mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-06-14 12:09:30 +00:00
a3f8f036fe
An application that triggers a state transition from PLAYING to PAUSED needs to acquire the LIVE_LOCK. Consequently the LIVE_LOCK must not be taken while pushing anything on the pads because this operation might get blocked by something that cannot be unblocked without the application being able to proceed with the state transitions for other elements in the pipeline. This commit extends the previous behaviour where the live lock was released before pushing buffers (indirectly through the unlock before subclass->create) to now also include unlocking before pushing events. The issue was discovered in a case for WebRTC where the application tried to shut down a pipeline but an event originating from a video source element (based on basesrc) was in the process of being pushed down the pipeline when it got stuck on the STREAM_LOCK for the pad after the rtpgccbwe element. This lock in turn was held by the rtcpgccbwe element as it was in the process of pushing data down the pipeline but was stuck on the blocking probes installed on dtlssrtpenc to prevent data from flowing before dtls keys had been negotiated. What should have happened here is that the blocking probes should be removed, but that can only happen if the application may continue driving the state transitions. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6671>
4271 lines
124 KiB
C
4271 lines
124 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2000,2005 Wim Taymans <wim@fluendo.com>
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*
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* gstbasesrc.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstbasesrc
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* @title: GstBaseSrc
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* @short_description: Base class for getrange based source elements
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* @see_also: #GstPushSrc, #GstBaseTransform, #GstBaseSink
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*
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* This is a generic base class for source elements. The following
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* types of sources are supported:
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*
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* * random access sources like files
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* * seekable sources
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* * live sources
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*
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* The source can be configured to operate in any #GstFormat with the
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* gst_base_src_set_format() method. The currently set format determines
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* the format of the internal #GstSegment and any %GST_EVENT_SEGMENT
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* events. The default format for #GstBaseSrc is %GST_FORMAT_BYTES.
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*
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* #GstBaseSrc always supports push mode scheduling. If the following
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* conditions are met, it also supports pull mode scheduling:
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*
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* * The format is set to %GST_FORMAT_BYTES (default).
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* * #GstBaseSrcClass::is_seekable returns %TRUE.
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*
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* If all the conditions are met for operating in pull mode, #GstBaseSrc is
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* automatically seekable in push mode as well. The following conditions must
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* be met to make the element seekable in push mode when the format is not
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* %GST_FORMAT_BYTES:
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*
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* * #GstBaseSrcClass::is_seekable returns %TRUE.
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* * #GstBaseSrcClass::query can convert all supported seek formats to the
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* internal format as set with gst_base_src_set_format().
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* * #GstBaseSrcClass::do_seek is implemented, performs the seek and returns
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* %TRUE.
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*
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* When the element does not meet the requirements to operate in pull mode, the
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* offset and length in the #GstBaseSrcClass::create method should be ignored.
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* It is recommended to subclass #GstPushSrc instead, in this situation. If the
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* element can operate in pull mode but only with specific offsets and
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* lengths, it is allowed to generate an error when the wrong values are passed
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* to the #GstBaseSrcClass::create function.
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*
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* #GstBaseSrc has support for live sources. Live sources are sources that when
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* paused discard data, such as audio or video capture devices. A typical live
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* source also produces data at a fixed rate and thus provides a clock to publish
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* this rate.
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* Use gst_base_src_set_live() to activate the live source mode.
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*
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* A live source does not produce data in the PAUSED state. This means that the
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* #GstBaseSrcClass::create method will not be called in PAUSED but only in
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* PLAYING. To signal the pipeline that the element will not produce data, the
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* return value from the READY to PAUSED state will be
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* %GST_STATE_CHANGE_NO_PREROLL.
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*
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* A typical live source will timestamp the buffers it creates with the
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* current running time of the pipeline. This is one reason why a live source
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* can only produce data in the PLAYING state, when the clock is actually
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* distributed and running.
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*
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* Live sources that synchronize and block on the clock (an audio source, for
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* example) can use gst_base_src_wait_playing() when the
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* #GstBaseSrcClass::create function was interrupted by a state change to
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* PAUSED.
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*
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* The #GstBaseSrcClass::get_times method can be used to implement pseudo-live
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* sources. It only makes sense to implement the #GstBaseSrcClass::get_times
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* function if the source is a live source. The #GstBaseSrcClass::get_times
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* function should return timestamps starting from 0, as if it were a non-live
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* source. The base class will make sure that the timestamps are transformed
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* into the current running_time. The base source will then wait for the
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* calculated running_time before pushing out the buffer.
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*
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* For live sources, the base class will by default report a latency of 0.
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* For pseudo live sources, the base class will by default measure the difference
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* between the first buffer timestamp and the start time of get_times and will
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* report this value as the latency.
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* Subclasses should override the query function when this behaviour is not
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* acceptable.
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*
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* There is only support in #GstBaseSrc for exactly one source pad, which
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* should be named "src". A source implementation (subclass of #GstBaseSrc)
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* should install a pad template in its class_init function, like so:
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* |[<!-- language="C" -->
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* static void
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* my_element_class_init (GstMyElementClass *klass)
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* {
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* GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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* // srctemplate should be a #GstStaticPadTemplate with direction
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* // %GST_PAD_SRC and name "src"
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* gst_element_class_add_static_pad_template (gstelement_class, &srctemplate);
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*
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* gst_element_class_set_static_metadata (gstelement_class,
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* "Source name",
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* "Source",
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* "My Source element",
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* "The author <my.sink@my.email>");
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* }
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* ]|
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*
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* ## Controlled shutdown of live sources in applications
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*
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* Applications that record from a live source may want to stop recording
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* in a controlled way, so that the recording is stopped, but the data
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* already in the pipeline is processed to the end (remember that many live
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* sources would go on recording forever otherwise). For that to happen the
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* application needs to make the source stop recording and send an EOS
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* event down the pipeline. The application would then wait for an
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* EOS message posted on the pipeline's bus to know when all data has
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* been processed and the pipeline can safely be stopped.
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*
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* An application may send an EOS event to a source element to make it
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* perform the EOS logic (send EOS event downstream or post a
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* %GST_MESSAGE_SEGMENT_DONE on the bus). This can typically be done
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* with the gst_element_send_event() function on the element or its parent bin.
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*
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* After the EOS has been sent to the element, the application should wait for
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* an EOS message to be posted on the pipeline's bus. Once this EOS message is
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* received, it may safely shut down the entire pipeline.
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*
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include <stdlib.h>
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#include <string.h>
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#include <gst/gst_private.h>
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#include <gst/glib-compat-private.h>
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#include "gstbasesrc.h"
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#include <glib/gi18n-lib.h>
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GST_DEBUG_CATEGORY_STATIC (gst_base_src_debug);
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#define GST_CAT_DEFAULT gst_base_src_debug
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#define GST_LIVE_GET_LOCK(elem) (&GST_BASE_SRC_CAST(elem)->live_lock)
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#define GST_LIVE_LOCK(elem) g_mutex_lock(GST_LIVE_GET_LOCK(elem))
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#define GST_LIVE_TRYLOCK(elem) g_mutex_trylock(GST_LIVE_GET_LOCK(elem))
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#define GST_LIVE_UNLOCK(elem) g_mutex_unlock(GST_LIVE_GET_LOCK(elem))
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#define GST_LIVE_GET_COND(elem) (&GST_BASE_SRC_CAST(elem)->live_cond)
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#define GST_LIVE_WAIT(elem) g_cond_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem))
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#define GST_LIVE_WAIT_UNTIL(elem, end_time) g_cond_timed_wait (GST_LIVE_GET_COND (elem), GST_LIVE_GET_LOCK (elem), end_time)
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#define GST_LIVE_SIGNAL(elem) g_cond_signal (GST_LIVE_GET_COND (elem));
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#define GST_LIVE_BROADCAST(elem) g_cond_broadcast (GST_LIVE_GET_COND (elem));
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#define GST_ASYNC_GET_COND(elem) (&GST_BASE_SRC_CAST(elem)->priv->async_cond)
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#define GST_ASYNC_WAIT(elem) g_cond_wait (GST_ASYNC_GET_COND (elem), GST_OBJECT_GET_LOCK (elem))
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#define GST_ASYNC_SIGNAL(elem) g_cond_signal (GST_ASYNC_GET_COND (elem));
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#define CLEAR_PENDING_EOS(bsrc) \
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G_STMT_START { \
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g_atomic_int_set (&bsrc->priv->has_pending_eos, FALSE); \
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gst_event_replace (&bsrc->priv->pending_eos, NULL); \
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} G_STMT_END
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/* BaseSrc signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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#define DEFAULT_BLOCKSIZE 4096
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#define DEFAULT_NUM_BUFFERS -1
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#define DEFAULT_DO_TIMESTAMP FALSE
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/* FIXME 2.0: automatic_eos should probably be disabled by default,
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* see https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1330 */
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#define DEFAULT_AUTOMATIC_EOS TRUE
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enum
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{
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PROP_0,
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PROP_BLOCKSIZE,
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PROP_NUM_BUFFERS,
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#ifndef GST_REMOVE_DEPRECATED
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PROP_TYPEFIND,
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#endif
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PROP_DO_TIMESTAMP,
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PROP_AUTOMATIC_EOS
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};
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/* The basesrc implementation need to respect the following locking order:
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* 1. STREAM_LOCK
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* 2. LIVE_LOCK
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* 3. OBJECT_LOCK
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*/
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struct _GstBaseSrcPrivate
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{
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gboolean discont; /* STREAM_LOCK */
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gboolean flushing; /* LIVE_LOCK */
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GstFlowReturn start_result; /* OBJECT_LOCK */
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gboolean async; /* OBJECT_LOCK */
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/* if a stream-start event should be sent */
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gboolean stream_start_pending; /* STREAM_LOCK */
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/* if segment should be sent and a
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* seqnum if it was originated by a seek */
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gboolean segment_pending; /* OBJECT_LOCK */
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guint32 segment_seqnum; /* OBJECT_LOCK */
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/* if EOS is pending (atomic) */
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GstEvent *pending_eos; /* OBJECT_LOCK */
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gint has_pending_eos; /* atomic */
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/* if the eos was caused by a forced eos from the application */
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gboolean forced_eos; /* LIVE_LOCK */
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/* startup latency is the time it takes between going to PLAYING and producing
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* the first BUFFER with running_time 0. This value is included in the latency
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* reporting. */
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GstClockTime latency; /* OBJECT_LOCK */
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/* timestamp offset, this is the offset add to the values of gst_times for
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* pseudo live sources */
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GstClockTimeDiff ts_offset; /* OBJECT_LOCK */
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gboolean do_timestamp; /* OBJECT_LOCK */
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gint dynamic_size; /* atomic */
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gint automatic_eos; /* atomic */
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/* stream sequence number */
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guint32 seqnum; /* STREAM_LOCK */
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/* pending events (TAG, CUSTOM_BOTH, CUSTOM_DOWNSTREAM) to be
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* pushed in the data stream */
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GList *pending_events; /* OBJECT_LOCK */
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gint have_events; /* OBJECT_LOCK */
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/* QoS *//* with LOCK */
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gdouble proportion; /* OBJECT_LOCK */
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GstClockTime earliest_time; /* OBJECT_LOCK */
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GstBufferPool *pool; /* OBJECT_LOCK */
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GstAllocator *allocator; /* OBJECT_LOCK */
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GstAllocationParams params; /* OBJECT_LOCK */
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GCond async_cond; /* OBJECT_LOCK */
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/* for _submit_buffer_list() */
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GstBufferList *pending_bufferlist;
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/* to delay unlock_stop */
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gboolean pending_unlock_stop;
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};
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#define BASE_SRC_HAS_PENDING_BUFFER_LIST(src) \
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((src)->priv->pending_bufferlist != NULL)
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static GstElementClass *parent_class = NULL;
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static gint private_offset = 0;
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static void gst_base_src_class_init (GstBaseSrcClass * klass);
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static void gst_base_src_init (GstBaseSrc * src, gpointer g_class);
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static void gst_base_src_finalize (GObject * object);
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GType
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gst_base_src_get_type (void)
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{
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static gsize base_src_type = 0;
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if (g_once_init_enter (&base_src_type)) {
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GType _type;
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static const GTypeInfo base_src_info = {
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sizeof (GstBaseSrcClass),
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NULL,
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NULL,
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(GClassInitFunc) gst_base_src_class_init,
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NULL,
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NULL,
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sizeof (GstBaseSrc),
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0,
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(GInstanceInitFunc) gst_base_src_init,
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};
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_type = g_type_register_static (GST_TYPE_ELEMENT,
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"GstBaseSrc", &base_src_info, G_TYPE_FLAG_ABSTRACT);
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private_offset =
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g_type_add_instance_private (_type, sizeof (GstBaseSrcPrivate));
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g_once_init_leave (&base_src_type, _type);
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}
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return base_src_type;
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}
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static inline GstBaseSrcPrivate *
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gst_base_src_get_instance_private (GstBaseSrc * self)
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{
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return (G_STRUCT_MEMBER_P (self, private_offset));
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}
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static GstCaps *gst_base_src_default_get_caps (GstBaseSrc * bsrc,
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GstCaps * filter);
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static GstCaps *gst_base_src_default_fixate (GstBaseSrc * src, GstCaps * caps);
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static GstCaps *gst_base_src_fixate (GstBaseSrc * src, GstCaps * caps);
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static gboolean gst_base_src_is_random_access (GstBaseSrc * src);
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static gboolean gst_base_src_activate_mode (GstPad * pad, GstObject * parent,
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GstPadMode mode, gboolean active);
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static void gst_base_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_base_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_base_src_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static gboolean gst_base_src_send_event (GstElement * elem, GstEvent * event);
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static gboolean gst_base_src_default_event (GstBaseSrc * src, GstEvent * event);
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static gboolean gst_base_src_query (GstPad * pad, GstObject * parent,
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GstQuery * query);
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static void gst_base_src_set_pool_flushing (GstBaseSrc * basesrc,
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gboolean flushing);
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static gboolean gst_base_src_default_negotiate (GstBaseSrc * basesrc);
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static gboolean gst_base_src_default_do_seek (GstBaseSrc * src,
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GstSegment * segment);
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static gboolean gst_base_src_default_query (GstBaseSrc * src, GstQuery * query);
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static gboolean gst_base_src_default_prepare_seek_segment (GstBaseSrc * src,
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GstEvent * event, GstSegment * segment);
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static GstFlowReturn gst_base_src_default_create (GstBaseSrc * basesrc,
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guint64 offset, guint size, GstBuffer ** buf);
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static GstFlowReturn gst_base_src_default_alloc (GstBaseSrc * basesrc,
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guint64 offset, guint size, GstBuffer ** buf);
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static gboolean gst_base_src_decide_allocation_default (GstBaseSrc * basesrc,
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GstQuery * query);
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static gboolean gst_base_src_set_flushing (GstBaseSrc * basesrc,
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gboolean flushing);
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static gboolean gst_base_src_start (GstBaseSrc * basesrc);
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static gboolean gst_base_src_stop (GstBaseSrc * basesrc);
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static GstStateChangeReturn gst_base_src_change_state (GstElement * element,
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GstStateChange transition);
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static void gst_base_src_loop (GstPad * pad);
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static GstFlowReturn gst_base_src_getrange (GstPad * pad, GstObject * parent,
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guint64 offset, guint length, GstBuffer ** buf);
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static GstFlowReturn gst_base_src_get_range (GstBaseSrc * src, guint64 offset,
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guint length, GstBuffer ** buf);
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static gboolean gst_base_src_seekable (GstBaseSrc * src);
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static gboolean gst_base_src_negotiate_unlocked (GstBaseSrc * basesrc);
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static gboolean gst_base_src_update_length (GstBaseSrc * src, guint64 offset,
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guint * length, gboolean force);
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static void
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gst_base_src_class_init (GstBaseSrcClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
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if (private_offset != 0)
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g_type_class_adjust_private_offset (klass, &private_offset);
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GST_DEBUG_CATEGORY_INIT (gst_base_src_debug, "basesrc", 0, "basesrc element");
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_base_src_finalize;
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gobject_class->set_property = gst_base_src_set_property;
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gobject_class->get_property = gst_base_src_get_property;
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g_object_class_install_property (gobject_class, PROP_BLOCKSIZE,
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g_param_spec_uint ("blocksize", "Block size",
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"Size in bytes to read per buffer (-1 = default)", 0, G_MAXUINT,
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DEFAULT_BLOCKSIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NUM_BUFFERS,
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g_param_spec_int ("num-buffers", "num-buffers",
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"Number of buffers to output before sending EOS (-1 = unlimited)",
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-1, G_MAXINT, DEFAULT_NUM_BUFFERS, G_PARAM_READWRITE |
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G_PARAM_STATIC_STRINGS));
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#ifndef GST_REMOVE_DEPRECATED
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g_object_class_install_property (gobject_class, PROP_TYPEFIND,
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g_param_spec_boolean ("typefind", "Typefind",
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"Run typefind before negotiating (deprecated, non-functional)", FALSE,
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G_PARAM_READWRITE | G_PARAM_DEPRECATED | G_PARAM_STATIC_STRINGS));
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#endif
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g_object_class_install_property (gobject_class, PROP_DO_TIMESTAMP,
|
|
g_param_spec_boolean ("do-timestamp", "Do timestamp",
|
|
"Apply current stream time to buffers", DEFAULT_DO_TIMESTAMP,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstBaseSrc:automatic-eos:
|
|
*
|
|
* See gst_base_src_set_automatic_eos()
|
|
*
|
|
* Since: 1.24
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_AUTOMATIC_EOS,
|
|
g_param_spec_boolean ("automatic-eos", "Automatic EOS",
|
|
"Automatically EOS when the segment is done", DEFAULT_AUTOMATIC_EOS,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_base_src_change_state);
|
|
gstelement_class->send_event = GST_DEBUG_FUNCPTR (gst_base_src_send_event);
|
|
|
|
klass->get_caps = GST_DEBUG_FUNCPTR (gst_base_src_default_get_caps);
|
|
klass->negotiate = GST_DEBUG_FUNCPTR (gst_base_src_default_negotiate);
|
|
klass->fixate = GST_DEBUG_FUNCPTR (gst_base_src_default_fixate);
|
|
klass->prepare_seek_segment =
|
|
GST_DEBUG_FUNCPTR (gst_base_src_default_prepare_seek_segment);
|
|
klass->do_seek = GST_DEBUG_FUNCPTR (gst_base_src_default_do_seek);
|
|
klass->query = GST_DEBUG_FUNCPTR (gst_base_src_default_query);
|
|
klass->event = GST_DEBUG_FUNCPTR (gst_base_src_default_event);
|
|
klass->create = GST_DEBUG_FUNCPTR (gst_base_src_default_create);
|
|
klass->alloc = GST_DEBUG_FUNCPTR (gst_base_src_default_alloc);
|
|
klass->decide_allocation =
|
|
GST_DEBUG_FUNCPTR (gst_base_src_decide_allocation_default);
|
|
|
|
/* Registering debug symbols for function pointers */
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_activate_mode);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_event);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_query);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_getrange);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_base_src_fixate);
|
|
}
|
|
|
|
static void
|
|
gst_base_src_init (GstBaseSrc * basesrc, gpointer g_class)
|
|
{
|
|
GstPad *pad;
|
|
GstPadTemplate *pad_template;
|
|
|
|
basesrc->priv = gst_base_src_get_instance_private (basesrc);
|
|
|
|
basesrc->is_live = FALSE;
|
|
g_mutex_init (&basesrc->live_lock);
|
|
g_cond_init (&basesrc->live_cond);
|
|
basesrc->num_buffers = DEFAULT_NUM_BUFFERS;
|
|
basesrc->num_buffers_left = -1;
|
|
g_atomic_int_set (&basesrc->priv->automatic_eos, DEFAULT_AUTOMATIC_EOS);
|
|
|
|
basesrc->can_activate_push = TRUE;
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (g_class), "src");
|
|
g_return_if_fail (pad_template != NULL);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "creating src pad");
|
|
pad = gst_pad_new_from_template (pad_template, "src");
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "setting functions on src pad");
|
|
gst_pad_set_activatemode_function (pad, gst_base_src_activate_mode);
|
|
gst_pad_set_event_function (pad, gst_base_src_event);
|
|
gst_pad_set_query_function (pad, gst_base_src_query);
|
|
gst_pad_set_getrange_function (pad, gst_base_src_getrange);
|
|
|
|
/* hold pointer to pad */
|
|
basesrc->srcpad = pad;
|
|
GST_DEBUG_OBJECT (basesrc, "adding src pad");
|
|
gst_element_add_pad (GST_ELEMENT (basesrc), pad);
|
|
|
|
basesrc->blocksize = DEFAULT_BLOCKSIZE;
|
|
basesrc->clock_id = NULL;
|
|
/* we operate in BYTES by default */
|
|
gst_base_src_set_format (basesrc, GST_FORMAT_BYTES);
|
|
basesrc->priv->do_timestamp = DEFAULT_DO_TIMESTAMP;
|
|
g_atomic_int_set (&basesrc->priv->have_events, FALSE);
|
|
|
|
g_cond_init (&basesrc->priv->async_cond);
|
|
basesrc->priv->start_result = GST_FLOW_FLUSHING;
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTED);
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
GST_OBJECT_FLAG_SET (basesrc, GST_ELEMENT_FLAG_SOURCE);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "init done");
|
|
}
|
|
|
|
static void
|
|
gst_base_src_finalize (GObject * object)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
GstEvent **event_p;
|
|
|
|
basesrc = GST_BASE_SRC (object);
|
|
|
|
g_mutex_clear (&basesrc->live_lock);
|
|
g_cond_clear (&basesrc->live_cond);
|
|
g_cond_clear (&basesrc->priv->async_cond);
|
|
|
|
event_p = &basesrc->pending_seek;
|
|
gst_event_replace (event_p, NULL);
|
|
|
|
if (basesrc->priv->pending_events) {
|
|
g_list_foreach (basesrc->priv->pending_events, (GFunc) gst_event_unref,
|
|
NULL);
|
|
g_list_free (basesrc->priv->pending_events);
|
|
}
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
/* Call with LIVE_LOCK held */
|
|
static GstFlowReturn
|
|
gst_base_src_wait_playing_unlocked (GstBaseSrc * src)
|
|
{
|
|
while (G_UNLIKELY (!src->live_running && !src->priv->flushing)) {
|
|
/* block until the state changes, or we get a flush, or something */
|
|
GST_DEBUG_OBJECT (src, "live source waiting for running state");
|
|
GST_LIVE_WAIT (src);
|
|
GST_DEBUG_OBJECT (src, "live source unlocked");
|
|
}
|
|
|
|
if (src->priv->flushing)
|
|
goto flushing;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_base_src_wait_playing:
|
|
* @src: the src
|
|
*
|
|
* If the #GstBaseSrcClass::create 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 produce 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_FLUSHING).
|
|
*
|
|
* Returns: %GST_FLOW_OK if @src is PLAYING and processing can
|
|
* continue. Any other return value should be returned from the create vmethod.
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_src_wait_playing (GstBaseSrc * src)
|
|
{
|
|
GstFlowReturn ret;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), GST_FLOW_ERROR);
|
|
|
|
GST_LIVE_LOCK (src);
|
|
ret = gst_base_src_wait_playing_unlocked (src);
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_live:
|
|
* @src: base source instance
|
|
* @live: new live-mode
|
|
*
|
|
* If the element listens to a live source, @live should
|
|
* be set to %TRUE.
|
|
*
|
|
* A live source will not produce data in the PAUSED state and
|
|
* will therefore not be able to participate in the PREROLL phase
|
|
* of a pipeline. To signal this fact to the application and the
|
|
* pipeline, the state change return value of the live source will
|
|
* be GST_STATE_CHANGE_NO_PREROLL.
|
|
*/
|
|
void
|
|
gst_base_src_set_live (GstBaseSrc * src, gboolean live)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->is_live = live;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_is_live:
|
|
* @src: base source instance
|
|
*
|
|
* Check if an element is in live mode.
|
|
*
|
|
* Returns: %TRUE if element is in live mode.
|
|
*/
|
|
gboolean
|
|
gst_base_src_is_live (GstBaseSrc * src)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
result = src->is_live;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_format:
|
|
* @src: base source instance
|
|
* @format: the format to use
|
|
*
|
|
* Sets the default format of the source. This will be the format used
|
|
* for sending SEGMENT events and for performing seeks.
|
|
*
|
|
* If a format of GST_FORMAT_BYTES is set, the element will be able to
|
|
* operate in pull mode if the #GstBaseSrcClass::is_seekable returns %TRUE.
|
|
*
|
|
* This function must only be called in states < %GST_STATE_PAUSED.
|
|
*/
|
|
void
|
|
gst_base_src_set_format (GstBaseSrc * src, GstFormat format)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
g_return_if_fail (GST_STATE (src) <= GST_STATE_READY);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
gst_segment_init (&src->segment, format);
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_dynamic_size:
|
|
* @src: base source instance
|
|
* @dynamic: new dynamic size mode
|
|
*
|
|
* If not @dynamic, size is only updated when needed, such as when trying to
|
|
* read past current tracked size. Otherwise, size is checked for upon each
|
|
* read.
|
|
*/
|
|
void
|
|
gst_base_src_set_dynamic_size (GstBaseSrc * src, gboolean dynamic)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
g_atomic_int_set (&src->priv->dynamic_size, dynamic);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_automatic_eos:
|
|
* @src: base source instance
|
|
* @automatic_eos: automatic eos
|
|
*
|
|
* If @automatic_eos is %TRUE, @src will automatically go EOS if a buffer
|
|
* after the total size is returned. By default this is %TRUE but sources
|
|
* that can't return an authoritative size and only know that they're EOS
|
|
* when trying to read more should set this to %FALSE.
|
|
*
|
|
* When @src operates in %GST_FORMAT_TIME, #GstBaseSrc will send an EOS
|
|
* when a buffer outside of the currently configured segment is pushed if
|
|
* @automatic_eos is %TRUE. Since 1.16, if @automatic_eos is %FALSE an
|
|
* EOS will be pushed only when the #GstBaseSrcClass::create implementation
|
|
* returns %GST_FLOW_EOS.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
void
|
|
gst_base_src_set_automatic_eos (GstBaseSrc * src, gboolean automatic_eos)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
g_atomic_int_set (&src->priv->automatic_eos, automatic_eos);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_async:
|
|
* @src: base source instance
|
|
* @async: new async mode
|
|
*
|
|
* Configure async behaviour in @src, no state change will block. The open,
|
|
* close, start, stop, play and pause virtual methods will be executed in a
|
|
* different thread and are thus allowed to perform blocking operations. Any
|
|
* blocking operation should be unblocked with the unlock vmethod.
|
|
*/
|
|
void
|
|
gst_base_src_set_async (GstBaseSrc * src, gboolean async)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->async = async;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_is_async:
|
|
* @src: base source instance
|
|
*
|
|
* Get the current async behaviour of @src. See also gst_base_src_set_async().
|
|
*
|
|
* Returns: %TRUE if @src is operating in async mode.
|
|
*/
|
|
gboolean
|
|
gst_base_src_is_async (GstBaseSrc * src)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
res = src->priv->async;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_base_src_query_latency:
|
|
* @src: the source
|
|
* @live: (out) (allow-none): if the source is live
|
|
* @min_latency: (out) (allow-none): the min latency of the source
|
|
* @max_latency: (out) (allow-none): the max latency of the source
|
|
*
|
|
* Query the source for the latency parameters. @live will be %TRUE when @src is
|
|
* configured as a live source. @min_latency and @max_latency will be set
|
|
* to the difference between the running time and the timestamp of the first
|
|
* buffer.
|
|
*
|
|
* This function is mostly used by subclasses.
|
|
*
|
|
* Returns: %TRUE if the query succeeded.
|
|
*/
|
|
gboolean
|
|
gst_base_src_query_latency (GstBaseSrc * src, gboolean * live,
|
|
GstClockTime * min_latency, GstClockTime * max_latency)
|
|
{
|
|
GstClockTime min;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
if (live)
|
|
*live = src->is_live;
|
|
|
|
/* if we have a startup latency, report this one, else report 0. Subclasses
|
|
* are supposed to override the query function if they want something
|
|
* else. */
|
|
if (src->priv->latency != -1)
|
|
min = src->priv->latency;
|
|
else
|
|
min = 0;
|
|
|
|
if (min_latency)
|
|
*min_latency = min;
|
|
if (max_latency)
|
|
*max_latency = min;
|
|
|
|
GST_LOG_OBJECT (src, "latency: live %d, min %" GST_TIME_FORMAT
|
|
", max %" GST_TIME_FORMAT, src->is_live, GST_TIME_ARGS (min),
|
|
GST_TIME_ARGS (min));
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_blocksize:
|
|
* @src: the source
|
|
* @blocksize: the new blocksize in bytes
|
|
*
|
|
* Set the number of bytes that @src will push out with each buffer. When
|
|
* @blocksize is set to -1, a default length will be used.
|
|
*/
|
|
void
|
|
gst_base_src_set_blocksize (GstBaseSrc * src, guint blocksize)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->blocksize = blocksize;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_get_blocksize:
|
|
* @src: the source
|
|
*
|
|
* Get the number of bytes that @src will push out with each buffer.
|
|
*
|
|
* Returns: the number of bytes pushed with each buffer.
|
|
*/
|
|
guint
|
|
gst_base_src_get_blocksize (GstBaseSrc * src)
|
|
{
|
|
gint res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), 0);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
res = src->blocksize;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
|
|
/**
|
|
* gst_base_src_set_do_timestamp:
|
|
* @src: the source
|
|
* @timestamp: enable or disable timestamping
|
|
*
|
|
* Configure @src to automatically timestamp outgoing buffers based on the
|
|
* current running_time of the pipeline. This property is mostly useful for live
|
|
* sources.
|
|
*/
|
|
void
|
|
gst_base_src_set_do_timestamp (GstBaseSrc * src, gboolean timestamp)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->do_timestamp = timestamp;
|
|
if (timestamp && src->segment.format != GST_FORMAT_TIME)
|
|
gst_segment_init (&src->segment, GST_FORMAT_TIME);
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_get_do_timestamp:
|
|
* @src: the source
|
|
*
|
|
* Query if @src timestamps outgoing buffers based on the current running_time.
|
|
*
|
|
* Returns: %TRUE if the base class will automatically timestamp outgoing buffers.
|
|
*/
|
|
gboolean
|
|
gst_base_src_get_do_timestamp (GstBaseSrc * src)
|
|
{
|
|
gboolean res;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
res = src->priv->do_timestamp;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_new_seamless_segment:
|
|
* @src: The source
|
|
* @start: The new start value for the segment
|
|
* @stop: Stop value for the new segment
|
|
* @time: The new time value for the start of the new segment
|
|
*
|
|
* Prepare a new seamless segment for emission downstream. This function must
|
|
* only be called by derived sub-classes, and only from the #GstBaseSrcClass::create function,
|
|
* as the stream-lock needs to be held.
|
|
*
|
|
* The format for the new segment will be the current format of the source, as
|
|
* configured with gst_base_src_set_format()
|
|
*
|
|
* Returns: %TRUE if preparation of the seamless segment succeeded.
|
|
*
|
|
* Deprecated: 1.18: Use gst_base_src_new_segment()
|
|
*/
|
|
gboolean
|
|
gst_base_src_new_seamless_segment (GstBaseSrc * src, gint64 start, gint64 stop,
|
|
gint64 time)
|
|
{
|
|
gboolean res = TRUE;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
|
|
src->segment.base = gst_segment_to_running_time (&src->segment,
|
|
src->segment.format, src->segment.position);
|
|
src->segment.position = src->segment.start = start;
|
|
src->segment.stop = stop;
|
|
src->segment.time = time;
|
|
|
|
/* Mark pending segment. Will be sent before next data */
|
|
src->priv->segment_pending = TRUE;
|
|
src->priv->segment_seqnum = gst_util_seqnum_next ();
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"Starting new seamless segment. Start %" GST_TIME_FORMAT " stop %"
|
|
GST_TIME_FORMAT " time %" GST_TIME_FORMAT " base %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (start), GST_TIME_ARGS (stop), GST_TIME_ARGS (time),
|
|
GST_TIME_ARGS (src->segment.base));
|
|
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
src->priv->discont = TRUE;
|
|
src->running = TRUE;
|
|
|
|
return res;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_new_segment:
|
|
* @src: a #GstBaseSrc
|
|
* @segment: a pointer to a #GstSegment
|
|
*
|
|
* Prepare a new segment for emission downstream. This function must
|
|
* only be called by derived sub-classes, and only from the #GstBaseSrcClass::create function,
|
|
* as the stream-lock needs to be held.
|
|
*
|
|
* The format for the @segment must be identical with the current format
|
|
* of the source, as configured with gst_base_src_set_format().
|
|
*
|
|
* The format of @src must not be %GST_FORMAT_UNDEFINED and the format
|
|
* should be configured via gst_base_src_set_format() before calling this method.
|
|
*
|
|
* Returns: %TRUE if preparation of new segment succeeded.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
gboolean
|
|
gst_base_src_new_segment (GstBaseSrc * src, const GstSegment * segment)
|
|
{
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
g_return_val_if_fail (segment != NULL, FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
|
|
if (src->segment.format == GST_FORMAT_UNDEFINED) {
|
|
/* subclass must set valid format before calling this method */
|
|
GST_WARNING_OBJECT (src, "segment format is not configured yet, ignore");
|
|
GST_OBJECT_UNLOCK (src);
|
|
return FALSE;
|
|
}
|
|
|
|
if (src->segment.format != segment->format) {
|
|
GST_WARNING_OBJECT (src, "segment format mismatched, ignore");
|
|
GST_OBJECT_UNLOCK (src);
|
|
return FALSE;
|
|
}
|
|
|
|
gst_segment_copy_into (segment, &src->segment);
|
|
|
|
/* Mark pending segment. Will be sent before next data */
|
|
src->priv->segment_pending = TRUE;
|
|
src->priv->segment_seqnum = gst_util_seqnum_next ();
|
|
|
|
GST_DEBUG_OBJECT (src, "Starting new segment %" GST_SEGMENT_FORMAT, segment);
|
|
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
src->running = TRUE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* called with STREAM_LOCK */
|
|
static gboolean
|
|
gst_base_src_send_stream_start (GstBaseSrc * src)
|
|
{
|
|
gboolean ret = TRUE;
|
|
|
|
if (src->priv->stream_start_pending) {
|
|
gchar *stream_id;
|
|
GstEvent *event;
|
|
|
|
stream_id =
|
|
gst_pad_create_stream_id (src->srcpad, GST_ELEMENT_CAST (src), NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "Pushing STREAM_START");
|
|
event = gst_event_new_stream_start (stream_id);
|
|
gst_event_set_group_id (event, gst_util_group_id_next ());
|
|
|
|
ret = gst_pad_push_event (src->srcpad, event);
|
|
src->priv->stream_start_pending = FALSE;
|
|
g_free (stream_id);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_set_caps:
|
|
* @src: a #GstBaseSrc
|
|
* @caps: (transfer none): a #GstCaps
|
|
*
|
|
* Set new caps on the basesrc source pad.
|
|
*
|
|
* Returns: %TRUE if the caps could be set
|
|
*/
|
|
gboolean
|
|
gst_base_src_set_caps (GstBaseSrc * src, GstCaps * caps)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean res = TRUE;
|
|
GstCaps *current_caps;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
gst_base_src_send_stream_start (src);
|
|
|
|
current_caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (src));
|
|
if (current_caps && gst_caps_is_equal (current_caps, caps)) {
|
|
GST_DEBUG_OBJECT (src, "New caps equal to old ones: %" GST_PTR_FORMAT,
|
|
caps);
|
|
res = TRUE;
|
|
} else {
|
|
if (bclass->set_caps)
|
|
res = bclass->set_caps (src, caps);
|
|
|
|
if (res)
|
|
res = gst_pad_push_event (src->srcpad, gst_event_new_caps (caps));
|
|
}
|
|
|
|
if (current_caps)
|
|
gst_caps_unref (current_caps);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_src_default_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
|
|
{
|
|
GstCaps *caps = NULL;
|
|
GstPadTemplate *pad_template;
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
|
|
|
|
pad_template =
|
|
gst_element_class_get_pad_template (GST_ELEMENT_CLASS (bclass), "src");
|
|
|
|
if (pad_template != NULL) {
|
|
caps = gst_pad_template_get_caps (pad_template);
|
|
|
|
if (filter) {
|
|
GstCaps *intersection;
|
|
|
|
intersection =
|
|
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (caps);
|
|
caps = intersection;
|
|
}
|
|
}
|
|
return caps;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_src_default_fixate (GstBaseSrc * bsrc, GstCaps * caps)
|
|
{
|
|
GST_DEBUG_OBJECT (bsrc, "using default caps fixate function");
|
|
return gst_caps_fixate (caps);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_base_src_fixate (GstBaseSrc * bsrc, GstCaps * caps)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (bsrc);
|
|
|
|
if (bclass->fixate)
|
|
caps = bclass->fixate (bsrc, caps);
|
|
|
|
return caps;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_default_query (GstBaseSrc * src, GstQuery * query)
|
|
{
|
|
gboolean res;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_POSITION:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_position (query, &format, NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "position query in format %s",
|
|
gst_format_get_name (format));
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_PERCENT:
|
|
{
|
|
gint64 percent;
|
|
gint64 position;
|
|
gint64 duration;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
position = src->segment.position;
|
|
duration = src->segment.duration;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (position != -1 && duration != -1) {
|
|
if (position < duration)
|
|
percent = gst_util_uint64_scale (GST_FORMAT_PERCENT_MAX, position,
|
|
duration);
|
|
else
|
|
percent = GST_FORMAT_PERCENT_MAX;
|
|
} else
|
|
percent = -1;
|
|
|
|
gst_query_set_position (query, GST_FORMAT_PERCENT, percent);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
{
|
|
gint64 position;
|
|
GstFormat seg_format;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
position =
|
|
gst_segment_to_stream_time (&src->segment, src->segment.format,
|
|
src->segment.position);
|
|
seg_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (position != -1) {
|
|
/* convert to requested format */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, seg_format,
|
|
position, format, &position);
|
|
} else
|
|
res = TRUE;
|
|
|
|
if (res)
|
|
gst_query_set_position (query, format, position);
|
|
|
|
break;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
GstFormat format;
|
|
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "duration query in format %s",
|
|
gst_format_get_name (format));
|
|
|
|
switch (format) {
|
|
case GST_FORMAT_PERCENT:
|
|
gst_query_set_duration (query, GST_FORMAT_PERCENT,
|
|
GST_FORMAT_PERCENT_MAX);
|
|
res = TRUE;
|
|
break;
|
|
default:
|
|
{
|
|
gint64 duration;
|
|
GstFormat seg_format;
|
|
guint length = 0;
|
|
|
|
/* may have to refresh duration */
|
|
gst_base_src_update_length (src, 0, &length,
|
|
g_atomic_int_get (&src->priv->dynamic_size));
|
|
|
|
/* this is the duration as configured by the subclass. */
|
|
GST_OBJECT_LOCK (src);
|
|
duration = src->segment.duration;
|
|
seg_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
GST_LOG_OBJECT (src, "duration %" G_GINT64_FORMAT ", format %s",
|
|
duration, gst_format_get_name (seg_format));
|
|
|
|
if (duration != -1) {
|
|
/* convert to requested format, if this fails, we have a duration
|
|
* but we cannot answer the query, we must return FALSE. */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, seg_format,
|
|
duration, format, &duration);
|
|
} else {
|
|
/* The subclass did not configure a duration, we assume that the
|
|
* media has an unknown duration then and we return TRUE to report
|
|
* this. Note that this is not the same as returning FALSE, which
|
|
* means that we cannot report the duration at all. */
|
|
res = TRUE;
|
|
}
|
|
|
|
if (res)
|
|
gst_query_set_duration (query, format, duration);
|
|
|
|
break;
|
|
}
|
|
}
|
|
break;
|
|
}
|
|
|
|
case GST_QUERY_SEEKING:
|
|
{
|
|
GstFormat format, seg_format;
|
|
gint64 duration;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
duration = src->segment.duration;
|
|
seg_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
|
|
if (format == seg_format) {
|
|
gst_query_set_seeking (query, seg_format,
|
|
gst_base_src_seekable (src), 0, duration);
|
|
res = TRUE;
|
|
} else {
|
|
/* FIXME 2.0: return TRUE + seekable=FALSE for SEEKING query here */
|
|
/* Don't reply to the query to make up for demuxers which don't
|
|
* handle the SEEKING query yet. Players like Totem will fall back
|
|
* to the duration when the SEEKING query isn't answered. */
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_SEGMENT:
|
|
{
|
|
GstFormat format;
|
|
gint64 start, stop;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
|
|
format = src->segment.format;
|
|
|
|
start =
|
|
gst_segment_to_stream_time (&src->segment, format,
|
|
src->segment.start);
|
|
if ((stop = src->segment.stop) == -1)
|
|
stop = src->segment.duration;
|
|
else
|
|
stop = gst_segment_to_stream_time (&src->segment, format, stop);
|
|
|
|
gst_query_set_segment (query, src->segment.rate, format, start, stop);
|
|
|
|
GST_OBJECT_UNLOCK (src);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
|
|
case GST_QUERY_FORMATS:
|
|
{
|
|
gst_query_set_formats (query, 3, GST_FORMAT_DEFAULT,
|
|
GST_FORMAT_BYTES, GST_FORMAT_PERCENT);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_CONVERT:
|
|
{
|
|
GstFormat src_fmt, dest_fmt;
|
|
gint64 src_val, dest_val;
|
|
|
|
gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
|
|
|
|
/* we can only convert between equal formats... */
|
|
if (src_fmt == dest_fmt) {
|
|
dest_val = src_val;
|
|
res = TRUE;
|
|
} else
|
|
res = FALSE;
|
|
|
|
gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
|
|
break;
|
|
}
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
GstClockTime min, max;
|
|
gboolean live;
|
|
|
|
/* Subclasses should override and implement something useful */
|
|
res = gst_base_src_query_latency (src, &live, &min, &max);
|
|
|
|
GST_LOG_OBJECT (src, "report latency: live %d, min %" GST_TIME_FORMAT
|
|
", max %" GST_TIME_FORMAT, live, GST_TIME_ARGS (min),
|
|
GST_TIME_ARGS (max));
|
|
|
|
gst_query_set_latency (query, live, min, max);
|
|
break;
|
|
}
|
|
case GST_QUERY_JITTER:
|
|
case GST_QUERY_RATE:
|
|
res = FALSE;
|
|
break;
|
|
case GST_QUERY_BUFFERING:
|
|
{
|
|
GstFormat format, seg_format;
|
|
gint64 start, stop, estimated;
|
|
|
|
gst_query_parse_buffering_range (query, &format, NULL, NULL, NULL);
|
|
|
|
GST_DEBUG_OBJECT (src, "buffering query in format %s",
|
|
gst_format_get_name (format));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
if (src->random_access) {
|
|
estimated = 0;
|
|
start = 0;
|
|
if (format == GST_FORMAT_PERCENT)
|
|
stop = GST_FORMAT_PERCENT_MAX;
|
|
else
|
|
stop = src->segment.duration;
|
|
} else {
|
|
estimated = -1;
|
|
start = -1;
|
|
stop = -1;
|
|
}
|
|
seg_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
/* convert to required format. When the conversion fails, we can't answer
|
|
* the query. When the value is unknown, we can don't perform conversion
|
|
* but report TRUE. */
|
|
if (format != GST_FORMAT_PERCENT && stop != -1) {
|
|
res = gst_pad_query_convert (src->srcpad, seg_format,
|
|
stop, format, &stop);
|
|
} else {
|
|
res = TRUE;
|
|
}
|
|
if (res && format != GST_FORMAT_PERCENT && start != -1)
|
|
res = gst_pad_query_convert (src->srcpad, seg_format,
|
|
start, format, &start);
|
|
|
|
gst_query_set_buffering_range (query, format, start, stop, estimated);
|
|
break;
|
|
}
|
|
case GST_QUERY_SCHEDULING:
|
|
{
|
|
gboolean random_access;
|
|
|
|
random_access = gst_base_src_is_random_access (src);
|
|
|
|
/* we can operate in getrange mode if the native format is bytes
|
|
* and we are seekable, this condition is set in the random_access
|
|
* flag and is set in the _start() method. */
|
|
gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEEKABLE, 1, -1, 0);
|
|
if (random_access)
|
|
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PULL);
|
|
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH);
|
|
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
GstCaps *caps = NULL, *filter;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
if (GST_PAD_IS_FIXED_CAPS (src->srcpad)) {
|
|
caps = gst_pad_get_current_caps (src->srcpad);
|
|
}
|
|
|
|
if (caps == NULL && bclass->get_caps) {
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = bclass->get_caps (src, filter);
|
|
}
|
|
|
|
if (caps != NULL) {
|
|
res = TRUE;
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
}
|
|
case GST_QUERY_URI:{
|
|
if (GST_IS_URI_HANDLER (src)) {
|
|
gchar *uri = gst_uri_handler_get_uri (GST_URI_HANDLER (src));
|
|
|
|
if (uri != NULL) {
|
|
gst_query_set_uri (query, uri);
|
|
g_free (uri);
|
|
res = TRUE;
|
|
} else {
|
|
res = FALSE;
|
|
}
|
|
} else {
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
GST_DEBUG_OBJECT (src, "query %s returns %d", GST_QUERY_TYPE_NAME (query),
|
|
res);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
src = GST_BASE_SRC (parent);
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->query)
|
|
result = bclass->query (src, query);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_default_do_seek (GstBaseSrc * src, 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 (src, "Can't do a default seek");
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_do_seek (GstBaseSrc * src, GstSegment * segment)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
GST_INFO_OBJECT (src, "seeking: %" GST_SEGMENT_FORMAT, segment);
|
|
|
|
if (bclass->do_seek)
|
|
result = bclass->do_seek (src, segment);
|
|
|
|
return result;
|
|
}
|
|
|
|
#define SEEK_TYPE_IS_RELATIVE(t) (((t) != GST_SEEK_TYPE_NONE) && ((t) != GST_SEEK_TYPE_SET))
|
|
|
|
static gboolean
|
|
gst_base_src_default_prepare_seek_segment (GstBaseSrc * src, 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 start_type, stop_type;
|
|
gint64 start, stop;
|
|
GstSeekFlags flags;
|
|
GstFormat seek_format, dest_format;
|
|
gdouble rate;
|
|
gboolean update;
|
|
gboolean res = TRUE;
|
|
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags,
|
|
&start_type, &start, &stop_type, &stop);
|
|
dest_format = segment->format;
|
|
|
|
if (seek_format == dest_format) {
|
|
gst_segment_do_seek (segment, rate, seek_format, flags,
|
|
start_type, start, stop_type, stop, &update);
|
|
return TRUE;
|
|
}
|
|
|
|
if (start_type != GST_SEEK_TYPE_NONE) {
|
|
/* FIXME: Handle seek_end by converting the input segment vals */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, seek_format, start, dest_format,
|
|
&start);
|
|
start_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
if (res && stop_type != GST_SEEK_TYPE_NONE) {
|
|
/* FIXME: Handle seek_end by converting the input segment vals */
|
|
res =
|
|
gst_pad_query_convert (src->srcpad, seek_format, stop, dest_format,
|
|
&stop);
|
|
stop_type = GST_SEEK_TYPE_SET;
|
|
}
|
|
|
|
/* And finally, configure our output segment in the desired format */
|
|
if (res) {
|
|
res =
|
|
gst_segment_do_seek (segment, rate, dest_format, flags, start_type,
|
|
start, stop_type, stop, &update);
|
|
}
|
|
|
|
if (!res)
|
|
goto no_format;
|
|
|
|
return res;
|
|
|
|
no_format:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "undefined format given, seek aborted.");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_prepare_seek_segment (GstBaseSrc * src, GstEvent * event,
|
|
GstSegment * seeksegment)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->prepare_seek_segment)
|
|
result = bclass->prepare_seek_segment (src, event, seeksegment);
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_src_default_alloc (GstBaseSrc * src, guint64 offset,
|
|
guint size, GstBuffer ** buffer)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstBaseSrcPrivate *priv = src->priv;
|
|
GstBufferPool *pool = NULL;
|
|
GstAllocator *allocator = NULL;
|
|
GstAllocationParams params;
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
if (priv->pool) {
|
|
pool = gst_object_ref (priv->pool);
|
|
} else if (priv->allocator) {
|
|
allocator = gst_object_ref (priv->allocator);
|
|
}
|
|
params = priv->params;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (pool) {
|
|
ret = gst_buffer_pool_acquire_buffer (pool, buffer, NULL);
|
|
} else if (size != -1) {
|
|
*buffer = gst_buffer_new_allocate (allocator, size, ¶ms);
|
|
if (G_UNLIKELY (*buffer == NULL))
|
|
goto alloc_failed;
|
|
|
|
ret = GST_FLOW_OK;
|
|
} else {
|
|
GST_WARNING_OBJECT (src, "Not trying to alloc %u bytes. Blocksize not set?",
|
|
size);
|
|
goto alloc_failed;
|
|
}
|
|
|
|
done:
|
|
if (pool)
|
|
gst_object_unref (pool);
|
|
if (allocator)
|
|
gst_object_unref (allocator);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
alloc_failed:
|
|
{
|
|
GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size);
|
|
ret = GST_FLOW_ERROR;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_src_default_create (GstBaseSrc * src, guint64 offset,
|
|
guint size, GstBuffer ** buffer)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
GstFlowReturn ret;
|
|
GstBuffer *res_buf;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (G_UNLIKELY (!bclass->alloc))
|
|
goto no_function;
|
|
if (G_UNLIKELY (!bclass->fill))
|
|
goto no_function;
|
|
|
|
if (*buffer == NULL) {
|
|
/* downstream did not provide us with a buffer to fill, allocate one
|
|
* ourselves */
|
|
ret = bclass->alloc (src, offset, size, &res_buf);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto alloc_failed;
|
|
} else {
|
|
res_buf = *buffer;
|
|
}
|
|
|
|
if (G_LIKELY (size > 0)) {
|
|
/* only call fill when there is a size */
|
|
ret = bclass->fill (src, offset, size, res_buf);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto not_ok;
|
|
}
|
|
|
|
*buffer = res_buf;
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
no_function:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no fill or alloc function");
|
|
return GST_FLOW_NOT_SUPPORTED;
|
|
}
|
|
alloc_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "Failed to allocate buffer of %u bytes", size);
|
|
return ret;
|
|
}
|
|
not_ok:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "fill returned %d (%s)", ret,
|
|
gst_flow_get_name (ret));
|
|
if (*buffer == NULL)
|
|
gst_buffer_unref (res_buf);
|
|
return ret;
|
|
}
|
|
}
|
|
|
|
/* this code implements the seeking. It is a good example
|
|
* handling all cases.
|
|
*
|
|
* A seek updates the currently configured segment.start
|
|
* and segment.stop values based on the SEEK_TYPE. If the
|
|
* segment.start value is updated, a seek to this new position
|
|
* should be performed.
|
|
*
|
|
* The seek can only be executed when we are not currently
|
|
* streaming any data, to make sure that this is the case, we
|
|
* acquire the STREAM_LOCK which is taken when we are in the
|
|
* _loop() function or when a getrange() is called. Normally
|
|
* we will not receive a seek if we are operating in pull mode
|
|
* though. When we operate as a live source we might block on the live
|
|
* cond, which does not release the STREAM_LOCK. Therefore we will try
|
|
* to grab the LIVE_LOCK instead of the STREAM_LOCK to make sure it is
|
|
* safe to perform the seek.
|
|
*
|
|
* When we are in the loop() function, we might be in the middle
|
|
* of pushing a buffer, which might block in a sink. To make sure
|
|
* that the push gets unblocked we push out a FLUSH_START event.
|
|
* Our loop function will get a FLUSHING return value from
|
|
* the push and will pause, effectively releasing the STREAM_LOCK.
|
|
*
|
|
* For a non-flushing seek, we pause the task, which might eventually
|
|
* release the STREAM_LOCK. We say eventually because when the sink
|
|
* blocks on the sample we might wait a very long time until the sink
|
|
* unblocks the sample. In any case we acquire the STREAM_LOCK and
|
|
* can continue the seek. A non-flushing seek is normally done in a
|
|
* running pipeline to perform seamless playback, this means that the sink is
|
|
* PLAYING and will return from its chain function.
|
|
* In the case of a non-flushing seek we need to make sure that the
|
|
* data we output after the seek is continuous with the previous data,
|
|
* this is because a non-flushing seek does not reset the running-time
|
|
* to 0. We do this by closing the currently running segment, ie. sending
|
|
* a new_segment event with the stop position set to the last processed
|
|
* position.
|
|
*
|
|
* After updating the segment.start/stop values, we prepare for
|
|
* streaming again. We push out a FLUSH_STOP to make the peer pad
|
|
* accept data again and we start our task again.
|
|
*
|
|
* A segment seek posts a message on the bus saying that the playback
|
|
* of the segment started. We store the segment flag internally because
|
|
* when we reach the segment.stop we have to post a segment.done
|
|
* instead of EOS when doing a segment seek.
|
|
*/
|
|
static gboolean
|
|
gst_base_src_perform_seek (GstBaseSrc * src, GstEvent * event, gboolean unlock)
|
|
{
|
|
gboolean res = TRUE, tres;
|
|
gdouble rate;
|
|
GstFormat seek_format, dest_format;
|
|
GstSeekFlags flags;
|
|
GstSeekType start_type, stop_type;
|
|
gint64 start, stop;
|
|
gboolean flush;
|
|
gboolean update;
|
|
gboolean relative_seek = FALSE;
|
|
gboolean seekseg_configured = FALSE;
|
|
GstSegment seeksegment;
|
|
guint32 seqnum;
|
|
GstEvent *tevent;
|
|
|
|
GST_DEBUG_OBJECT (src, "doing seek: %" GST_PTR_FORMAT, event);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
dest_format = src->segment.format;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (event) {
|
|
gst_event_parse_seek (event, &rate, &seek_format, &flags,
|
|
&start_type, &start, &stop_type, &stop);
|
|
|
|
relative_seek = SEEK_TYPE_IS_RELATIVE (start_type) ||
|
|
SEEK_TYPE_IS_RELATIVE (stop_type);
|
|
|
|
if (dest_format != seek_format && !relative_seek) {
|
|
/* If we have an ABSOLUTE position (SEEK_SET only), we can convert it
|
|
* here before taking the stream lock, otherwise we must convert it later,
|
|
* once we have the stream lock and can read the last configures segment
|
|
* start and stop positions */
|
|
gst_segment_init (&seeksegment, dest_format);
|
|
|
|
if (!gst_base_src_prepare_seek_segment (src, event, &seeksegment))
|
|
goto prepare_failed;
|
|
|
|
seekseg_configured = TRUE;
|
|
}
|
|
|
|
flush = flags & GST_SEEK_FLAG_FLUSH;
|
|
seqnum = gst_event_get_seqnum (event);
|
|
} else {
|
|
flush = FALSE;
|
|
/* get next seqnum */
|
|
seqnum = gst_util_seqnum_next ();
|
|
}
|
|
|
|
/* send flush start */
|
|
if (flush) {
|
|
tevent = gst_event_new_flush_start ();
|
|
gst_event_set_seqnum (tevent, seqnum);
|
|
gst_pad_push_event (src->srcpad, tevent);
|
|
} else
|
|
gst_pad_pause_task (src->srcpad);
|
|
|
|
/* unblock streaming thread. */
|
|
if (unlock)
|
|
gst_base_src_set_flushing (src, TRUE);
|
|
|
|
/* grab streaming lock, this should eventually be possible, either
|
|
* because the task is paused, our streaming thread stopped
|
|
* or because our peer is flushing. */
|
|
GST_PAD_STREAM_LOCK (src->srcpad);
|
|
if (G_UNLIKELY (src->priv->seqnum == seqnum)) {
|
|
/* we have seen this event before, issue a warning for now */
|
|
GST_WARNING_OBJECT (src, "duplicate event found %" G_GUINT32_FORMAT,
|
|
seqnum);
|
|
} else {
|
|
src->priv->seqnum = seqnum;
|
|
GST_DEBUG_OBJECT (src, "seek with seqnum %" G_GUINT32_FORMAT, seqnum);
|
|
}
|
|
|
|
if (unlock)
|
|
gst_base_src_set_flushing (src, FALSE);
|
|
|
|
/* 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 succeeds. */
|
|
if (!seekseg_configured) {
|
|
memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
|
|
|
|
/* now configure the final seek segment */
|
|
if (event) {
|
|
if (seeksegment.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_src_prepare_seek_segment (src, event, &seeksegment)) {
|
|
GST_DEBUG_OBJECT (src, "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_do_seek (&seeksegment, rate, seek_format, flags,
|
|
start_type, start, stop_type, stop, &update);
|
|
}
|
|
}
|
|
/* Else, no seek event passed, so we're just (re)starting the
|
|
current segment. */
|
|
}
|
|
|
|
if (res) {
|
|
GST_DEBUG_OBJECT (src, "segment configured from %" G_GINT64_FORMAT
|
|
" to %" G_GINT64_FORMAT ", position %" G_GINT64_FORMAT,
|
|
seeksegment.start, seeksegment.stop, seeksegment.position);
|
|
|
|
/* do the seek, segment.position contains the new position. */
|
|
res = gst_base_src_do_seek (src, &seeksegment);
|
|
}
|
|
|
|
/* and prepare to continue streaming */
|
|
if (flush) {
|
|
tevent = gst_event_new_flush_stop (TRUE);
|
|
gst_event_set_seqnum (tevent, seqnum);
|
|
/* send flush stop, peer will accept data and events again. We
|
|
* are not yet providing data as we still have the STREAM_LOCK. */
|
|
gst_pad_push_event (src->srcpad, tevent);
|
|
}
|
|
|
|
/* The subclass must have converted the segment to the processing format
|
|
* by now */
|
|
if (res && seeksegment.format != dest_format) {
|
|
GST_DEBUG_OBJECT (src, "Subclass failed to prepare a seek segment "
|
|
"in the correct format. Aborting seek.");
|
|
res = FALSE;
|
|
}
|
|
|
|
/* if the seek was successful, we update our real segment and push
|
|
* out the new segment. */
|
|
if (res) {
|
|
GST_OBJECT_LOCK (src);
|
|
memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (seeksegment.flags & GST_SEGMENT_FLAG_SEGMENT) {
|
|
GstMessage *message;
|
|
|
|
message = gst_message_new_segment_start (GST_OBJECT (src),
|
|
seeksegment.format, seeksegment.position);
|
|
gst_message_set_seqnum (message, seqnum);
|
|
|
|
gst_element_post_message (GST_ELEMENT (src), message);
|
|
}
|
|
|
|
src->priv->segment_pending = TRUE;
|
|
src->priv->segment_seqnum = seqnum;
|
|
}
|
|
|
|
src->priv->discont = TRUE;
|
|
src->running = TRUE;
|
|
/* and restart the task in case it got paused explicitly or by
|
|
* the FLUSH_START event we pushed out. */
|
|
tres = gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop,
|
|
src->srcpad, NULL);
|
|
if (res && !tres)
|
|
res = FALSE;
|
|
|
|
/* and release the lock again so we can continue streaming */
|
|
GST_PAD_STREAM_UNLOCK (src->srcpad);
|
|
|
|
return res;
|
|
|
|
/* ERROR */
|
|
prepare_failed:
|
|
GST_DEBUG_OBJECT (src, "Preparing the seek failed before flushing. "
|
|
"Aborting seek");
|
|
return FALSE;
|
|
}
|
|
|
|
/* all events send to this element directly. This is mainly done from the
|
|
* application.
|
|
*/
|
|
static gboolean
|
|
gst_base_src_send_event (GstElement * element, GstEvent * event)
|
|
{
|
|
GstBaseSrc *src;
|
|
gboolean result = FALSE;
|
|
GstBaseSrcClass *bclass;
|
|
|
|
src = GST_BASE_SRC (element);
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
GST_DEBUG_OBJECT (src, "handling event %p %" GST_PTR_FORMAT, event, event);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
/* bidirectional events */
|
|
case GST_EVENT_FLUSH_START:
|
|
GST_DEBUG_OBJECT (src, "pushing flush-start event downstream");
|
|
|
|
result = gst_pad_push_event (src->srcpad, event);
|
|
gst_base_src_set_flushing (src, TRUE);
|
|
event = NULL;
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
{
|
|
gboolean start;
|
|
|
|
GST_PAD_STREAM_LOCK (src->srcpad);
|
|
gst_base_src_set_flushing (src, FALSE);
|
|
|
|
GST_DEBUG_OBJECT (src, "pushing flush-stop event downstream");
|
|
result = gst_pad_push_event (src->srcpad, event);
|
|
|
|
/* For external flush, restart the task .. */
|
|
GST_LIVE_LOCK (src);
|
|
src->priv->segment_pending = TRUE;
|
|
|
|
GST_OBJECT_LOCK (src->srcpad);
|
|
start = (GST_PAD_MODE (src->srcpad) == GST_PAD_MODE_PUSH);
|
|
GST_OBJECT_UNLOCK (src->srcpad);
|
|
|
|
/* ... and for live sources, only if in playing state */
|
|
if (src->is_live) {
|
|
if (!src->live_running)
|
|
start = FALSE;
|
|
}
|
|
|
|
if (start)
|
|
gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop,
|
|
src->srcpad, NULL);
|
|
|
|
GST_LIVE_UNLOCK (src);
|
|
GST_PAD_STREAM_UNLOCK (src->srcpad);
|
|
|
|
event = NULL;
|
|
break;
|
|
}
|
|
|
|
/* downstream serialized events */
|
|
case GST_EVENT_EOS:
|
|
{
|
|
gboolean push_mode;
|
|
|
|
/* queue EOS and make sure the task or pull function performs the EOS
|
|
* actions.
|
|
*
|
|
* For push mode, This will be done in 3 steps. It is required to not
|
|
* block here as gst_element_send_event() will hold the STATE_LOCK, hence
|
|
* blocking would prevent asynchronous state change to complete.
|
|
*
|
|
* 1. We stop the streaming thread
|
|
* 2. We set the pending eos
|
|
* 3. We start the streaming thread again, so it is performed
|
|
* asynchronously.
|
|
*
|
|
* For pull mode, we simply mark the pending EOS without flushing.
|
|
*/
|
|
|
|
GST_OBJECT_LOCK (src->srcpad);
|
|
push_mode = GST_PAD_MODE (src->srcpad) == GST_PAD_MODE_PUSH;
|
|
GST_OBJECT_UNLOCK (src->srcpad);
|
|
|
|
if (push_mode) {
|
|
gst_base_src_set_flushing (src, TRUE);
|
|
|
|
GST_PAD_STREAM_LOCK (src->srcpad);
|
|
gst_base_src_set_flushing (src, FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
g_atomic_int_set (&src->priv->has_pending_eos, TRUE);
|
|
if (src->priv->pending_eos)
|
|
gst_event_unref (src->priv->pending_eos);
|
|
src->priv->pending_eos = event;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"EOS marked, start task for asynchronous handling");
|
|
gst_pad_start_task (src->srcpad, (GstTaskFunction) gst_base_src_loop,
|
|
src->srcpad, NULL);
|
|
|
|
GST_PAD_STREAM_UNLOCK (src->srcpad);
|
|
} else {
|
|
/* In pull mode, we need not to return flushing to downstream, though
|
|
* the stream lock is not kept after getrange was unblocked */
|
|
GST_OBJECT_LOCK (src);
|
|
g_atomic_int_set (&src->priv->has_pending_eos, TRUE);
|
|
if (src->priv->pending_eos)
|
|
gst_event_unref (src->priv->pending_eos);
|
|
src->priv->pending_eos = event;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
gst_base_src_set_pool_flushing (src, TRUE);
|
|
if (bclass->unlock)
|
|
bclass->unlock (src);
|
|
|
|
GST_PAD_STREAM_LOCK (src->srcpad);
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (src);
|
|
gst_base_src_set_pool_flushing (src, TRUE);
|
|
GST_PAD_STREAM_UNLOCK (src->srcpad);
|
|
}
|
|
|
|
|
|
event = NULL;
|
|
result = TRUE;
|
|
break;
|
|
}
|
|
case GST_EVENT_SEGMENT:
|
|
/* sending random SEGMENT downstream can break sync. */
|
|
break;
|
|
case GST_EVENT_TAG:
|
|
case GST_EVENT_SINK_MESSAGE:
|
|
case GST_EVENT_CUSTOM_DOWNSTREAM:
|
|
case GST_EVENT_CUSTOM_BOTH:
|
|
case GST_EVENT_PROTECTION:
|
|
/* Insert TAG, CUSTOM_DOWNSTREAM, CUSTOM_BOTH, PROTECTION in the dataflow */
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->pending_events =
|
|
g_list_append (src->priv->pending_events, event);
|
|
g_atomic_int_set (&src->priv->have_events, TRUE);
|
|
GST_OBJECT_UNLOCK (src);
|
|
event = NULL;
|
|
result = TRUE;
|
|
break;
|
|
case GST_EVENT_BUFFERSIZE:
|
|
/* does not seem to make much sense currently */
|
|
break;
|
|
|
|
/* upstream events */
|
|
case GST_EVENT_QOS:
|
|
/* elements should override send_event and do something */
|
|
break;
|
|
case GST_EVENT_SEEK:
|
|
{
|
|
gboolean started;
|
|
|
|
GST_OBJECT_LOCK (src->srcpad);
|
|
if (GST_PAD_MODE (src->srcpad) == GST_PAD_MODE_PULL)
|
|
goto wrong_mode;
|
|
started = GST_PAD_MODE (src->srcpad) == GST_PAD_MODE_PUSH;
|
|
GST_OBJECT_UNLOCK (src->srcpad);
|
|
|
|
if (started) {
|
|
GST_DEBUG_OBJECT (src, "performing seek");
|
|
/* when we are running in push mode, we can execute the
|
|
* seek right now. */
|
|
result = gst_base_src_perform_seek (src, event, TRUE);
|
|
} else {
|
|
GstEvent **event_p;
|
|
|
|
/* else we store the event and execute the seek when we
|
|
* get activated */
|
|
GST_OBJECT_LOCK (src);
|
|
GST_DEBUG_OBJECT (src, "queueing seek");
|
|
event_p = &src->pending_seek;
|
|
gst_event_replace ((GstEvent **) event_p, event);
|
|
GST_OBJECT_UNLOCK (src);
|
|
/* assume the seek will work */
|
|
result = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
case GST_EVENT_NAVIGATION:
|
|
/* could make sense for elements that do something with navigation events
|
|
* but then they would need to override the send_event function */
|
|
break;
|
|
case GST_EVENT_LATENCY:
|
|
/* does not seem to make sense currently */
|
|
break;
|
|
|
|
/* custom events */
|
|
case GST_EVENT_CUSTOM_UPSTREAM:
|
|
/* override send_event if you want this */
|
|
break;
|
|
case GST_EVENT_CUSTOM_DOWNSTREAM_OOB:
|
|
case GST_EVENT_CUSTOM_BOTH_OOB:
|
|
/* insert a random custom event into the pipeline */
|
|
GST_DEBUG_OBJECT (src, "pushing custom OOB event downstream");
|
|
result = gst_pad_push_event (src->srcpad, event);
|
|
/* we gave away the ref to the event in the push */
|
|
event = NULL;
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
done:
|
|
/* if we still have a ref to the event, unref it now */
|
|
if (event)
|
|
gst_event_unref (event);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_mode:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "cannot perform seek when operating in pull mode");
|
|
GST_OBJECT_UNLOCK (src->srcpad);
|
|
result = FALSE;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_seekable (GstBaseSrc * src)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
if (bclass->is_seekable)
|
|
return bclass->is_seekable (src);
|
|
else
|
|
return FALSE;
|
|
}
|
|
|
|
static void
|
|
gst_base_src_update_qos (GstBaseSrc * src,
|
|
gdouble proportion, GstClockTimeDiff diff, GstClockTime timestamp)
|
|
{
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_QOS, src,
|
|
"qos: proportion: %lf, diff %" G_GINT64_FORMAT ", timestamp %"
|
|
GST_TIME_FORMAT, proportion, diff, GST_TIME_ARGS (timestamp));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
src->priv->proportion = proportion;
|
|
src->priv->earliest_time = timestamp + diff;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_base_src_default_event (GstBaseSrc * src, GstEvent * event)
|
|
{
|
|
gboolean result;
|
|
|
|
GST_DEBUG_OBJECT (src, "handle event %" GST_PTR_FORMAT, event);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
/* is normally called when in push mode */
|
|
if (!gst_base_src_seekable (src))
|
|
goto not_seekable;
|
|
|
|
result = gst_base_src_perform_seek (src, event, TRUE);
|
|
break;
|
|
case GST_EVENT_FLUSH_START:
|
|
/* cancel any blocking getrange, is normally called
|
|
* when in pull mode. */
|
|
result = gst_base_src_set_flushing (src, TRUE);
|
|
break;
|
|
case GST_EVENT_FLUSH_STOP:
|
|
result = gst_base_src_set_flushing (src, FALSE);
|
|
break;
|
|
case GST_EVENT_QOS:
|
|
{
|
|
gdouble proportion;
|
|
GstClockTimeDiff diff;
|
|
GstClockTime timestamp;
|
|
|
|
gst_event_parse_qos (event, NULL, &proportion, &diff, ×tamp);
|
|
gst_base_src_update_qos (src, proportion, diff, timestamp);
|
|
result = TRUE;
|
|
break;
|
|
}
|
|
case GST_EVENT_RECONFIGURE:
|
|
result = TRUE;
|
|
break;
|
|
case GST_EVENT_LATENCY:
|
|
result = TRUE;
|
|
break;
|
|
default:
|
|
result = FALSE;
|
|
break;
|
|
}
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
not_seekable:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "is not seekable");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = FALSE;
|
|
|
|
src = GST_BASE_SRC (parent);
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
if (bclass->event) {
|
|
if (!(result = bclass->event (src, event)))
|
|
goto subclass_failed;
|
|
}
|
|
|
|
done:
|
|
gst_event_unref (event);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
subclass_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "subclass refused event");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_src_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstBaseSrc *src;
|
|
|
|
src = GST_BASE_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_BLOCKSIZE:
|
|
gst_base_src_set_blocksize (src, g_value_get_uint (value));
|
|
break;
|
|
case PROP_NUM_BUFFERS:
|
|
src->num_buffers = g_value_get_int (value);
|
|
break;
|
|
#ifndef GST_REMOVE_DEPRECATED
|
|
case PROP_TYPEFIND:
|
|
src->typefind = g_value_get_boolean (value);
|
|
break;
|
|
#endif
|
|
case PROP_DO_TIMESTAMP:
|
|
gst_base_src_set_do_timestamp (src, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_AUTOMATIC_EOS:
|
|
gst_base_src_set_automatic_eos (src, g_value_get_boolean (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_src_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstBaseSrc *src;
|
|
|
|
src = GST_BASE_SRC (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_BLOCKSIZE:
|
|
g_value_set_uint (value, gst_base_src_get_blocksize (src));
|
|
break;
|
|
case PROP_NUM_BUFFERS:
|
|
g_value_set_int (value, src->num_buffers);
|
|
break;
|
|
#ifndef GST_REMOVE_DEPRECATED
|
|
case PROP_TYPEFIND:
|
|
g_value_set_boolean (value, src->typefind);
|
|
break;
|
|
#endif
|
|
case PROP_DO_TIMESTAMP:
|
|
g_value_set_boolean (value, gst_base_src_get_do_timestamp (src));
|
|
break;
|
|
case PROP_AUTOMATIC_EOS:
|
|
g_value_set_boolean (value, g_atomic_int_get (&src->priv->automatic_eos));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* with STREAM_LOCK and LOCK */
|
|
static GstClockReturn
|
|
gst_base_src_wait (GstBaseSrc * basesrc, GstClock * clock, GstClockTime time)
|
|
{
|
|
GstClockReturn ret;
|
|
GstClockID id;
|
|
|
|
id = gst_clock_new_single_shot_id (clock, time);
|
|
|
|
basesrc->clock_id = id;
|
|
/* release the live lock while waiting */
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
ret = gst_clock_id_wait (id, NULL);
|
|
|
|
GST_LIVE_LOCK (basesrc);
|
|
gst_clock_id_unref (id);
|
|
basesrc->clock_id = NULL;
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* perform synchronisation on a buffer.
|
|
* with STREAM_LOCK.
|
|
*/
|
|
static GstClockReturn
|
|
gst_base_src_do_sync (GstBaseSrc * basesrc, GstBuffer * buffer)
|
|
{
|
|
GstClockReturn result;
|
|
GstClockTime start, end;
|
|
GstBaseSrcClass *bclass;
|
|
GstClockTime base_time;
|
|
GstClock *clock;
|
|
GstClockTime now = GST_CLOCK_TIME_NONE, pts, dts, timestamp;
|
|
gboolean do_timestamp, first, pseudo_live, is_live;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
start = end = -1;
|
|
if (bclass->get_times)
|
|
bclass->get_times (basesrc, buffer, &start, &end);
|
|
|
|
/* get buffer timestamp */
|
|
dts = GST_BUFFER_DTS (buffer);
|
|
pts = GST_BUFFER_PTS (buffer);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (dts))
|
|
timestamp = dts;
|
|
else
|
|
timestamp = pts;
|
|
|
|
/* grab the lock to prepare for clocking and calculate the startup
|
|
* latency. */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
|
|
is_live = basesrc->is_live;
|
|
/* if we are asked to sync against the clock we are a pseudo live element */
|
|
pseudo_live = (start != -1 && is_live);
|
|
/* check for the first buffer */
|
|
first = (basesrc->priv->latency == -1);
|
|
|
|
if (timestamp != -1 && pseudo_live) {
|
|
GstClockTime latency;
|
|
|
|
/* we have a timestamp and a sync time, latency is the diff */
|
|
if (timestamp <= start)
|
|
latency = start - timestamp;
|
|
else
|
|
latency = 0;
|
|
|
|
if (first) {
|
|
GST_DEBUG_OBJECT (basesrc, "pseudo_live with latency %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (latency));
|
|
/* first time we calculate latency, just configure */
|
|
basesrc->priv->latency = latency;
|
|
} else {
|
|
if (basesrc->priv->latency != latency) {
|
|
/* we have a new latency, FIXME post latency message */
|
|
basesrc->priv->latency = latency;
|
|
GST_DEBUG_OBJECT (basesrc, "latency changed to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (latency));
|
|
}
|
|
}
|
|
} else if (first) {
|
|
GST_DEBUG_OBJECT (basesrc, "no latency needed, live %d, sync %d",
|
|
is_live, start != -1);
|
|
basesrc->priv->latency = 0;
|
|
}
|
|
|
|
/* get clock, if no clock, we can't sync or do timestamps */
|
|
if ((clock = GST_ELEMENT_CLOCK (basesrc)) == NULL)
|
|
goto no_clock;
|
|
else
|
|
gst_object_ref (clock);
|
|
|
|
base_time = GST_ELEMENT_CAST (basesrc)->base_time;
|
|
|
|
do_timestamp = basesrc->priv->do_timestamp;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
/* first buffer, calculate the timestamp offset */
|
|
if (first) {
|
|
GstClockTime running_time;
|
|
|
|
now = gst_clock_get_time (clock);
|
|
running_time = now - base_time;
|
|
|
|
GST_LOG_OBJECT (basesrc,
|
|
"startup PTS: %" GST_TIME_FORMAT ", DTS %" GST_TIME_FORMAT
|
|
", running_time %" GST_TIME_FORMAT, GST_TIME_ARGS (pts),
|
|
GST_TIME_ARGS (dts), GST_TIME_ARGS (running_time));
|
|
|
|
if (pseudo_live && timestamp != -1) {
|
|
/* live source and we need to sync, add startup latency to all timestamps
|
|
* to get the real running_time. Live sources should always timestamp
|
|
* according to the current running time. */
|
|
basesrc->priv->ts_offset = GST_CLOCK_DIFF (timestamp, running_time);
|
|
|
|
GST_LOG_OBJECT (basesrc, "live with sync, ts_offset %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (basesrc->priv->ts_offset));
|
|
} else {
|
|
basesrc->priv->ts_offset = 0;
|
|
GST_LOG_OBJECT (basesrc, "no timestamp offset needed");
|
|
}
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (dts)) {
|
|
if (do_timestamp) {
|
|
dts = running_time;
|
|
} else if (!GST_CLOCK_TIME_IS_VALID (pts)) {
|
|
if (GST_CLOCK_TIME_IS_VALID (basesrc->segment.start)) {
|
|
dts = basesrc->segment.start;
|
|
} else {
|
|
dts = 0;
|
|
}
|
|
}
|
|
GST_BUFFER_DTS (buffer) = dts;
|
|
|
|
GST_LOG_OBJECT (basesrc, "created DTS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (dts));
|
|
}
|
|
} else {
|
|
/* not the first buffer, the timestamp is the diff between the clock and
|
|
* base_time */
|
|
if (do_timestamp && !GST_CLOCK_TIME_IS_VALID (dts)) {
|
|
now = gst_clock_get_time (clock);
|
|
|
|
dts = now - base_time;
|
|
GST_BUFFER_DTS (buffer) = dts;
|
|
|
|
GST_LOG_OBJECT (basesrc, "created DTS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (dts));
|
|
}
|
|
}
|
|
if (!GST_CLOCK_TIME_IS_VALID (pts)) {
|
|
if (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT))
|
|
pts = dts;
|
|
|
|
GST_BUFFER_PTS (buffer) = dts;
|
|
|
|
GST_LOG_OBJECT (basesrc, "created PTS %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (pts));
|
|
}
|
|
|
|
/* if we don't have a buffer timestamp, we don't sync */
|
|
if (!GST_CLOCK_TIME_IS_VALID (start))
|
|
goto no_sync;
|
|
|
|
if (is_live) {
|
|
/* for pseudo live sources, add our ts_offset to the timestamp */
|
|
if (GST_CLOCK_TIME_IS_VALID (pts))
|
|
GST_BUFFER_PTS (buffer) += basesrc->priv->ts_offset;
|
|
if (GST_CLOCK_TIME_IS_VALID (dts))
|
|
GST_BUFFER_DTS (buffer) += basesrc->priv->ts_offset;
|
|
start += basesrc->priv->ts_offset;
|
|
}
|
|
|
|
GST_LOG_OBJECT (basesrc,
|
|
"waiting for clock, base time %" GST_TIME_FORMAT
|
|
", stream_start %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (base_time), GST_TIME_ARGS (start));
|
|
|
|
result = gst_base_src_wait (basesrc, clock, start + base_time);
|
|
|
|
gst_object_unref (clock);
|
|
|
|
GST_LOG_OBJECT (basesrc, "clock entry done: %d", result);
|
|
|
|
return result;
|
|
|
|
/* special cases */
|
|
no_clock:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "we have no clock");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
return GST_CLOCK_OK;
|
|
}
|
|
no_sync:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "no sync needed");
|
|
gst_object_unref (clock);
|
|
return GST_CLOCK_OK;
|
|
}
|
|
}
|
|
|
|
/* Called with STREAM_LOCK and LIVE_LOCK */
|
|
static gboolean
|
|
gst_base_src_update_length (GstBaseSrc * src, guint64 offset, guint * length,
|
|
gboolean force)
|
|
{
|
|
guint64 size, maxsize;
|
|
GstBaseSrcClass *bclass;
|
|
gint64 stop;
|
|
|
|
/* only operate if we are working with bytes */
|
|
if (src->segment.format != GST_FORMAT_BYTES)
|
|
return TRUE;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
stop = src->segment.stop;
|
|
/* get total file size */
|
|
size = src->segment.duration;
|
|
|
|
/* when not doing automatic EOS, just use the stop position. We don't use
|
|
* the size to check for EOS */
|
|
if (!g_atomic_int_get (&src->priv->automatic_eos))
|
|
maxsize = stop;
|
|
/* Otherwise, the max amount of bytes to read is the total
|
|
* size or up to the segment.stop if present. */
|
|
else if (stop != -1)
|
|
maxsize = size != -1 ? MIN (size, stop) : stop;
|
|
else
|
|
maxsize = size;
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"reading offset %" G_GUINT64_FORMAT ", length %u, size %" G_GINT64_FORMAT
|
|
", segment.stop %" G_GINT64_FORMAT ", maxsize %" G_GINT64_FORMAT, offset,
|
|
*length, size, stop, maxsize);
|
|
|
|
/* check size if we have one */
|
|
if (maxsize != -1) {
|
|
/* if we run past the end, check if the file became bigger and
|
|
* retry. Mind wrap when checking. */
|
|
if (G_UNLIKELY (offset >= maxsize || offset + *length >= maxsize || force)) {
|
|
/* see if length of the file changed */
|
|
if (bclass->get_size)
|
|
if (!bclass->get_size (src, &size))
|
|
size = -1;
|
|
|
|
/* when not doing automatic EOS, just use the stop position. We don't use
|
|
* the size to check for EOS */
|
|
if (!g_atomic_int_get (&src->priv->automatic_eos))
|
|
maxsize = stop;
|
|
/* Otherwise, the max amount of bytes to read is the total
|
|
* size or up to the segment.stop if present. */
|
|
else if (stop != -1)
|
|
maxsize = size != -1 ? MIN (size, stop) : stop;
|
|
else
|
|
maxsize = size;
|
|
|
|
if (maxsize != -1) {
|
|
/* if we are at or past the end, EOS */
|
|
if (G_UNLIKELY (offset >= maxsize))
|
|
goto unexpected_length;
|
|
|
|
/* else we can clip to the end */
|
|
if (G_UNLIKELY (offset + *length >= maxsize))
|
|
*length = maxsize - offset;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* keep track of current duration. segment is in bytes, we checked
|
|
* that above. */
|
|
GST_OBJECT_LOCK (src);
|
|
src->segment.duration = size;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
unexpected_length:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "processing at or past EOS");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* must be called with LIVE_LOCK */
|
|
static GstFlowReturn
|
|
gst_base_src_get_range (GstBaseSrc * src, guint64 offset, guint length,
|
|
GstBuffer ** buf)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstBaseSrcClass *bclass;
|
|
GstClockReturn status;
|
|
GstBuffer *res_buf;
|
|
GstBuffer *in_buf;
|
|
gboolean own_res_buf;
|
|
gboolean pending_unlock_stop;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (src);
|
|
|
|
again:
|
|
if (src->is_live) {
|
|
if (G_UNLIKELY (!src->live_running)) {
|
|
ret = gst_base_src_wait_playing_unlocked (src);
|
|
if (ret != GST_FLOW_OK)
|
|
goto stopped;
|
|
}
|
|
}
|
|
|
|
if (G_UNLIKELY (!GST_BASE_SRC_IS_STARTED (src)
|
|
&& !GST_BASE_SRC_IS_STARTING (src)))
|
|
goto not_started;
|
|
|
|
if (G_UNLIKELY (!bclass->create))
|
|
goto no_function;
|
|
|
|
if (G_UNLIKELY (!gst_base_src_update_length (src, offset, &length, FALSE)))
|
|
goto unexpected_length;
|
|
|
|
/* track position */
|
|
GST_OBJECT_LOCK (src);
|
|
if (src->segment.format == GST_FORMAT_BYTES)
|
|
src->segment.position = offset;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
/* normally we don't count buffers */
|
|
if (G_UNLIKELY (src->num_buffers_left >= 0)) {
|
|
if (src->num_buffers_left == 0)
|
|
goto reached_num_buffers;
|
|
else
|
|
src->num_buffers_left--;
|
|
}
|
|
|
|
/* don't enter the create function if a pending EOS event was set. For the
|
|
* logic of the has_pending_eos, check the event function of this class. */
|
|
if (G_UNLIKELY (g_atomic_int_get (&src->priv->has_pending_eos))) {
|
|
src->priv->forced_eos = TRUE;
|
|
goto eos;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"calling create offset %" G_GUINT64_FORMAT " length %u, time %"
|
|
G_GINT64_FORMAT, offset, length, src->segment.time);
|
|
|
|
res_buf = in_buf = *buf;
|
|
own_res_buf = (*buf == NULL);
|
|
|
|
retry_create:
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
/* Undo our unlocking initiated from set_playing function. This is done here
|
|
* since we require the stream lock to call this virtual function. */
|
|
GST_OBJECT_LOCK (src);
|
|
pending_unlock_stop = src->priv->pending_unlock_stop;
|
|
src->priv->pending_unlock_stop = FALSE;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
if (pending_unlock_stop && bclass->unlock_stop)
|
|
bclass->unlock_stop (src);
|
|
|
|
ret = bclass->create (src, offset, length, &res_buf);
|
|
GST_LIVE_LOCK (src);
|
|
|
|
/* As we released the LIVE_LOCK, the state may have changed */
|
|
if (src->is_live) {
|
|
if (G_UNLIKELY (!src->live_running)) {
|
|
GstFlowReturn wait_ret;
|
|
|
|
/* no need keep old buffer while in pause */
|
|
if (ret == GST_FLOW_OK && own_res_buf)
|
|
gst_clear_buffer (&res_buf);
|
|
gst_clear_buffer_list (&src->priv->pending_bufferlist);
|
|
|
|
wait_ret = gst_base_src_wait_playing_unlocked (src);
|
|
if (wait_ret != GST_FLOW_OK) {
|
|
ret = wait_ret;
|
|
goto stopped;
|
|
}
|
|
|
|
goto retry_create;
|
|
}
|
|
}
|
|
|
|
/* The create function could be unlocked because we have a pending EOS. It's
|
|
* possible that we have a valid buffer from create that we need to
|
|
* discard when the create function returned _OK. */
|
|
if (G_UNLIKELY (g_atomic_int_get (&src->priv->has_pending_eos))) {
|
|
if (ret == GST_FLOW_OK) {
|
|
if (own_res_buf)
|
|
gst_clear_buffer (&res_buf);
|
|
gst_clear_buffer_list (&src->priv->pending_bufferlist);
|
|
}
|
|
src->priv->forced_eos = TRUE;
|
|
goto eos;
|
|
}
|
|
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK))
|
|
goto not_ok;
|
|
|
|
/* fallback in case the create function didn't fill a provided buffer */
|
|
if (in_buf != NULL && res_buf != in_buf) {
|
|
GstMapInfo info;
|
|
gsize copied_size;
|
|
|
|
GST_CAT_DEBUG_OBJECT (GST_CAT_PERFORMANCE, src, "create function didn't "
|
|
"fill the provided buffer, copying");
|
|
|
|
if (!gst_buffer_map (in_buf, &info, GST_MAP_WRITE))
|
|
goto map_failed;
|
|
|
|
copied_size = gst_buffer_extract (res_buf, 0, info.data, info.size);
|
|
gst_buffer_unmap (in_buf, &info);
|
|
gst_buffer_set_size (in_buf, copied_size);
|
|
|
|
gst_buffer_copy_into (in_buf, res_buf, GST_BUFFER_COPY_METADATA, 0, -1);
|
|
|
|
gst_buffer_unref (res_buf);
|
|
res_buf = in_buf;
|
|
}
|
|
|
|
if (res_buf == NULL) {
|
|
GstBufferList *pending_list = src->priv->pending_bufferlist;
|
|
|
|
if (pending_list == NULL || gst_buffer_list_length (pending_list) == 0)
|
|
goto null_buffer;
|
|
|
|
res_buf = gst_buffer_list_get_writable (pending_list, 0);
|
|
own_res_buf = FALSE;
|
|
}
|
|
|
|
/* no timestamp set and we are at offset 0, we can timestamp with 0 */
|
|
if (offset == 0 && src->segment.time == 0
|
|
&& GST_BUFFER_DTS (res_buf) == -1 && !src->is_live) {
|
|
GST_DEBUG_OBJECT (src, "setting first timestamp to 0");
|
|
res_buf = gst_buffer_make_writable (res_buf);
|
|
GST_BUFFER_DTS (res_buf) = 0;
|
|
}
|
|
|
|
/* now sync before pushing the buffer */
|
|
status = gst_base_src_do_sync (src, res_buf);
|
|
|
|
/* waiting for the clock could have made us flushing */
|
|
if (G_UNLIKELY (src->priv->flushing))
|
|
goto flushing;
|
|
|
|
switch (status) {
|
|
case GST_CLOCK_EARLY:
|
|
/* the buffer is too late. We currently don't drop the buffer. */
|
|
GST_DEBUG_OBJECT (src, "buffer too late!, returning anyway");
|
|
break;
|
|
case GST_CLOCK_OK:
|
|
/* buffer synchronised properly */
|
|
GST_DEBUG_OBJECT (src, "buffer ok");
|
|
break;
|
|
case GST_CLOCK_UNSCHEDULED:
|
|
/* this case is triggered when we were waiting for the clock and
|
|
* it got unlocked because we did a state change. In any case, get rid of
|
|
* the buffer. */
|
|
if (own_res_buf)
|
|
gst_buffer_unref (res_buf);
|
|
|
|
if (!src->live_running) {
|
|
/* We return FLUSHING when we are not running to stop the dataflow also
|
|
* get rid of the produced buffer. */
|
|
GST_DEBUG_OBJECT (src,
|
|
"clock was unscheduled (%d), returning FLUSHING", status);
|
|
ret = GST_FLOW_FLUSHING;
|
|
} else {
|
|
/* If we are running when this happens, we quickly switched between
|
|
* pause and playing. We try to produce a new buffer */
|
|
GST_DEBUG_OBJECT (src,
|
|
"clock was unscheduled (%d), but we are running", status);
|
|
goto again;
|
|
}
|
|
break;
|
|
default:
|
|
/* all other result values are unexpected and errors */
|
|
GST_ELEMENT_ERROR (src, CORE, CLOCK,
|
|
(_("Internal clock error.")),
|
|
("clock returned unexpected return value %d", status));
|
|
if (own_res_buf)
|
|
gst_buffer_unref (res_buf);
|
|
ret = GST_FLOW_ERROR;
|
|
break;
|
|
}
|
|
if (G_LIKELY (ret == GST_FLOW_OK))
|
|
*buf = res_buf;
|
|
|
|
return ret;
|
|
|
|
/* ERROR */
|
|
stopped:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "wait_playing returned %d (%s)", ret,
|
|
gst_flow_get_name (ret));
|
|
return ret;
|
|
}
|
|
not_ok:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "create returned %d (%s)", ret,
|
|
gst_flow_get_name (ret));
|
|
return ret;
|
|
}
|
|
map_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (src, RESOURCE, BUSY,
|
|
(_("Failed to map buffer.")),
|
|
("failed to map result buffer in WRITE mode"));
|
|
if (own_res_buf)
|
|
gst_buffer_unref (res_buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
not_started:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "getrange but not started");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
no_function:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "no create function");
|
|
return GST_FLOW_NOT_SUPPORTED;
|
|
}
|
|
unexpected_length:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "unexpected length %u (offset=%" G_GUINT64_FORMAT
|
|
", size=%" G_GINT64_FORMAT ")", length, offset, src->segment.duration);
|
|
return GST_FLOW_EOS;
|
|
}
|
|
reached_num_buffers:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "sent all buffers");
|
|
return GST_FLOW_EOS;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
if (own_res_buf)
|
|
gst_buffer_unref (res_buf);
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
eos:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are EOS");
|
|
return GST_FLOW_EOS;
|
|
}
|
|
null_buffer:
|
|
{
|
|
GST_ELEMENT_ERROR (src, STREAM, FAILED,
|
|
(_("Internal data flow error.")),
|
|
("Subclass %s neither returned a buffer nor submitted a buffer list "
|
|
"from its create function", G_OBJECT_TYPE_NAME (src)));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_base_src_getrange (GstPad * pad, GstObject * parent, guint64 offset,
|
|
guint length, GstBuffer ** buf)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstFlowReturn res;
|
|
|
|
src = GST_BASE_SRC_CAST (parent);
|
|
|
|
GST_LIVE_LOCK (src);
|
|
if (G_UNLIKELY (src->priv->flushing))
|
|
goto flushing;
|
|
|
|
res = gst_base_src_get_range (src, offset, length, buf);
|
|
|
|
done:
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
return res;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
res = GST_FLOW_FLUSHING;
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_is_random_access (GstBaseSrc * src)
|
|
{
|
|
/* we need to start the basesrc to check random access */
|
|
if (!GST_BASE_SRC_IS_STARTED (src)) {
|
|
GST_LOG_OBJECT (src, "doing start/stop to check get_range support");
|
|
if (G_LIKELY (gst_base_src_start (src))) {
|
|
if (gst_base_src_start_wait (src) != GST_FLOW_OK)
|
|
goto start_failed;
|
|
gst_base_src_stop (src);
|
|
}
|
|
}
|
|
|
|
return src->random_access;
|
|
|
|
/* ERRORS */
|
|
start_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "failed to start");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/* Called with STREAM_LOCK */
|
|
static void
|
|
gst_base_src_loop (GstPad * pad)
|
|
{
|
|
GstBaseSrc *src;
|
|
GstBuffer *buf = NULL;
|
|
GstFlowReturn ret;
|
|
gint64 position;
|
|
gboolean eos;
|
|
guint blocksize;
|
|
GList *pending_events = NULL, *tmp;
|
|
GstEvent *seg_event = NULL;
|
|
|
|
eos = FALSE;
|
|
|
|
src = GST_BASE_SRC (GST_OBJECT_PARENT (pad));
|
|
|
|
/* Just leave immediately if we're flushing */
|
|
GST_LIVE_LOCK (src);
|
|
if (G_UNLIKELY (src->priv->flushing || GST_PAD_IS_FLUSHING (pad)))
|
|
goto flushing;
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
/* Just return if EOS is pushed again, as the app might be unaware that an
|
|
* EOS have been sent already */
|
|
if (GST_PAD_IS_EOS (pad)) {
|
|
GST_DEBUG_OBJECT (src, "Pad is marked as EOS, pause the task");
|
|
gst_pad_pause_task (pad);
|
|
goto done;
|
|
}
|
|
|
|
gst_base_src_send_stream_start (src);
|
|
|
|
/* The stream-start event could've caused something to flush us */
|
|
GST_LIVE_LOCK (src);
|
|
if (G_UNLIKELY (src->priv->flushing || GST_PAD_IS_FLUSHING (pad)))
|
|
goto flushing;
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
/* check if we need to renegotiate */
|
|
if (gst_pad_check_reconfigure (pad)) {
|
|
if (!gst_base_src_negotiate_unlocked (src)) {
|
|
gst_pad_mark_reconfigure (pad);
|
|
if (GST_PAD_IS_FLUSHING (pad)) {
|
|
GST_LIVE_LOCK (src);
|
|
goto flushing;
|
|
} else {
|
|
goto negotiate_failed;
|
|
}
|
|
}
|
|
}
|
|
|
|
GST_LIVE_LOCK (src);
|
|
|
|
if (G_UNLIKELY (src->priv->flushing || GST_PAD_IS_FLUSHING (pad)))
|
|
goto flushing;
|
|
|
|
blocksize = src->blocksize;
|
|
|
|
/* if we operate in bytes, we can calculate an offset */
|
|
if (src->segment.format == GST_FORMAT_BYTES) {
|
|
position = src->segment.position;
|
|
/* for negative rates, start with subtracting the blocksize */
|
|
if (src->segment.rate < 0.0) {
|
|
/* we cannot go below segment.start */
|
|
if (position > src->segment.start + blocksize)
|
|
position -= blocksize;
|
|
else {
|
|
/* last block, remainder up to segment.start */
|
|
blocksize = position - src->segment.start;
|
|
position = src->segment.start;
|
|
}
|
|
}
|
|
} else
|
|
position = -1;
|
|
|
|
GST_LOG_OBJECT (src, "next_ts %" GST_TIME_FORMAT " size %u",
|
|
GST_TIME_ARGS (position), blocksize);
|
|
|
|
/* clean up just in case we got interrupted or so last time round */
|
|
if (src->priv->pending_bufferlist != NULL) {
|
|
gst_buffer_list_unref (src->priv->pending_bufferlist);
|
|
src->priv->pending_bufferlist = NULL;
|
|
}
|
|
|
|
ret = gst_base_src_get_range (src, position, blocksize, &buf);
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
|
|
GST_INFO_OBJECT (src, "pausing after gst_base_src_get_range() = %s",
|
|
gst_flow_get_name (ret));
|
|
GST_LIVE_UNLOCK (src);
|
|
goto pause;
|
|
}
|
|
|
|
/* Note: at this point buf might be a single buf returned which we own or
|
|
* the first buf of a pending buffer list submitted via submit_buffer_list(),
|
|
* in which case the buffer is owned by the pending buffer list and not us. */
|
|
g_assert (buf != NULL);
|
|
|
|
/* push events to close/start our segment before we push the buffer. */
|
|
if (G_UNLIKELY (src->priv->segment_pending)) {
|
|
/* generate the event but do not send until outside of live_lock */
|
|
seg_event = gst_event_new_segment (&src->segment);
|
|
|
|
gst_event_set_seqnum (seg_event, src->priv->segment_seqnum);
|
|
src->priv->segment_seqnum = gst_util_seqnum_next ();
|
|
src->priv->segment_pending = FALSE;
|
|
}
|
|
|
|
/* collect any pending events */
|
|
if (g_atomic_int_get (&src->priv->have_events)) {
|
|
GST_OBJECT_LOCK (src);
|
|
/* take the events */
|
|
pending_events = src->priv->pending_events;
|
|
src->priv->pending_events = NULL;
|
|
g_atomic_int_set (&src->priv->have_events, FALSE);
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
GST_LIVE_UNLOCK (src);
|
|
|
|
/* now outside the live_lock we can push the segment event */
|
|
if (G_UNLIKELY (seg_event))
|
|
gst_pad_push_event (pad, seg_event);
|
|
|
|
/* and the pending events if any */
|
|
if (G_UNLIKELY (pending_events != NULL)) {
|
|
for (tmp = pending_events; tmp; tmp = g_list_next (tmp)) {
|
|
GstEvent *ev = (GstEvent *) tmp->data;
|
|
gst_pad_push_event (pad, ev);
|
|
}
|
|
g_list_free (pending_events);
|
|
}
|
|
|
|
/* figure out the new position */
|
|
switch (src->segment.format) {
|
|
case GST_FORMAT_BYTES:
|
|
{
|
|
guint bufsize = gst_buffer_get_size (buf);
|
|
|
|
/* we subtracted above for negative rates */
|
|
if (src->segment.rate >= 0.0)
|
|
position += bufsize;
|
|
break;
|
|
}
|
|
case GST_FORMAT_TIME:
|
|
{
|
|
GstClockTime start, duration;
|
|
|
|
start = GST_BUFFER_TIMESTAMP (buf);
|
|
duration = GST_BUFFER_DURATION (buf);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (start))
|
|
position = start;
|
|
else
|
|
position = src->segment.position;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (duration)) {
|
|
if (src->segment.rate >= 0.0)
|
|
position += duration;
|
|
}
|
|
break;
|
|
}
|
|
case GST_FORMAT_DEFAULT:
|
|
if (src->segment.rate >= 0.0)
|
|
position = GST_BUFFER_OFFSET_END (buf);
|
|
else
|
|
position = GST_BUFFER_OFFSET (buf);
|
|
break;
|
|
default:
|
|
position = -1;
|
|
break;
|
|
}
|
|
if (position != -1) {
|
|
if (src->segment.rate >= 0.0) {
|
|
/* positive rate, check if we reached the stop */
|
|
if (src->segment.stop != -1) {
|
|
if (position >= src->segment.stop) {
|
|
if (g_atomic_int_get (&src->priv->automatic_eos))
|
|
eos = TRUE;
|
|
position = src->segment.stop;
|
|
}
|
|
}
|
|
} else {
|
|
/* negative rate, check if we reached the start. start is always set to
|
|
* something different from -1 */
|
|
if (position <= src->segment.start) {
|
|
if (g_atomic_int_get (&src->priv->automatic_eos))
|
|
eos = TRUE;
|
|
position = src->segment.start;
|
|
}
|
|
/* when going reverse, all buffers are DISCONT */
|
|
src->priv->discont = TRUE;
|
|
}
|
|
GST_OBJECT_LOCK (src);
|
|
src->segment.position = position;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
if (G_UNLIKELY (src->priv->discont)) {
|
|
GST_INFO_OBJECT (src, "marking pending DISCONT");
|
|
buf = gst_buffer_make_writable (buf);
|
|
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
|
|
src->priv->discont = FALSE;
|
|
}
|
|
|
|
/* push buffer or buffer list */
|
|
if (src->priv->pending_bufferlist != NULL) {
|
|
ret = gst_pad_push_list (pad, src->priv->pending_bufferlist);
|
|
src->priv->pending_bufferlist = NULL;
|
|
} else {
|
|
ret = gst_pad_push (pad, buf);
|
|
}
|
|
|
|
if (G_UNLIKELY (ret != GST_FLOW_OK)) {
|
|
if (ret == GST_FLOW_NOT_NEGOTIATED) {
|
|
goto not_negotiated;
|
|
}
|
|
GST_INFO_OBJECT (src, "pausing after gst_pad_push() = %s",
|
|
gst_flow_get_name (ret));
|
|
goto pause;
|
|
}
|
|
|
|
/* Segment pending means that a new segment was configured
|
|
* during this loop run */
|
|
if (G_UNLIKELY (eos && !src->priv->segment_pending)) {
|
|
GST_INFO_OBJECT (src, "pausing after end of segment");
|
|
ret = GST_FLOW_EOS;
|
|
goto pause;
|
|
}
|
|
|
|
done:
|
|
return;
|
|
|
|
/* special cases */
|
|
not_negotiated:
|
|
{
|
|
if (gst_pad_needs_reconfigure (pad)) {
|
|
GST_DEBUG_OBJECT (src, "Retrying to renegotiate");
|
|
return;
|
|
}
|
|
/* fallthrough when push returns NOT_NEGOTIATED and we don't have
|
|
* a pending negotiation request on our srcpad */
|
|
}
|
|
negotiate_failed:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "Not negotiated");
|
|
ret = GST_FLOW_NOT_NEGOTIATED;
|
|
goto pause;
|
|
}
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (src, "we are flushing");
|
|
GST_LIVE_UNLOCK (src);
|
|
ret = GST_FLOW_FLUSHING;
|
|
goto pause;
|
|
}
|
|
pause:
|
|
{
|
|
GstEvent *event;
|
|
|
|
GST_DEBUG_OBJECT (src, "pausing task, reason %s", gst_flow_get_name (ret));
|
|
src->running = FALSE;
|
|
gst_pad_pause_task (pad);
|
|
if (ret == GST_FLOW_EOS) {
|
|
gboolean flag_segment;
|
|
GstFormat format;
|
|
gint64 position;
|
|
|
|
flag_segment = (src->segment.flags & GST_SEGMENT_FLAG_SEGMENT) != 0;
|
|
format = src->segment.format;
|
|
position = src->segment.position;
|
|
|
|
/* perform EOS logic */
|
|
if (src->priv->forced_eos) {
|
|
g_assert (g_atomic_int_get (&src->priv->has_pending_eos));
|
|
GST_OBJECT_LOCK (src);
|
|
event = src->priv->pending_eos;
|
|
src->priv->pending_eos = NULL;
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
} else if (flag_segment) {
|
|
GstMessage *message;
|
|
|
|
message = gst_message_new_segment_done (GST_OBJECT_CAST (src),
|
|
format, position);
|
|
gst_message_set_seqnum (message, src->priv->seqnum);
|
|
gst_element_post_message (GST_ELEMENT_CAST (src), message);
|
|
event = gst_event_new_segment_done (format, position);
|
|
gst_event_set_seqnum (event, src->priv->seqnum);
|
|
|
|
} else {
|
|
event = gst_event_new_eos ();
|
|
gst_event_set_seqnum (event, src->priv->seqnum);
|
|
}
|
|
|
|
gst_pad_push_event (pad, event);
|
|
src->priv->forced_eos = FALSE;
|
|
|
|
} else if (ret == GST_FLOW_NOT_LINKED || ret <= GST_FLOW_EOS) {
|
|
event = gst_event_new_eos ();
|
|
gst_event_set_seqnum (event, src->priv->seqnum);
|
|
/* for fatal errors we post an error message, post the error
|
|
* first so the app knows about the error first.
|
|
* Also don't do this for FLUSHING because it happens
|
|
* due to flushing and posting an error message because of
|
|
* that is the wrong thing to do, e.g. when we're doing
|
|
* a flushing seek. */
|
|
GST_ELEMENT_FLOW_ERROR (src, ret);
|
|
gst_pad_push_event (pad, event);
|
|
}
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_set_allocation (GstBaseSrc * basesrc, GstBufferPool * pool,
|
|
GstAllocator * allocator, const GstAllocationParams * params)
|
|
{
|
|
GstAllocator *oldalloc;
|
|
GstBufferPool *oldpool;
|
|
GstBaseSrcPrivate *priv = basesrc->priv;
|
|
|
|
if (pool) {
|
|
GST_DEBUG_OBJECT (basesrc, "activate pool");
|
|
if (!gst_buffer_pool_set_active (pool, TRUE))
|
|
goto activate_failed;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
oldpool = priv->pool;
|
|
priv->pool = pool;
|
|
|
|
oldalloc = priv->allocator;
|
|
priv->allocator = allocator;
|
|
|
|
if (priv->pool)
|
|
gst_object_ref (priv->pool);
|
|
if (priv->allocator)
|
|
gst_object_ref (priv->allocator);
|
|
|
|
if (params)
|
|
priv->params = *params;
|
|
else
|
|
gst_allocation_params_init (&priv->params);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
if (oldpool) {
|
|
/* only deactivate if the pool is not the one we're using */
|
|
if (oldpool != pool) {
|
|
GST_DEBUG_OBJECT (basesrc, "deactivate old pool");
|
|
gst_buffer_pool_set_active (oldpool, FALSE);
|
|
}
|
|
gst_object_unref (oldpool);
|
|
}
|
|
if (oldalloc) {
|
|
gst_object_unref (oldalloc);
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
activate_failed:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "failed to activate bufferpool.");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_base_src_set_pool_flushing (GstBaseSrc * basesrc, gboolean flushing)
|
|
{
|
|
GstBaseSrcPrivate *priv = basesrc->priv;
|
|
GstBufferPool *pool;
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
if ((pool = priv->pool))
|
|
pool = gst_object_ref (pool);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
if (pool) {
|
|
gst_buffer_pool_set_flushing (pool, flushing);
|
|
gst_object_unref (pool);
|
|
}
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_base_src_decide_allocation_default (GstBaseSrc * basesrc, GstQuery * query)
|
|
{
|
|
GstCaps *outcaps;
|
|
GstBufferPool *pool;
|
|
guint size, min, max;
|
|
GstAllocator *allocator;
|
|
GstAllocationParams params;
|
|
GstStructure *config;
|
|
gboolean update_allocator;
|
|
|
|
gst_query_parse_allocation (query, &outcaps, NULL);
|
|
|
|
/* we got configuration from our peer or the decide_allocation method,
|
|
* parse them */
|
|
if (gst_query_get_n_allocation_params (query) > 0) {
|
|
/* try the allocator */
|
|
gst_query_parse_nth_allocation_param (query, 0, &allocator, ¶ms);
|
|
update_allocator = TRUE;
|
|
} else {
|
|
allocator = NULL;
|
|
gst_allocation_params_init (¶ms);
|
|
update_allocator = FALSE;
|
|
}
|
|
|
|
if (gst_query_get_n_allocation_pools (query) > 0) {
|
|
gst_query_parse_nth_allocation_pool (query, 0, &pool, &size, &min, &max);
|
|
|
|
if (pool == NULL) {
|
|
/* no pool, we can make our own */
|
|
GST_DEBUG_OBJECT (basesrc, "no pool, making new pool");
|
|
pool = gst_buffer_pool_new ();
|
|
}
|
|
} else {
|
|
pool = NULL;
|
|
size = min = max = 0;
|
|
}
|
|
|
|
/* now configure */
|
|
if (pool) {
|
|
config = gst_buffer_pool_get_config (pool);
|
|
gst_buffer_pool_config_set_params (config, outcaps, size, min, max);
|
|
gst_buffer_pool_config_set_allocator (config, allocator, ¶ms);
|
|
|
|
/* buffer pool may have to do some changes */
|
|
if (!gst_buffer_pool_set_config (pool, config)) {
|
|
config = gst_buffer_pool_get_config (pool);
|
|
|
|
/* If change are not acceptable, fallback to generic pool */
|
|
if (!gst_buffer_pool_config_validate_params (config, outcaps, size, min,
|
|
max)) {
|
|
GST_DEBUG_OBJECT (basesrc, "unsupported pool, making new pool");
|
|
|
|
gst_object_unref (pool);
|
|
pool = gst_buffer_pool_new ();
|
|
gst_buffer_pool_config_set_params (config, outcaps, size, min, max);
|
|
gst_buffer_pool_config_set_allocator (config, allocator, ¶ms);
|
|
}
|
|
|
|
if (!gst_buffer_pool_set_config (pool, config))
|
|
goto config_failed;
|
|
}
|
|
}
|
|
|
|
if (update_allocator)
|
|
gst_query_set_nth_allocation_param (query, 0, allocator, ¶ms);
|
|
else
|
|
gst_query_add_allocation_param (query, allocator, ¶ms);
|
|
if (allocator)
|
|
gst_object_unref (allocator);
|
|
|
|
if (pool) {
|
|
gst_query_set_nth_allocation_pool (query, 0, pool, size, min, max);
|
|
gst_object_unref (pool);
|
|
}
|
|
|
|
return TRUE;
|
|
|
|
config_failed:
|
|
GST_ELEMENT_ERROR (basesrc, RESOURCE, SETTINGS,
|
|
("Failed to configure the buffer pool"),
|
|
("Configuration is most likely invalid, please report this issue."));
|
|
gst_object_unref (pool);
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_prepare_allocation (GstBaseSrc * basesrc, GstCaps * caps)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = TRUE;
|
|
GstQuery *query;
|
|
GstBufferPool *pool = NULL;
|
|
GstAllocator *allocator = NULL;
|
|
GstAllocationParams params;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
/* make query and let peer pad answer, we don't really care if it worked or
|
|
* not, if it failed, the allocation query would contain defaults and the
|
|
* subclass would then set better values if needed */
|
|
query = gst_query_new_allocation (caps, TRUE);
|
|
if (!gst_pad_peer_query (basesrc->srcpad, query)) {
|
|
/* not a problem, just debug a little */
|
|
GST_DEBUG_OBJECT (basesrc, "peer ALLOCATION query failed");
|
|
}
|
|
|
|
g_assert (bclass->decide_allocation != NULL);
|
|
result = bclass->decide_allocation (basesrc, query);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "ALLOCATION (%d) params: %" GST_PTR_FORMAT, result,
|
|
query);
|
|
|
|
if (!result)
|
|
goto no_decide_allocation;
|
|
|
|
/* we got configuration from our peer or the decide_allocation method,
|
|
* parse them */
|
|
if (gst_query_get_n_allocation_params (query) > 0) {
|
|
gst_query_parse_nth_allocation_param (query, 0, &allocator, ¶ms);
|
|
} else {
|
|
allocator = NULL;
|
|
gst_allocation_params_init (¶ms);
|
|
}
|
|
|
|
if (gst_query_get_n_allocation_pools (query) > 0)
|
|
gst_query_parse_nth_allocation_pool (query, 0, &pool, NULL, NULL, NULL);
|
|
|
|
result = gst_base_src_set_allocation (basesrc, pool, allocator, ¶ms);
|
|
|
|
if (allocator)
|
|
gst_object_unref (allocator);
|
|
if (pool)
|
|
gst_object_unref (pool);
|
|
|
|
gst_query_unref (query);
|
|
|
|
return result;
|
|
|
|
/* Errors */
|
|
no_decide_allocation:
|
|
{
|
|
GST_WARNING_OBJECT (basesrc, "Subclass failed to decide allocation");
|
|
gst_query_unref (query);
|
|
|
|
return result;
|
|
}
|
|
}
|
|
|
|
/* default negotiation code.
|
|
*
|
|
* Take intersection between src and sink pads, take first
|
|
* caps and fixate.
|
|
*/
|
|
static gboolean
|
|
gst_base_src_default_negotiate (GstBaseSrc * basesrc)
|
|
{
|
|
GstCaps *thiscaps;
|
|
GstCaps *caps = NULL;
|
|
GstCaps *peercaps = NULL;
|
|
gboolean result = FALSE;
|
|
|
|
/* first see what is possible on our source pad */
|
|
thiscaps = gst_pad_query_caps (GST_BASE_SRC_PAD (basesrc), NULL);
|
|
GST_DEBUG_OBJECT (basesrc, "caps of src: %" GST_PTR_FORMAT, thiscaps);
|
|
/* nothing or anything is allowed, we're done */
|
|
if (thiscaps == NULL || gst_caps_is_any (thiscaps))
|
|
goto no_nego_needed;
|
|
|
|
if (G_UNLIKELY (gst_caps_is_empty (thiscaps)))
|
|
goto no_caps;
|
|
|
|
/* get the peer caps */
|
|
peercaps = gst_pad_peer_query_caps (GST_BASE_SRC_PAD (basesrc), thiscaps);
|
|
GST_DEBUG_OBJECT (basesrc, "caps of peer: %" GST_PTR_FORMAT, peercaps);
|
|
if (peercaps) {
|
|
/* The result is already a subset of our caps */
|
|
caps = peercaps;
|
|
gst_caps_unref (thiscaps);
|
|
} else {
|
|
/* no peer, work with our own caps then */
|
|
caps = thiscaps;
|
|
}
|
|
if (caps && !gst_caps_is_empty (caps)) {
|
|
/* now fixate */
|
|
GST_DEBUG_OBJECT (basesrc, "have caps: %" GST_PTR_FORMAT, caps);
|
|
if (gst_caps_is_any (caps)) {
|
|
GST_DEBUG_OBJECT (basesrc, "any caps, we stop");
|
|
/* hmm, still anything, so element can do anything and
|
|
* nego is not needed */
|
|
result = TRUE;
|
|
} else {
|
|
caps = gst_base_src_fixate (basesrc, caps);
|
|
GST_DEBUG_OBJECT (basesrc, "fixated to: %" GST_PTR_FORMAT, caps);
|
|
if (gst_caps_is_fixed (caps)) {
|
|
/* yay, fixed caps, use those then, it's possible that the subclass does
|
|
* not accept this caps after all and we have to fail. */
|
|
result = gst_base_src_set_caps (basesrc, caps);
|
|
}
|
|
}
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
GST_DEBUG_OBJECT (basesrc, "no common caps");
|
|
}
|
|
return result;
|
|
|
|
no_nego_needed:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "no negotiation needed");
|
|
if (thiscaps)
|
|
gst_caps_unref (thiscaps);
|
|
return TRUE;
|
|
}
|
|
no_caps:
|
|
{
|
|
GST_ELEMENT_ERROR (basesrc, STREAM, FORMAT,
|
|
("No supported formats found"),
|
|
("This element did not produce valid caps"));
|
|
if (thiscaps)
|
|
gst_caps_unref (thiscaps);
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_negotiate_unlocked (GstBaseSrc * basesrc)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "starting negotiation");
|
|
|
|
if (G_LIKELY (bclass->negotiate))
|
|
result = bclass->negotiate (basesrc);
|
|
else
|
|
result = TRUE;
|
|
|
|
if (G_LIKELY (result)) {
|
|
GstCaps *caps;
|
|
|
|
caps = gst_pad_get_current_caps (basesrc->srcpad);
|
|
|
|
result = gst_base_src_prepare_allocation (basesrc, caps);
|
|
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_negotiate:
|
|
* @src: base source instance
|
|
*
|
|
* Negotiates src pad caps with downstream elements.
|
|
* Unmarks GST_PAD_FLAG_NEED_RECONFIGURE in any case. But marks it again
|
|
* if #GstBaseSrcClass::negotiate fails.
|
|
*
|
|
* Do not call this in the #GstBaseSrcClass::fill vmethod. Call this in
|
|
* #GstBaseSrcClass::create or in #GstBaseSrcClass::alloc, _before_ any
|
|
* buffer is allocated.
|
|
*
|
|
* Returns: %TRUE if the negotiation succeeded, else %FALSE.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
gboolean
|
|
gst_base_src_negotiate (GstBaseSrc * src)
|
|
{
|
|
gboolean ret = TRUE;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
|
|
GST_PAD_STREAM_LOCK (src->srcpad);
|
|
gst_pad_check_reconfigure (src->srcpad);
|
|
ret = gst_base_src_negotiate_unlocked (src);
|
|
if (!ret)
|
|
gst_pad_mark_reconfigure (src->srcpad);
|
|
GST_PAD_STREAM_UNLOCK (src->srcpad);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_start (GstBaseSrc * basesrc)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result;
|
|
|
|
GST_LIVE_LOCK (basesrc);
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
if (GST_BASE_SRC_IS_STARTING (basesrc))
|
|
goto was_starting;
|
|
if (GST_BASE_SRC_IS_STARTED (basesrc))
|
|
goto was_started;
|
|
|
|
basesrc->priv->start_result = GST_FLOW_FLUSHING;
|
|
GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
gst_segment_init (&basesrc->segment, basesrc->segment.format);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
basesrc->num_buffers_left = basesrc->num_buffers;
|
|
basesrc->running = FALSE;
|
|
basesrc->priv->segment_pending = FALSE;
|
|
basesrc->priv->segment_seqnum = gst_util_seqnum_next ();
|
|
basesrc->priv->forced_eos = FALSE;
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
if (bclass->start)
|
|
result = bclass->start (basesrc);
|
|
else
|
|
result = TRUE;
|
|
|
|
if (!result)
|
|
goto could_not_start;
|
|
|
|
if (!gst_base_src_is_async (basesrc)) {
|
|
gst_base_src_start_complete (basesrc, GST_FLOW_OK);
|
|
/* not really waiting here, we call this to get the result
|
|
* from the start_complete call */
|
|
result = gst_base_src_start_wait (basesrc) == GST_FLOW_OK;
|
|
}
|
|
|
|
return result;
|
|
|
|
/* ERROR */
|
|
was_starting:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "was starting");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
return TRUE;
|
|
}
|
|
was_started:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "was started");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
return TRUE;
|
|
}
|
|
could_not_start:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "could not start");
|
|
/* subclass is supposed to post a message but we post one as a fallback
|
|
* just in case. We don't have to call _stop. */
|
|
GST_ELEMENT_ERROR (basesrc, CORE, STATE_CHANGE, (NULL),
|
|
("Failed to start"));
|
|
gst_base_src_start_complete (basesrc, GST_FLOW_ERROR);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_start_complete:
|
|
* @basesrc: base source instance
|
|
* @ret: a #GstFlowReturn
|
|
*
|
|
* Complete an asynchronous start operation. When the subclass overrides the
|
|
* start method, it should call gst_base_src_start_complete() when the start
|
|
* operation completes either from the same thread or from an asynchronous
|
|
* helper thread.
|
|
*/
|
|
void
|
|
gst_base_src_start_complete (GstBaseSrc * basesrc, GstFlowReturn ret)
|
|
{
|
|
gboolean have_size;
|
|
guint64 size;
|
|
gboolean seekable;
|
|
GstFormat format;
|
|
GstPadMode mode;
|
|
GstEvent *event;
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
goto error;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "starting source");
|
|
format = basesrc->segment.format;
|
|
|
|
/* figure out the size */
|
|
have_size = FALSE;
|
|
size = -1;
|
|
if (format == GST_FORMAT_BYTES) {
|
|
GstBaseSrcClass *bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
if (bclass->get_size) {
|
|
if (!(have_size = bclass->get_size (basesrc, &size)))
|
|
size = -1;
|
|
}
|
|
GST_DEBUG_OBJECT (basesrc, "setting size %" G_GUINT64_FORMAT, size);
|
|
/* only update the size when operating in bytes, subclass is supposed
|
|
* to set duration in the start method for other formats */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
basesrc->segment.duration = size;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (basesrc,
|
|
"format: %s, have size: %d, size: %" G_GUINT64_FORMAT ", duration: %"
|
|
G_GINT64_FORMAT, gst_format_get_name (format), have_size, size,
|
|
basesrc->segment.duration);
|
|
|
|
seekable = gst_base_src_seekable (basesrc);
|
|
GST_DEBUG_OBJECT (basesrc, "is seekable: %d", seekable);
|
|
|
|
/* update for random access flag */
|
|
basesrc->random_access = seekable && format == GST_FORMAT_BYTES;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "is random_access: %d", basesrc->random_access);
|
|
|
|
gst_pad_mark_reconfigure (GST_BASE_SRC_PAD (basesrc));
|
|
|
|
GST_OBJECT_LOCK (basesrc->srcpad);
|
|
mode = GST_PAD_MODE (basesrc->srcpad);
|
|
GST_OBJECT_UNLOCK (basesrc->srcpad);
|
|
|
|
/* take the stream lock here, we only want to let the task run when we have
|
|
* set the STARTED flag */
|
|
GST_PAD_STREAM_LOCK (basesrc->srcpad);
|
|
switch (mode) {
|
|
case GST_PAD_MODE_PUSH:
|
|
/* do initial seek, which will start the task */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
event = basesrc->pending_seek;
|
|
basesrc->pending_seek = NULL;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
/* The perform seek code will start the task when finished. We don't have to
|
|
* unlock the streaming thread because it is not running yet */
|
|
if (G_UNLIKELY (!gst_base_src_perform_seek (basesrc, event, FALSE)))
|
|
goto seek_failed;
|
|
|
|
if (event)
|
|
gst_event_unref (event);
|
|
break;
|
|
case GST_PAD_MODE_PULL:
|
|
/* if not random_access, we cannot operate in pull mode for now */
|
|
if (G_UNLIKELY (!basesrc->random_access))
|
|
goto no_get_range;
|
|
break;
|
|
default:
|
|
goto not_activated_yet;
|
|
break;
|
|
}
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
GST_OBJECT_FLAG_SET (basesrc, GST_BASE_SRC_FLAG_STARTED);
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
basesrc->priv->start_result = ret;
|
|
GST_ASYNC_SIGNAL (basesrc);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
GST_PAD_STREAM_UNLOCK (basesrc->srcpad);
|
|
|
|
return;
|
|
|
|
seek_failed:
|
|
{
|
|
GST_PAD_STREAM_UNLOCK (basesrc->srcpad);
|
|
GST_ERROR_OBJECT (basesrc, "Failed to perform initial seek");
|
|
gst_base_src_stop (basesrc);
|
|
if (event)
|
|
gst_event_unref (event);
|
|
ret = GST_FLOW_ERROR;
|
|
goto error;
|
|
}
|
|
no_get_range:
|
|
{
|
|
GST_PAD_STREAM_UNLOCK (basesrc->srcpad);
|
|
gst_base_src_stop (basesrc);
|
|
GST_ERROR_OBJECT (basesrc, "Cannot operate in pull mode, stopping");
|
|
ret = GST_FLOW_ERROR;
|
|
goto error;
|
|
}
|
|
not_activated_yet:
|
|
{
|
|
GST_PAD_STREAM_UNLOCK (basesrc->srcpad);
|
|
gst_base_src_stop (basesrc);
|
|
GST_INFO_OBJECT (basesrc, "pad not activated yet");
|
|
ret = GST_FLOW_ERROR;
|
|
goto error;
|
|
}
|
|
error:
|
|
{
|
|
GST_OBJECT_LOCK (basesrc);
|
|
basesrc->priv->start_result = ret;
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
GST_ASYNC_SIGNAL (basesrc);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
return;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_start_wait:
|
|
* @basesrc: base source instance
|
|
*
|
|
* Wait until the start operation completes.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
gst_base_src_start_wait (GstBaseSrc * basesrc)
|
|
{
|
|
GstFlowReturn result;
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
while (GST_BASE_SRC_IS_STARTING (basesrc)) {
|
|
GST_ASYNC_WAIT (basesrc);
|
|
}
|
|
result = basesrc->priv->start_result;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "got %s", gst_flow_get_name (result));
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_stop (GstBaseSrc * basesrc)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
gboolean result = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "stopping source");
|
|
|
|
/* flush all */
|
|
gst_base_src_set_flushing (basesrc, TRUE);
|
|
|
|
/* stop the task */
|
|
gst_pad_stop_task (basesrc->srcpad);
|
|
/* stop flushing, this will balance unlock/unlock_stop calls */
|
|
gst_base_src_set_flushing (basesrc, FALSE);
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
if (!GST_BASE_SRC_IS_STARTED (basesrc) && !GST_BASE_SRC_IS_STARTING (basesrc))
|
|
goto was_stopped;
|
|
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTING);
|
|
GST_OBJECT_FLAG_UNSET (basesrc, GST_BASE_SRC_FLAG_STARTED);
|
|
basesrc->priv->start_result = GST_FLOW_FLUSHING;
|
|
GST_ASYNC_SIGNAL (basesrc);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
if (bclass->stop)
|
|
result = bclass->stop (basesrc);
|
|
|
|
if (basesrc->priv->pending_bufferlist != NULL) {
|
|
gst_buffer_list_unref (basesrc->priv->pending_bufferlist);
|
|
basesrc->priv->pending_bufferlist = NULL;
|
|
}
|
|
|
|
gst_base_src_set_allocation (basesrc, NULL, NULL, NULL);
|
|
|
|
return result;
|
|
|
|
was_stopped:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "was stopped");
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
/* start or stop flushing dataprocessing
|
|
*/
|
|
static gboolean
|
|
gst_base_src_set_flushing (GstBaseSrc * basesrc, gboolean flushing)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
GST_DEBUG_OBJECT (basesrc, "flushing %d", flushing);
|
|
|
|
if (flushing) {
|
|
gst_base_src_set_pool_flushing (basesrc, TRUE);
|
|
/* unlock any subclasses to allow turning off the streaming thread */
|
|
if (bclass->unlock)
|
|
bclass->unlock (basesrc);
|
|
}
|
|
|
|
/* the live lock is released when we are blocked, waiting for playing,
|
|
* when we sync to the clock or creating a buffer */
|
|
GST_LIVE_LOCK (basesrc);
|
|
basesrc->priv->flushing = flushing;
|
|
if (flushing) {
|
|
/* clear pending EOS if any */
|
|
if (g_atomic_int_get (&basesrc->priv->has_pending_eos)) {
|
|
GST_OBJECT_LOCK (basesrc);
|
|
CLEAR_PENDING_EOS (basesrc);
|
|
basesrc->priv->forced_eos = FALSE;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
}
|
|
|
|
/* unblock clock sync (if any) or any other blocking thing */
|
|
if (basesrc->clock_id)
|
|
gst_clock_id_unschedule (basesrc->clock_id);
|
|
} else {
|
|
gst_base_src_set_pool_flushing (basesrc, FALSE);
|
|
|
|
/* Drop all delayed events */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
if (basesrc->priv->pending_events) {
|
|
g_list_foreach (basesrc->priv->pending_events, (GFunc) gst_event_unref,
|
|
NULL);
|
|
g_list_free (basesrc->priv->pending_events);
|
|
basesrc->priv->pending_events = NULL;
|
|
g_atomic_int_set (&basesrc->priv->have_events, FALSE);
|
|
}
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
}
|
|
|
|
GST_LIVE_SIGNAL (basesrc);
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
if (!flushing) {
|
|
/* Now wait for the stream lock to be released and clear our unlock request */
|
|
GST_PAD_STREAM_LOCK (basesrc->srcpad);
|
|
if (bclass->unlock_stop)
|
|
bclass->unlock_stop (basesrc);
|
|
GST_PAD_STREAM_UNLOCK (basesrc->srcpad);
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* the purpose of this function is to make sure that a live source blocks in the
|
|
* LIVE lock or leaves the LIVE lock and continues playing. */
|
|
static gboolean
|
|
gst_base_src_set_playing (GstBaseSrc * basesrc, gboolean live_play)
|
|
{
|
|
GstBaseSrcClass *bclass;
|
|
|
|
bclass = GST_BASE_SRC_GET_CLASS (basesrc);
|
|
|
|
/* we are now able to grab the LIVE lock, when we get it, we can be
|
|
* waiting for PLAYING while blocked in the LIVE cond or we can be waiting
|
|
* for the clock. */
|
|
GST_LIVE_LOCK (basesrc);
|
|
GST_DEBUG_OBJECT (basesrc, "unschedule clock");
|
|
|
|
/* unlock subclasses locked in ::create, we only do this when we stop playing. */
|
|
if (!live_play) {
|
|
GST_DEBUG_OBJECT (basesrc, "unlock");
|
|
if (bclass->unlock)
|
|
bclass->unlock (basesrc);
|
|
}
|
|
|
|
/* unblock clock sync (if any) */
|
|
if (basesrc->clock_id)
|
|
gst_clock_id_unschedule (basesrc->clock_id);
|
|
|
|
/* configure what to do when we get to the LIVE lock. */
|
|
GST_DEBUG_OBJECT (basesrc, "live running %d", live_play);
|
|
basesrc->live_running = live_play;
|
|
|
|
if (live_play) {
|
|
gboolean start;
|
|
|
|
/* for live sources we restart the timestamp correction */
|
|
GST_OBJECT_LOCK (basesrc);
|
|
basesrc->priv->latency = -1;
|
|
basesrc->priv->pending_unlock_stop = TRUE;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
/* have to restart the task in case it stopped because of the unlock when
|
|
* we went to PAUSED. Only do this if we operating in push mode. */
|
|
GST_OBJECT_LOCK (basesrc->srcpad);
|
|
start = (GST_PAD_MODE (basesrc->srcpad) == GST_PAD_MODE_PUSH);
|
|
GST_OBJECT_UNLOCK (basesrc->srcpad);
|
|
if (start)
|
|
gst_pad_start_task (basesrc->srcpad, (GstTaskFunction) gst_base_src_loop,
|
|
basesrc->srcpad, NULL);
|
|
GST_DEBUG_OBJECT (basesrc, "signal");
|
|
GST_LIVE_SIGNAL (basesrc);
|
|
}
|
|
GST_LIVE_UNLOCK (basesrc);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_push (GstPad * pad, GstObject * parent, gboolean active)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
|
|
basesrc = GST_BASE_SRC (parent);
|
|
|
|
/* prepare subclass first */
|
|
if (active) {
|
|
GST_DEBUG_OBJECT (basesrc, "Activating in push mode");
|
|
|
|
if (G_UNLIKELY (!basesrc->can_activate_push))
|
|
goto no_push_activation;
|
|
|
|
if (G_UNLIKELY (!gst_base_src_start (basesrc)))
|
|
goto error_start;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesrc, "Deactivating in push mode");
|
|
/* now we can stop the source */
|
|
if (G_UNLIKELY (!gst_base_src_stop (basesrc)))
|
|
goto error_stop;
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_push_activation:
|
|
{
|
|
GST_WARNING_OBJECT (basesrc, "Subclass disabled push-mode activation");
|
|
return FALSE;
|
|
}
|
|
error_start:
|
|
{
|
|
GST_WARNING_OBJECT (basesrc, "Failed to start in push mode");
|
|
return FALSE;
|
|
}
|
|
error_stop:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "Failed to stop in push mode");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_pull (GstPad * pad, GstObject * parent, gboolean active)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
|
|
basesrc = GST_BASE_SRC (parent);
|
|
|
|
/* prepare subclass first */
|
|
if (active) {
|
|
GST_DEBUG_OBJECT (basesrc, "Activating in pull mode");
|
|
if (G_UNLIKELY (!gst_base_src_start (basesrc)))
|
|
goto error_start;
|
|
} else {
|
|
GST_DEBUG_OBJECT (basesrc, "Deactivating in pull mode");
|
|
if (G_UNLIKELY (!gst_base_src_stop (basesrc)))
|
|
goto error_stop;
|
|
}
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
error_start:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "Failed to start in pull mode");
|
|
return FALSE;
|
|
}
|
|
error_stop:
|
|
{
|
|
GST_ERROR_OBJECT (basesrc, "Failed to stop in pull mode");
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_base_src_activate_mode (GstPad * pad, GstObject * parent,
|
|
GstPadMode mode, gboolean active)
|
|
{
|
|
gboolean res;
|
|
GstBaseSrc *src = GST_BASE_SRC (parent);
|
|
|
|
src->priv->stream_start_pending = FALSE;
|
|
|
|
GST_DEBUG_OBJECT (pad, "activating in mode %d", mode);
|
|
|
|
switch (mode) {
|
|
case GST_PAD_MODE_PULL:
|
|
res = gst_base_src_activate_pull (pad, parent, active);
|
|
break;
|
|
case GST_PAD_MODE_PUSH:
|
|
src->priv->stream_start_pending = active;
|
|
res = gst_base_src_activate_push (pad, parent, active);
|
|
break;
|
|
default:
|
|
GST_LOG_OBJECT (pad, "unknown activation mode %d", mode);
|
|
res = FALSE;
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
|
|
static GstStateChangeReturn
|
|
gst_base_src_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstBaseSrc *basesrc;
|
|
GstStateChangeReturn result;
|
|
gboolean no_preroll = FALSE;
|
|
|
|
basesrc = GST_BASE_SRC (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
no_preroll = gst_base_src_is_live (basesrc);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
GST_DEBUG_OBJECT (basesrc, "PAUSED->PLAYING");
|
|
if (gst_base_src_is_live (basesrc)) {
|
|
/* now we can start playback */
|
|
gst_base_src_set_playing (basesrc, TRUE);
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if ((result =
|
|
GST_ELEMENT_CLASS (parent_class)->change_state (element,
|
|
transition)) == GST_STATE_CHANGE_FAILURE)
|
|
goto failure;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
GST_DEBUG_OBJECT (basesrc, "PLAYING->PAUSED");
|
|
if (gst_base_src_is_live (basesrc)) {
|
|
/* make sure we block in the live cond in PAUSED */
|
|
gst_base_src_set_playing (basesrc, FALSE);
|
|
no_preroll = TRUE;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
{
|
|
/* we don't need to unblock anything here, the pad deactivation code
|
|
* already did this */
|
|
if (g_atomic_int_get (&basesrc->priv->has_pending_eos)) {
|
|
GST_OBJECT_LOCK (basesrc);
|
|
CLEAR_PENDING_EOS (basesrc);
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
}
|
|
gst_event_replace (&basesrc->pending_seek, NULL);
|
|
break;
|
|
}
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (no_preroll && result == GST_STATE_CHANGE_SUCCESS)
|
|
result = GST_STATE_CHANGE_NO_PREROLL;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
failure:
|
|
{
|
|
GST_DEBUG_OBJECT (basesrc, "parent failed state change");
|
|
return result;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_get_buffer_pool:
|
|
* @src: a #GstBaseSrc
|
|
*
|
|
* Returns: (nullable) (transfer full): the instance of the #GstBufferPool used
|
|
* by the src; unref it after usage.
|
|
*/
|
|
GstBufferPool *
|
|
gst_base_src_get_buffer_pool (GstBaseSrc * src)
|
|
{
|
|
GstBufferPool *ret = NULL;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), NULL);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
if (src->priv->pool)
|
|
ret = gst_object_ref (src->priv->pool);
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
return ret;
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_get_allocator:
|
|
* @src: a #GstBaseSrc
|
|
* @allocator: (out) (optional) (nullable) (transfer full): the #GstAllocator
|
|
* used
|
|
* @params: (out caller-allocates) (optional): the #GstAllocationParams of @allocator
|
|
*
|
|
* Lets #GstBaseSrc sub-classes to know the memory @allocator
|
|
* used by the base class and its @params.
|
|
*
|
|
* Unref the @allocator after usage.
|
|
*/
|
|
void
|
|
gst_base_src_get_allocator (GstBaseSrc * src,
|
|
GstAllocator ** allocator, GstAllocationParams * params)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
if (allocator)
|
|
*allocator = src->priv->allocator ?
|
|
gst_object_ref (src->priv->allocator) : NULL;
|
|
|
|
if (params)
|
|
*params = src->priv->params;
|
|
GST_OBJECT_UNLOCK (src);
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_submit_buffer_list:
|
|
* @src: a #GstBaseSrc
|
|
* @buffer_list: (transfer full): a #GstBufferList
|
|
*
|
|
* Subclasses can call this from their create virtual method implementation
|
|
* to submit a buffer list to be pushed out later. This is useful in
|
|
* cases where the create function wants to produce multiple buffers to be
|
|
* pushed out in one go in form of a #GstBufferList, which can reduce overhead
|
|
* drastically, especially for packetised inputs (for data streams where
|
|
* the packetisation/chunking is not important it is usually more efficient
|
|
* to return larger buffers instead).
|
|
*
|
|
* Subclasses that use this function from their create function must return
|
|
* %GST_FLOW_OK and no buffer from their create virtual method implementation.
|
|
* If a buffer is returned after a buffer list has also been submitted via this
|
|
* function the behaviour is undefined.
|
|
*
|
|
* Subclasses must only call this function once per create function call and
|
|
* subclasses must only call this function when the source operates in push
|
|
* mode.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
void
|
|
gst_base_src_submit_buffer_list (GstBaseSrc * src, GstBufferList * buffer_list)
|
|
{
|
|
g_return_if_fail (GST_IS_BASE_SRC (src));
|
|
g_return_if_fail (GST_IS_BUFFER_LIST (buffer_list));
|
|
g_return_if_fail (BASE_SRC_HAS_PENDING_BUFFER_LIST (src) == FALSE);
|
|
|
|
/* we need it to be writable later in get_range() where we use get_writable */
|
|
src->priv->pending_bufferlist = gst_buffer_list_make_writable (buffer_list);
|
|
|
|
GST_LOG_OBJECT (src, "%u buffers submitted in buffer list",
|
|
gst_buffer_list_length (buffer_list));
|
|
}
|
|
|
|
/**
|
|
* gst_base_src_push_segment:
|
|
* @src: a #GstBaseSrc
|
|
* @segment: a pointer to a #GstSegment
|
|
*
|
|
* Send a new segment downstream. This function must
|
|
* only be called by derived sub-classes, and only from the #GstBaseSrcClass::create function,
|
|
* as the stream-lock needs to be held.
|
|
* This method also requires that an out caps has been configured, so
|
|
* gst_base_src_set_caps() needs to have been called before.
|
|
*
|
|
* The format for the @segment must be identical with the current format
|
|
* of the source, as configured with gst_base_src_set_format().
|
|
*
|
|
* The format of @src must not be %GST_FORMAT_UNDEFINED and the format
|
|
* should be configured via gst_base_src_set_format() before calling this method.
|
|
*
|
|
* This is a variant of gst_base_src_new_segment() sending the segment right away,
|
|
* which can be useful to ensure events ordering.
|
|
*
|
|
* Returns: %TRUE if sending of new segment succeeded.
|
|
*
|
|
* Since: 1.24
|
|
*/
|
|
gboolean
|
|
gst_base_src_push_segment (GstBaseSrc * src, const GstSegment * segment)
|
|
{
|
|
GstEvent *seg_event;
|
|
|
|
g_return_val_if_fail (GST_IS_BASE_SRC (src), FALSE);
|
|
g_return_val_if_fail (segment != NULL, FALSE);
|
|
|
|
GST_OBJECT_LOCK (src);
|
|
|
|
if (src->segment.format == GST_FORMAT_UNDEFINED) {
|
|
/* subclass must set valid format before calling this method */
|
|
GST_WARNING_OBJECT (src, "segment format is not configured yet, ignore");
|
|
GST_OBJECT_UNLOCK (src);
|
|
return FALSE;
|
|
}
|
|
|
|
if (src->segment.format != segment->format) {
|
|
GST_WARNING_OBJECT (src, "segment format mismatched, ignore");
|
|
GST_OBJECT_UNLOCK (src);
|
|
return FALSE;
|
|
}
|
|
|
|
gst_segment_copy_into (segment, &src->segment);
|
|
seg_event = gst_event_new_segment (&src->segment);
|
|
src->priv->segment_pending = FALSE;
|
|
src->priv->segment_seqnum = gst_util_seqnum_next ();
|
|
gst_event_set_seqnum (seg_event, src->priv->segment_seqnum);
|
|
|
|
GST_OBJECT_UNLOCK (src);
|
|
|
|
GST_DEBUG_OBJECT (src, "Sending new segment %" GST_SEGMENT_FORMAT, segment);
|
|
gst_pad_push_event (src->srcpad, seg_event);
|
|
|
|
src->running = TRUE;
|
|
|
|
return TRUE;
|
|
}
|