAM (event); |
| |
| switch (GST_EVENT_TYPE (event)) { |
| case GST_EVENT_LATENCY: |
| { |
| GstClockTime latency; |
| |
| gst_event_parse_latency (event, &latency); |
| |
| /* store the latency. We use this to adjust the running_time before syncing |
| * it to the clock. */ |
| GST_OBJECT_LOCK (element); |
| basesink->priv->latency = latency; |
| if (!basesink->priv->have_latency) |
| forward = FALSE; |
| GST_OBJECT_UNLOCK (element); |
| GST_DEBUG_OBJECT (basesink, "latency set to %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (latency)); |
| |
| /* We forward this event so that all elements know about the global pipeline |
| * latency. This is interesting for an element when it wants to figure out |
| * when a particular piece of data will be rendered. */ |
| break; |
| } |
| case GST_EVENT_SEEK: |
| /* in pull mode we will execute the seek */ |
| if (mode == GST_ACTIVATE_PULL) |
| result = gst_base_sink_perform_seek (basesink, pad, event); |
| break; |
| case GST_EVENT_STEP: |
| result = gst_base_sink_perform_step (basesink, pad, event); |
| forward = FALSE; |
| break; |
| default: |
| break; |
| } |
| |
| if (forward) { |
| result = gst_pad_push_event (pad, event); |
| } else { |
| /* not forwarded, unref the event */ |
| gst_event_unref (event); |
| } |
| |
| gst_object_unref (pad); |
| return result; |
| } |
| |
| static gboolean |
| gst_base_sink_peer_query (GstBaseSink * sink, GstQuery * query) |
| { |
| GstPad *peer; |
| gboolean res = FALSE; |
| |
| if ((peer = gst_pad_get_peer (sink->sinkpad))) { |
| res = gst_pad_query (peer, query); |
| gst_object_unref (peer); |
| } |
| return res; |
| } |
| |
| /* get the end position of the last seen object, this is used |
| * for EOS and for making sure that we don't report a position we |
| * have not reached yet. With LOCK. */ |
| static gboolean |
| gst_base_sink_get_position_last (GstBaseSink * basesink, GstFormat format, |
| gint64 * cur, gboolean * upstream) |
| { |
| GstFormat oformat; |
| GstSegment *segment; |
| gboolean ret = TRUE; |
| |
| segment = &basesink->segment; |
| oformat = segment->format; |
| |
| if (oformat == GST_FORMAT_TIME) { |
| /* return last observed stream time, we keep the stream time around in the |
| * time format. */ |
| *cur = basesink->priv->current_sstop; |
| } else { |
| /* convert last stop to stream time */ |
| *cur = gst_segment_to_stream_time (segment, oformat, segment->last_stop); |
| } |
| |
| if (*cur != -1 && oformat != format) { |
| GST_OBJECT_UNLOCK (basesink); |
| /* convert to the target format if we need to, release lock first */ |
| ret = |
| gst_pad_query_convert (basesink->sinkpad, oformat, *cur, &format, cur); |
| if (!ret) { |
| *cur = -1; |
| *upstream = TRUE; |
| } |
| GST_OBJECT_LOCK (basesink); |
| } |
| |
| GST_DEBUG_OBJECT (basesink, "POSITION: %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (*cur)); |
| |
| return ret; |
| } |
| |
| /* get the position when we are PAUSED, this is the stream time of the buffer |
| * that prerolled. If no buffer is prerolled (we are still flushing), this |
| * value will be -1. With LOCK. */ |
| static gboolean |
| gst_base_sink_get_position_paused (GstBaseSink * basesink, GstFormat format, |
| gint64 * cur, gboolean * upstream) |
| { |
| gboolean res; |
| gint64 time; |
| GstSegment *segment; |
| GstFormat oformat; |
| |
| /* we don't use the clip segment in pull mode, when seeking we update the |
| * main segment directly with the new segment values without it having to be |
| * activated by the rendering after preroll */ |
| if (basesink->pad_mode == GST_ACTIVATE_PUSH) |
| segment = basesink->abidata.ABI.clip_segment; |
| else |
| segment = &basesink->segment; |
| oformat = segment->format; |
| |
| if (oformat == GST_FORMAT_TIME) { |
| *cur = basesink->priv->current_sstart; |
| if (segment->rate < 0.0 && basesink->priv->current_sstop != -1) { |
| /* for reverse playback we prefer the stream time stop position if we have |
| * one */ |
| *cur = basesink->priv->current_sstop; |
| } |
| } else { |
| *cur = gst_segment_to_stream_time (segment, oformat, segment->last_stop); |
| } |
| |
| time = segment->time; |
| |
| if (*cur != -1) { |
| *cur = MAX (*cur, time); |
| GST_DEBUG_OBJECT (basesink, "POSITION as max: %" GST_TIME_FORMAT |
| ", time %" GST_TIME_FORMAT, GST_TIME_ARGS (*cur), GST_TIME_ARGS (time)); |
| } else { |
| /* we have no buffer, use the segment times. */ |
| if (segment->rate >= 0.0) { |
| /* forward, next position is always the time of the segment */ |
| *cur = time; |
| GST_DEBUG_OBJECT (basesink, "POSITION as time: %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (*cur)); |
| } else { |
| /* reverse, next expected timestamp is segment->stop. We use the function |
| * to get things right for negative applied_rates. */ |
| *cur = gst_segment_to_stream_time (segment, oformat, segment->stop); |
| GST_DEBUG_OBJECT (basesink, "reverse POSITION: %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (*cur)); |
| } |
| } |
| |
| res = (*cur != -1); |
| if (res && oformat != format) { |
| GST_OBJECT_UNLOCK (basesink); |
| res = |
| gst_pad_query_convert (basesink->sinkpad, oformat, *cur, &format, cur); |
| if (!res) { |
| *cur = -1; |
| *upstream = TRUE; |
| } |
| GST_OBJECT_LOCK (basesink); |
| } |
| |
| return res; |
| } |
| |
| static gboolean |
| gst_base_sink_get_position (GstBaseSink * basesink, GstFormat format, |
| gint64 * cur, gboolean * upstream) |
| { |
| GstClock *clock; |
| gboolean res = FALSE; |
| GstFormat oformat, tformat; |
| GstClockTime now, base, latency; |
| gint64 time, accum, duration; |
| gdouble rate; |
| gint64 last; |
| |
| GST_OBJECT_LOCK (basesink); |
| /* our intermediate time format */ |
| tformat = GST_FORMAT_TIME; |
| /* get the format in the segment */ |
| oformat = basesink->segment.format; |
| |
| /* can only give answer based on the clock if not EOS */ |
| if (G_UNLIKELY (basesink->eos)) |
| goto in_eos; |
| |
| /* we can only get the segment when we are not NULL or READY */ |
| if (!basesink->have_newsegment) |
| goto wrong_state; |
| |
| /* when not in PLAYING or when we're busy with a state change, we |
| * cannot read from the clock so we report time based on the |
| * last seen timestamp. */ |
| if (GST_STATE (basesink) != GST_STATE_PLAYING || |
| GST_STATE_PENDING (basesink) != GST_STATE_VOID_PENDING) |
| goto in_pause; |
| |
| /* we need to sync on the clock. */ |
| if (basesink->sync == FALSE) |
| goto no_sync; |
| |
| /* and we need a clock */ |
| if (G_UNLIKELY ((clock = GST_ELEMENT_CLOCK (basesink)) == NULL)) |
| goto no_sync; |
| |
| /* collect all data we need holding the lock */ |
| if (GST_CLOCK_TIME_IS_VALID (basesink->segment.time)) |
| time = basesink->segment.time; |
| else |
| time = 0; |
| |
| if (GST_CLOCK_TIME_IS_VALID (basesink->segment.stop)) |
| duration = basesink->segment.stop - basesink->segment.start; |
| else |
| duration = 0; |
| |
| base = GST_ELEMENT_CAST (basesink)->base_time; |
| accum = basesink->segment.accum; |
| rate = basesink->segment.rate * basesink->segment.applied_rate; |
| latency = basesink->priv->latency; |
| |
| gst_object_ref (clock); |
| |
| /* this function might release the LOCK */ |
| gst_base_sink_get_position_last (basesink, format, &last, upstream); |
| |
| /* need to release the object lock before we can get the time, |
| * a clock might take the LOCK of the provider, which could be |
| * a basesink subclass. */ |
| GST_OBJECT_UNLOCK (basesink); |
| |
| now = gst_clock_get_time (clock); |
| |
| if (oformat != tformat) { |
| /* convert accum, time and duration to time */ |
| if (!gst_pad_query_convert (basesink->sinkpad, oformat, accum, &tformat, |
| &accum)) |
| goto convert_failed; |
| if (!gst_pad_query_convert (basesink->sinkpad, oformat, duration, &tformat, |
| &duration)) |
| goto convert_failed; |
| if (!gst_pad_query_convert (basesink->sinkpad, oformat, time, &tformat, |
| &time)) |
| goto convert_failed; |
| } |
| |
| /* subtract base time and accumulated time from the clock time. |
| * Make sure we don't go negative. This is the current time in |
| * the segment which we need to scale with the combined |
| * rate and applied rate. */ |
| base += accum; |
| base += latency; |
| base = MIN (now, base); |
| |
| /* for negative rates we need to count back from from the segment |
| * duration. */ |
| if (rate < 0.0) |
| time += duration; |
| |
| *cur = time + gst_guint64_to_gdouble (now - base) * rate; |
| |
| /* never report more than last seen position */ |
| if (last != -1) |
| *cur = MIN (last, *cur); |
| |
| gst_object_unref (clock); |
| |
| GST_DEBUG_OBJECT (basesink, |
| "now %" GST_TIME_FORMAT " - base %" GST_TIME_FORMAT " - accum %" |
| GST_TIME_FORMAT " + time %" GST_TIME_FORMAT, |
| GST_TIME_ARGS (now), GST_TIME_ARGS (base), |
| GST_TIME_ARGS (accum), GST_TIME_ARGS (time)); |
| |
| if (oformat != format) { |
| /* convert time to final format */ |
| if (!gst_pad_query_convert (basesink->sinkpad, tformat, *cur, &format, cur)) |
| goto convert_failed; |
| } |
| |
| res = TRUE; |
| |
| done: |
| GST_DEBUG_OBJECT (basesink, "res: %d, POSITION: %" GST_TIME_FORMAT, |
| res, GST_TIME_ARGS (*cur)); |
| return res; |
| |
| /* special cases */ |
| in_eos: |
| { |
| GST_DEBUG_OBJECT (basesink, "position in EOS"); |
| res = gst_base_sink_get_position_last (basesink, format, cur, upstream); |
| GST_OBJECT_UNLOCK (basesink); |
| goto done; |
| } |
| in_pause: |
| { |
| GST_DEBUG_OBJECT (basesink, "position in PAUSED"); |
| res = gst_base_sink_get_position_paused (basesink, format, cur, upstream); |
| GST_OBJECT_UNLOCK (basesink); |
| goto done; |
| } |
| wrong_state: |
| { |
| /* in NULL or READY we always return FALSE and -1 */ |
| GST_DEBUG_OBJECT (basesink, "position in wrong state, return -1"); |
| res = FALSE; |
| *cur = -1; |
| GST_OBJECT_UNLOCK (basesink); |
| goto done; |
| } |
| no_sync: |
| { |
| /* report last seen timestamp if any, else ask upstream to answer */ |
| if ((*cur = basesink->priv->current_sstart) != -1) |
| res = TRUE; |
| else |
| *upstream = TRUE; |
| |
| GST_DEBUG_OBJECT (basesink, "no sync, res %d, POSITION %" GST_TIME_FORMAT, |
| res, GST_TIME_ARGS (*cur)); |
| GST_OBJECT_UNLOCK (basesink); |
| return res; |
| } |
| convert_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, "convert failed, try upstream"); |
| *upstream = TRUE; |
| return FALSE; |
| } |
| } |
| |
| static gboolean |
| gst_base_sink_query (GstElement * element, GstQuery * query) |
| { |
| gboolean res = FALSE; |
| |
| GstBaseSink *basesink = GST_BASE_SINK (element); |
| |
| switch (GST_QUERY_TYPE (query)) { |
| case GST_QUERY_POSITION: |
| { |
| gint64 cur = 0; |
| GstFormat format; |
| gboolean upstream = FALSE; |
| |
| gst_query_parse_position (query, &format, NULL); |
| |
| GST_DEBUG_OBJECT (basesink, "position format %d", format); |
| |
| /* first try to get the position based on the clock */ |
| if ((res = |
| gst_base_sink_get_position (basesink, format, &cur, &upstream))) { |
| gst_query_set_position (query, format, cur); |
| } else if (upstream) { |
| /* fallback to peer query */ |
| res = gst_base_sink_peer_query (basesink, query); |
| } |
| break; |
| } |
| case GST_QUERY_DURATION: |
| { |
| GstFormat format, uformat; |
| gint64 duration, uduration; |
| |
| gst_query_parse_duration (query, &format, NULL); |
| |
| GST_DEBUG_OBJECT (basesink, "duration query in format %s", |
| gst_format_get_name (format)); |
| |
| if (basesink->pad_mode == GST_ACTIVATE_PULL) { |
| uformat = GST_FORMAT_BYTES; |
| |
| /* get the duration in bytes, in pull mode that's all we are sure to |
| * know. We have to explicitly get this value from upstream instead of |
| * using our cached value because it might change. Duration caching |
| * should be done at a higher level. */ |
| res = gst_pad_query_peer_duration (basesink->sinkpad, &uformat, |
| &uduration); |
| if (res) { |
| gst_segment_set_duration (&basesink->segment, uformat, uduration); |
| if (format != uformat) { |
| /* convert to the requested format */ |
| res = gst_pad_query_convert (basesink->sinkpad, uformat, uduration, |
| &format, &duration); |
| } else { |
| duration = uduration; |
| } |
| if (res) { |
| /* set the result */ |
| gst_query_set_duration (query, format, duration); |
| } |
| } |
| } else { |
| /* in push mode we simply forward upstream */ |
| res = gst_base_sink_peer_query (basesink, query); |
| } |
| break; |
| } |
| case GST_QUERY_LATENCY: |
| { |
| gboolean live, us_live; |
| GstClockTime min, max; |
| |
| if ((res = gst_base_sink_query_latency (basesink, &live, &us_live, &min, |
| &max))) { |
| gst_query_set_latency (query, live, min, max); |
| } |
| break; |
| } |
| case GST_QUERY_JITTER: |
| break; |
| case GST_QUERY_RATE: |
| /* gst_query_set_rate (query, basesink->segment_rate); */ |
| res = TRUE; |
| break; |
| case GST_QUERY_SEGMENT: |
| { |
| /* FIXME, bring start/stop to stream time */ |
| gst_query_set_segment (query, basesink->segment.rate, |
| GST_FORMAT_TIME, basesink->segment.start, basesink->segment.stop); |
| break; |
| } |
| case GST_QUERY_SEEKING: |
| case GST_QUERY_CONVERT: |
| case GST_QUERY_FORMATS: |
| default: |
| res = gst_base_sink_peer_query (basesink, query); |
| break; |
| } |
| return res; |
| } |
| |
| static GstStateChangeReturn |
| gst_base_sink_change_state (GstElement * element, GstStateChange transition) |
| { |
| GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS; |
| GstBaseSink *basesink = GST_BASE_SINK (element); |
| GstBaseSinkClass *bclass; |
| GstBaseSinkPrivate *priv; |
| |
| priv = basesink->priv; |
| |
| bclass = GST_BASE_SINK_GET_CLASS (basesink); |
| |
| switch (transition) { |
| case GST_STATE_CHANGE_NULL_TO_READY: |
| if (bclass->start) |
| if (!bclass->start (basesink)) |
| goto start_failed; |
| break; |
| case GST_STATE_CHANGE_READY_TO_PAUSED: |
| /* need to complete preroll before this state change completes, there |
| * is no data flow in READY so we can safely assume we need to preroll. */ |
| GST_PAD_PREROLL_LOCK (basesink->sinkpad); |
| GST_DEBUG_OBJECT (basesink, "READY to PAUSED"); |
| basesink->have_newsegment = FALSE; |
| gst_segment_init (&basesink->segment, GST_FORMAT_UNDEFINED); |
| gst_segment_init (basesink->abidata.ABI.clip_segment, |
| GST_FORMAT_UNDEFINED); |
| basesink->offset = 0; |
| basesink->have_preroll = FALSE; |
| priv->step_unlock = FALSE; |
| basesink->need_preroll = TRUE; |
| basesink->playing_async = TRUE; |
| priv->current_sstart = -1; |
| priv->current_sstop = -1; |
| priv->eos_rtime = -1; |
| priv->latency = 0; |
| basesink->eos = FALSE; |
| priv->received_eos = FALSE; |
| gst_base_sink_reset_qos (basesink); |
| priv->commited = FALSE; |
| priv->call_preroll = TRUE; |
| priv->current_step.valid = FALSE; |
| priv->pending_step.valid = FALSE; |
| if (priv->async_enabled) { |
| GST_DEBUG_OBJECT (basesink, "doing async state change"); |
| /* when async enabled, post async-start message and return ASYNC from |
| * the state change function */ |
| ret = GST_STATE_CHANGE_ASYNC; |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_async_start (GST_OBJECT_CAST (basesink), FALSE)); |
| } else { |
| priv->have_latency = TRUE; |
| } |
| GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); |
| break; |
| case GST_STATE_CHANGE_PAUSED_TO_PLAYING: |
| GST_PAD_PREROLL_LOCK (basesink->sinkpad); |
| if (!gst_base_sink_needs_preroll (basesink)) { |
| GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, don't need preroll"); |
| /* no preroll needed anymore now. */ |
| basesink->playing_async = FALSE; |
| basesink->need_preroll = FALSE; |
| if (basesink->eos) { |
| GstMessage *message; |
| |
| /* need to post EOS message here */ |
| GST_DEBUG_OBJECT (basesink, "Now posting EOS"); |
| message = gst_message_new_eos (GST_OBJECT_CAST (basesink)); |
| gst_message_set_seqnum (message, basesink->priv->seqnum); |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), message); |
| } else { |
| GST_DEBUG_OBJECT (basesink, "signal preroll"); |
| GST_PAD_PREROLL_SIGNAL (basesink->sinkpad); |
| } |
| } else { |
| GST_DEBUG_OBJECT (basesink, "PAUSED to PLAYING, we are not prerolled"); |
| basesink->need_preroll = TRUE; |
| basesink->playing_async = TRUE; |
| priv->call_preroll = TRUE; |
| priv->commited = FALSE; |
| if (priv->async_enabled) { |
| GST_DEBUG_OBJECT (basesink, "doing async state change"); |
| ret = GST_STATE_CHANGE_ASYNC; |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_async_start (GST_OBJECT_CAST (basesink), FALSE)); |
| } |
| } |
| GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); |
| break; |
| default: |
| break; |
| } |
| |
| { |
| GstStateChangeReturn bret; |
| |
| bret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); |
| if (G_UNLIKELY (bret == GST_STATE_CHANGE_FAILURE)) |
| goto activate_failed; |
| } |
| |
| switch (transition) { |
| case GST_STATE_CHANGE_PLAYING_TO_PAUSED: |
| GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED"); |
| /* FIXME, make sure we cannot enter _render first */ |
| |
| /* we need to call ::unlock before locking PREROLL_LOCK |
| * since we lock it before going into ::render */ |
| if (bclass->unlock) |
| bclass->unlock (basesink); |
| |
| GST_PAD_PREROLL_LOCK (basesink->sinkpad); |
| /* now that we have the PREROLL lock, clear our unlock request */ |
| if (bclass->unlock_stop) |
| bclass->unlock_stop (basesink); |
| |
| /* we need preroll again and we set the flag before unlocking the clockid |
| * because if the clockid is unlocked before a current buffer expired, we |
| * can use that buffer to preroll with */ |
| basesink->need_preroll = TRUE; |
| |
| if (basesink->clock_id) { |
| gst_clock_id_unschedule (basesink->clock_id); |
| } |
| |
| /* if we don't have a preroll buffer we need to wait for a preroll and |
| * return ASYNC. */ |
| if (!gst_base_sink_needs_preroll (basesink)) { |
| GST_DEBUG_OBJECT (basesink, "PLAYING to PAUSED, we are prerolled"); |
| basesink->playing_async = FALSE; |
| } else { |
| if (GST_STATE_TARGET (GST_ELEMENT (basesink)) <= GST_STATE_READY) { |
| ret = GST_STATE_CHANGE_SUCCESS; |
| } else { |
| GST_DEBUG_OBJECT (basesink, |
| "PLAYING to PAUSED, we are not prerolled"); |
| basesink->playing_async = TRUE; |
| priv->commited = FALSE; |
| priv->call_preroll = TRUE; |
| if (priv->async_enabled) { |
| GST_DEBUG_OBJECT (basesink, "doing async state change"); |
| ret = GST_STATE_CHANGE_ASYNC; |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_async_start (GST_OBJECT_CAST (basesink), |
| FALSE)); |
| } |
| } |
| } |
| GST_DEBUG_OBJECT (basesink, "rendered: %" G_GUINT64_FORMAT |
| ", dropped: %" G_GUINT64_FORMAT, priv->rendered, priv->dropped); |
| |
| gst_base_sink_reset_qos (basesink); |
| GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); |
| break; |
| case GST_STATE_CHANGE_PAUSED_TO_READY: |
| GST_PAD_PREROLL_LOCK (basesink->sinkpad); |
| /* start by reseting our position state with the object lock so that the |
| * position query gets the right idea. We do this before we post the |
| * messages so that the message handlers pick this up. */ |
| GST_OBJECT_LOCK (basesink); |
| basesink->have_newsegment = FALSE; |
| priv->current_sstart = -1; |
| priv->current_sstop = -1; |
| priv->have_latency = FALSE; |
| GST_OBJECT_UNLOCK (basesink); |
| |
| gst_base_sink_set_last_buffer (basesink, NULL); |
| priv->call_preroll = FALSE; |
| |
| if (!priv->commited) { |
| if (priv->async_enabled) { |
| GST_DEBUG_OBJECT (basesink, "PAUSED to READY, posting async-done"); |
| |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_state_changed (GST_OBJECT_CAST (basesink), |
| GST_STATE_PLAYING, GST_STATE_PAUSED, GST_STATE_READY)); |
| |
| gst_element_post_message (GST_ELEMENT_CAST (basesink), |
| gst_message_new_async_done (GST_OBJECT_CAST (basesink))); |
| } |
| priv->commited = TRUE; |
| } else { |
| GST_DEBUG_OBJECT (basesink, "PAUSED to READY, don't need_preroll"); |
| } |
| GST_PAD_PREROLL_UNLOCK (basesink->sinkpad); |
| break; |
| case GST_STATE_CHANGE_READY_TO_NULL: |
| if (bclass->stop) { |
| if (!bclass->stop (basesink)) { |
| GST_WARNING_OBJECT (basesink, "failed to stop"); |
| } |
| } |
| gst_base_sink_set_last_buffer (basesink, NULL); |
| priv->call_preroll = FALSE; |
| break; |
| default: |
| break; |
| } |
| |
| return ret; |
| |
| /* ERRORS */ |
| start_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, "failed to start"); |
| return GST_STATE_CHANGE_FAILURE; |
| } |
| activate_failed: |
| { |
| GST_DEBUG_OBJECT (basesink, |
| "element failed to change states -- activation problem?"); |
| return GST_STATE_CHANGE_FAILURE; |
| } |
| } |