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d2a0b38717
This allows the new caps to be sent downstream as soon as possible instead of having to wait for the next buffer/buffer list. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/798>
2332 lines
68 KiB
C
2332 lines
68 KiB
C
/* GStreamer
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* Copyright (C) 2007 David Schleef <ds@schleef.org>
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* (C) 2008 Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:gstappsrc
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* @title: GstAppSrc
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* @short_description: Easy way for applications to inject buffers into a
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* pipeline
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* @see_also: #GstBaseSrc, appsink
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*
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* The appsrc element can be used by applications to insert data into a
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* GStreamer pipeline. Unlike most GStreamer elements, appsrc provides
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* external API functions.
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*
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* appsrc can be used by linking with the libgstapp library to access the
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* methods directly or by using the appsrc action signals.
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*
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* Before operating appsrc, the caps property must be set to fixed caps
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* describing the format of the data that will be pushed with appsrc. An
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* exception to this is when pushing buffers with unknown caps, in which case no
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* caps should be set. This is typically true of file-like sources that push raw
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* byte buffers. If you don't want to explicitly set the caps, you can use
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* gst_app_src_push_sample. This method gets the caps associated with the
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* sample and sets them on the appsrc replacing any previously set caps (if
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* different from sample's caps).
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*
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* The main way of handing data to the appsrc element is by calling the
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* gst_app_src_push_buffer() method or by emitting the push-buffer action signal.
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* This will put the buffer onto a queue from which appsrc will read from in its
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* streaming thread. It is important to note that data transport will not happen
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* from the thread that performed the push-buffer call.
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*
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* The "max-bytes" property controls how much data can be queued in appsrc
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* before appsrc considers the queue full. A filled internal queue will always
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* signal the "enough-data" signal, which signals the application that it should
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* stop pushing data into appsrc. The "block" property will cause appsrc to
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* block the push-buffer method until free data becomes available again.
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*
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* When the internal queue is running out of data, the "need-data" signal is
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* emitted, which signals the application that it should start pushing more data
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* into appsrc.
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*
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* In addition to the "need-data" and "enough-data" signals, appsrc can emit the
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* "seek-data" signal when the "stream-mode" property is set to "seekable" or
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* "random-access". The signal argument will contain the new desired position in
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* the stream expressed in the unit set with the "format" property. After
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* receiving the seek-data signal, the application should push-buffers from the
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* new position.
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*
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* These signals allow the application to operate the appsrc in two different
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* ways:
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*
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* The push mode, in which the application repeatedly calls the push-buffer/push-sample
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* method with a new buffer/sample. Optionally, the queue size in the appsrc
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* can be controlled with the enough-data and need-data signals by respectively
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* stopping/starting the push-buffer/push-sample calls. This is a typical
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* mode of operation for the stream-type "stream" and "seekable". Use this
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* mode when implementing various network protocols or hardware devices.
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*
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* The pull mode, in which the need-data signal triggers the next push-buffer call.
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* This mode is typically used in the "random-access" stream-type. Use this
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* mode for file access or other randomly accessible sources. In this mode, a
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* buffer of exactly the amount of bytes given by the need-data signal should be
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* pushed into appsrc.
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*
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* In all modes, the size property on appsrc should contain the total stream
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* size in bytes. Setting this property is mandatory in the random-access mode.
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* For the stream and seekable modes, setting this property is optional but
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* recommended.
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*
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* When the application has finished pushing data into appsrc, it should call
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* gst_app_src_end_of_stream() or emit the end-of-stream action signal. After
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* this call, no more buffers can be pushed into appsrc until a flushing seek
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* occurs or the state of the appsrc has gone through READY.
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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 <gst/gst.h>
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#include <gst/base/base.h>
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#include <string.h>
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#include "gstappsrc.h"
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typedef enum
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{
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NOONE_WAITING = 0,
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STREAM_WAITING = 1 << 0, /* streaming thread is waiting for application thread */
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APP_WAITING = 1 << 1, /* application thread is waiting for streaming thread */
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} GstAppSrcWaitStatus;
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typedef struct
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{
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GstAppSrcCallbacks callbacks;
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gpointer user_data;
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GDestroyNotify destroy_notify;
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gint ref_count;
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} Callbacks;
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static Callbacks *
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callbacks_ref (Callbacks * callbacks)
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{
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g_atomic_int_inc (&callbacks->ref_count);
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return callbacks;
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}
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static void
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callbacks_unref (Callbacks * callbacks)
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{
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if (!g_atomic_int_dec_and_test (&callbacks->ref_count))
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return;
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if (callbacks->destroy_notify)
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callbacks->destroy_notify (callbacks->user_data);
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g_free (callbacks);
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}
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struct _GstAppSrcPrivate
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{
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GCond cond;
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GMutex mutex;
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GstQueueArray *queue;
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GstAppSrcWaitStatus wait_status;
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GstCaps *last_caps;
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GstCaps *current_caps;
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GstSegment last_segment;
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GstSegment current_segment;
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gboolean pending_custom_segment;
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gint64 size;
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GstClockTime duration;
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GstAppStreamType stream_type;
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guint64 max_bytes;
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GstFormat format;
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gboolean block;
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gchar *uri;
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gboolean flushing;
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gboolean started;
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gboolean is_eos;
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guint64 queued_bytes;
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guint64 offset;
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GstAppStreamType current_type;
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guint64 min_latency;
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guint64 max_latency;
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gboolean emit_signals;
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guint min_percent;
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gboolean handle_segment_change;
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Callbacks *callbacks;
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};
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GST_DEBUG_CATEGORY_STATIC (app_src_debug);
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#define GST_CAT_DEFAULT app_src_debug
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enum
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{
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/* signals */
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SIGNAL_NEED_DATA,
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SIGNAL_ENOUGH_DATA,
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SIGNAL_SEEK_DATA,
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/* actions */
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SIGNAL_PUSH_BUFFER,
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SIGNAL_END_OF_STREAM,
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SIGNAL_PUSH_SAMPLE,
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SIGNAL_PUSH_BUFFER_LIST,
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LAST_SIGNAL
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};
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#define DEFAULT_PROP_SIZE -1
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#define DEFAULT_PROP_STREAM_TYPE GST_APP_STREAM_TYPE_STREAM
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#define DEFAULT_PROP_MAX_BYTES 200000
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#define DEFAULT_PROP_FORMAT GST_FORMAT_BYTES
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#define DEFAULT_PROP_BLOCK FALSE
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#define DEFAULT_PROP_IS_LIVE FALSE
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#define DEFAULT_PROP_MIN_LATENCY -1
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#define DEFAULT_PROP_MAX_LATENCY -1
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#define DEFAULT_PROP_EMIT_SIGNALS TRUE
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#define DEFAULT_PROP_MIN_PERCENT 0
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#define DEFAULT_PROP_CURRENT_LEVEL_BYTES 0
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#define DEFAULT_PROP_DURATION GST_CLOCK_TIME_NONE
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#define DEFAULT_PROP_HANDLE_SEGMENT_CHANGE FALSE
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enum
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{
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PROP_0,
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PROP_CAPS,
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PROP_SIZE,
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PROP_STREAM_TYPE,
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PROP_MAX_BYTES,
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PROP_FORMAT,
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PROP_BLOCK,
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PROP_IS_LIVE,
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PROP_MIN_LATENCY,
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PROP_MAX_LATENCY,
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PROP_EMIT_SIGNALS,
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PROP_MIN_PERCENT,
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PROP_CURRENT_LEVEL_BYTES,
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PROP_DURATION,
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PROP_HANDLE_SEGMENT_CHANGE,
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PROP_LAST
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};
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static GstStaticPadTemplate gst_app_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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static void gst_app_src_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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static void gst_app_src_dispose (GObject * object);
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static void gst_app_src_finalize (GObject * object);
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static void gst_app_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_app_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_app_src_send_event (GstElement * element, GstEvent * event);
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static void gst_app_src_set_latencies (GstAppSrc * appsrc,
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gboolean do_min, guint64 min, gboolean do_max, guint64 max);
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static gboolean gst_app_src_negotiate (GstBaseSrc * basesrc);
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static GstCaps *gst_app_src_internal_get_caps (GstBaseSrc * bsrc,
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GstCaps * filter);
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static GstFlowReturn gst_app_src_create (GstBaseSrc * bsrc, guint64 offset,
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guint size, GstBuffer ** buf);
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static gboolean gst_app_src_start (GstBaseSrc * bsrc);
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static gboolean gst_app_src_stop (GstBaseSrc * bsrc);
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static gboolean gst_app_src_unlock (GstBaseSrc * bsrc);
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static gboolean gst_app_src_unlock_stop (GstBaseSrc * bsrc);
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static gboolean gst_app_src_do_seek (GstBaseSrc * src, GstSegment * segment);
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static gboolean gst_app_src_is_seekable (GstBaseSrc * src);
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static gboolean gst_app_src_do_get_size (GstBaseSrc * src, guint64 * size);
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static gboolean gst_app_src_query (GstBaseSrc * src, GstQuery * query);
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static gboolean gst_app_src_event (GstBaseSrc * src, GstEvent * event);
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static GstFlowReturn gst_app_src_push_buffer_action (GstAppSrc * appsrc,
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GstBuffer * buffer);
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static GstFlowReturn gst_app_src_push_buffer_list_action (GstAppSrc * appsrc,
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GstBufferList * buffer_list);
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static GstFlowReturn gst_app_src_push_sample_action (GstAppSrc * appsrc,
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GstSample * sample);
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static guint gst_app_src_signals[LAST_SIGNAL] = { 0 };
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#define gst_app_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstAppSrc, gst_app_src, GST_TYPE_BASE_SRC,
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G_ADD_PRIVATE (GstAppSrc)
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_app_src_uri_handler_init));
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static void
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gst_app_src_class_init (GstAppSrcClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) klass;
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GstBaseSrcClass *basesrc_class = (GstBaseSrcClass *) klass;
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GST_DEBUG_CATEGORY_INIT (app_src_debug, "appsrc", 0, "appsrc element");
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gobject_class->dispose = gst_app_src_dispose;
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gobject_class->finalize = gst_app_src_finalize;
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gobject_class->set_property = gst_app_src_set_property;
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gobject_class->get_property = gst_app_src_get_property;
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/**
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* GstAppSrc:caps:
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*
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* The GstCaps that will negotiated downstream and will be put
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* on outgoing buffers.
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*/
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g_object_class_install_property (gobject_class, PROP_CAPS,
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g_param_spec_boxed ("caps", "Caps",
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"The allowed caps for the src pad", GST_TYPE_CAPS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:format:
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*
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* The format to use for segment events. When the source is producing
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* timestamped buffers this property should be set to GST_FORMAT_TIME.
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*/
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g_object_class_install_property (gobject_class, PROP_FORMAT,
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g_param_spec_enum ("format", "Format",
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"The format of the segment events and seek", GST_TYPE_FORMAT,
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DEFAULT_PROP_FORMAT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:size:
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*
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* The total size in bytes of the data stream. If the total size is known, it
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* is recommended to configure it with this property.
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*/
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g_object_class_install_property (gobject_class, PROP_SIZE,
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g_param_spec_int64 ("size", "Size",
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"The size of the data stream in bytes (-1 if unknown)",
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-1, G_MAXINT64, DEFAULT_PROP_SIZE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:stream-type:
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*
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* The type of stream that this source is producing. For seekable streams the
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* application should connect to the seek-data signal.
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*/
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g_object_class_install_property (gobject_class, PROP_STREAM_TYPE,
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g_param_spec_enum ("stream-type", "Stream Type",
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"the type of the stream", GST_TYPE_APP_STREAM_TYPE,
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DEFAULT_PROP_STREAM_TYPE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:max-bytes:
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*
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* The maximum amount of bytes that can be queued internally.
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* After the maximum amount of bytes are queued, appsrc will emit the
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* "enough-data" signal.
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*/
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g_object_class_install_property (gobject_class, PROP_MAX_BYTES,
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g_param_spec_uint64 ("max-bytes", "Max bytes",
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"The maximum number of bytes to queue internally (0 = unlimited)",
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0, G_MAXUINT64, DEFAULT_PROP_MAX_BYTES,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:block:
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*
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* When max-bytes are queued and after the enough-data signal has been emitted,
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* block any further push-buffer calls until the amount of queued bytes drops
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* below the max-bytes limit.
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*/
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g_object_class_install_property (gobject_class, PROP_BLOCK,
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g_param_spec_boolean ("block", "Block",
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"Block push-buffer when max-bytes are queued",
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DEFAULT_PROP_BLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:is-live:
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*
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* Instruct the source to behave like a live source. This includes that it
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* will only push out buffers in the PLAYING state.
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*/
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g_object_class_install_property (gobject_class, PROP_IS_LIVE,
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g_param_spec_boolean ("is-live", "Is Live",
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"Whether to act as a live source",
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DEFAULT_PROP_IS_LIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:min-latency:
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*
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* The minimum latency of the source. A value of -1 will use the default
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* latency calculations of #GstBaseSrc.
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*/
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g_object_class_install_property (gobject_class, PROP_MIN_LATENCY,
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g_param_spec_int64 ("min-latency", "Min Latency",
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"The minimum latency (-1 = default)",
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-1, G_MAXINT64, DEFAULT_PROP_MIN_LATENCY,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc::max-latency:
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*
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* The maximum latency of the source. A value of -1 means an unlimited amount
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* of latency.
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*/
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g_object_class_install_property (gobject_class, PROP_MAX_LATENCY,
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g_param_spec_int64 ("max-latency", "Max Latency",
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"The maximum latency (-1 = unlimited)",
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-1, G_MAXINT64, DEFAULT_PROP_MAX_LATENCY,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:emit-signals:
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*
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* Make appsrc emit the "need-data", "enough-data" and "seek-data" signals.
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* This option is by default enabled for backwards compatibility reasons but
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* can disabled when needed because signal emission is expensive.
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*/
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g_object_class_install_property (gobject_class, PROP_EMIT_SIGNALS,
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g_param_spec_boolean ("emit-signals", "Emit signals",
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"Emit need-data, enough-data and seek-data signals",
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DEFAULT_PROP_EMIT_SIGNALS,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:min-percent:
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*
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* Make appsrc emit the "need-data" signal when the amount of bytes in the
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* queue drops below this percentage of max-bytes.
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*/
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g_object_class_install_property (gobject_class, PROP_MIN_PERCENT,
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g_param_spec_uint ("min-percent", "Min Percent",
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"Emit need-data when queued bytes drops below this percent of max-bytes",
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0, 100, DEFAULT_PROP_MIN_PERCENT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:current-level-bytes:
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*
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* The number of currently queued bytes inside appsrc.
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*
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* Since: 1.2
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*/
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g_object_class_install_property (gobject_class, PROP_CURRENT_LEVEL_BYTES,
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g_param_spec_uint64 ("current-level-bytes", "Current Level Bytes",
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"The number of currently queued bytes",
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0, G_MAXUINT64, DEFAULT_PROP_CURRENT_LEVEL_BYTES,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:duration:
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*
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* The total duration in nanoseconds of the data stream. If the total duration is known, it
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* is recommended to configure it with this property.
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*
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* Since: 1.10
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*/
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g_object_class_install_property (gobject_class, PROP_DURATION,
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g_param_spec_uint64 ("duration", "Duration",
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"The duration of the data stream in nanoseconds (GST_CLOCK_TIME_NONE if unknown)",
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0, G_MAXUINT64, DEFAULT_PROP_DURATION,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstAppSrc:handle-segment-change:
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*
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* When enabled, appsrc will check GstSegment in GstSample which was
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* pushed via gst_app_src_push_sample() or "push-sample" signal action.
|
|
* If a GstSegment is changed, corresponding segment event will be followed
|
|
* by next data flow.
|
|
*
|
|
* FIXME: currently only GST_FORMAT_TIME format is supported and therefore
|
|
* GstAppSrc::format should be time. However, possibly #GstAppSrc can support
|
|
* other formats.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_HANDLE_SEGMENT_CHANGE,
|
|
g_param_spec_boolean ("handle-segment-change", "Handle Segment Change",
|
|
"Whether to detect and handle changed time format GstSegment in "
|
|
"GstSample. User should set valid GstSegment in GstSample. "
|
|
"Must set format property as \"time\" to enable this property",
|
|
DEFAULT_PROP_HANDLE_SEGMENT_CHANGE,
|
|
G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY |
|
|
G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstAppSrc:need-data:
|
|
* @appsrc: the appsrc element that emitted the signal
|
|
* @length: the amount of bytes needed.
|
|
*
|
|
* Signal that the source needs more data. In the callback or from another
|
|
* thread you should call push-buffer or end-of-stream.
|
|
*
|
|
* @length is just a hint and when it is set to -1, any number of bytes can be
|
|
* pushed into @appsrc.
|
|
*
|
|
* You can call push-buffer multiple times until the enough-data signal is
|
|
* fired.
|
|
*/
|
|
gst_app_src_signals[SIGNAL_NEED_DATA] =
|
|
g_signal_new ("need-data", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
|
|
G_STRUCT_OFFSET (GstAppSrcClass, need_data),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstAppSrc:enough-data:
|
|
* @appsrc: the appsrc element that emitted the signal
|
|
*
|
|
* Signal that the source has enough data. It is recommended that the
|
|
* application stops calling push-buffer until the need-data signal is
|
|
* emitted again to avoid excessive buffer queueing.
|
|
*/
|
|
gst_app_src_signals[SIGNAL_ENOUGH_DATA] =
|
|
g_signal_new ("enough-data", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
|
|
G_STRUCT_OFFSET (GstAppSrcClass, enough_data),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
/**
|
|
* GstAppSrc:seek-data:
|
|
* @appsrc: the appsrc element that emitted the signal
|
|
* @offset: the offset to seek to
|
|
*
|
|
* Seek to the given offset. The next push-buffer should produce buffers from
|
|
* the new @offset.
|
|
* This callback is only called for seekable stream types.
|
|
*
|
|
* Returns: %TRUE if the seek succeeded.
|
|
*/
|
|
gst_app_src_signals[SIGNAL_SEEK_DATA] =
|
|
g_signal_new ("seek-data", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
|
|
G_STRUCT_OFFSET (GstAppSrcClass, seek_data),
|
|
NULL, NULL, NULL, G_TYPE_BOOLEAN, 1, G_TYPE_UINT64);
|
|
|
|
/**
|
|
* GstAppSrc:push-buffer:
|
|
* @appsrc: the appsrc
|
|
* @buffer: a buffer to push
|
|
*
|
|
* Adds a buffer to the queue of buffers that the appsrc element will
|
|
* push to its source pad. This function does not take ownership of the
|
|
* buffer so the buffer needs to be unreffed after calling this function.
|
|
*
|
|
* When the block property is TRUE, this function can block until free space
|
|
* becomes available in the queue.
|
|
*/
|
|
gst_app_src_signals[SIGNAL_PUSH_BUFFER] =
|
|
g_signal_new ("push-buffer", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstAppSrcClass,
|
|
push_buffer), NULL, NULL, NULL,
|
|
GST_TYPE_FLOW_RETURN, 1, GST_TYPE_BUFFER);
|
|
|
|
/**
|
|
* GstAppSrc:push-buffer-list:
|
|
* @appsrc: the appsrc
|
|
* @buffer_list: a buffer list to push
|
|
*
|
|
* Adds a buffer list to the queue of buffers and buffer lists that the
|
|
* appsrc element will push to its source pad. This function does not take
|
|
* ownership of the buffer list so the buffer list needs to be unreffed
|
|
* after calling this function.
|
|
*
|
|
* When the block property is TRUE, this function can block until free space
|
|
* becomes available in the queue.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gst_app_src_signals[SIGNAL_PUSH_BUFFER_LIST] =
|
|
g_signal_new ("push-buffer-list", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstAppSrcClass,
|
|
push_buffer_list), NULL, NULL, NULL,
|
|
GST_TYPE_FLOW_RETURN, 1, GST_TYPE_BUFFER_LIST);
|
|
|
|
/**
|
|
* GstAppSrc:push-sample:
|
|
* @appsrc: the appsrc
|
|
* @sample: a sample from which extract buffer to push
|
|
*
|
|
* Extract a buffer from the provided sample and adds the extracted buffer
|
|
* to the queue of buffers that the appsrc element will
|
|
* push to its source pad. This function set the appsrc caps based on the caps
|
|
* in the sample and reset the caps if they change.
|
|
* Only the caps and the buffer of the provided sample are used and not
|
|
* for example the segment in the sample.
|
|
* This function does not take ownership of the
|
|
* sample so the sample needs to be unreffed after calling this function.
|
|
*
|
|
* When the block property is TRUE, this function can block until free space
|
|
* becomes available in the queue.
|
|
*
|
|
* Since: 1.6
|
|
*/
|
|
gst_app_src_signals[SIGNAL_PUSH_SAMPLE] =
|
|
g_signal_new ("push-sample", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstAppSrcClass,
|
|
push_sample), NULL, NULL, NULL,
|
|
GST_TYPE_FLOW_RETURN, 1, GST_TYPE_SAMPLE);
|
|
|
|
|
|
/**
|
|
* GstAppSrc:end-of-stream:
|
|
* @appsrc: the appsrc
|
|
*
|
|
* Notify @appsrc that no more buffer are available.
|
|
*/
|
|
gst_app_src_signals[SIGNAL_END_OF_STREAM] =
|
|
g_signal_new ("end-of-stream", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstAppSrcClass,
|
|
end_of_stream), NULL, NULL, NULL,
|
|
GST_TYPE_FLOW_RETURN, 0, G_TYPE_NONE);
|
|
|
|
gst_element_class_set_static_metadata (element_class, "AppSrc",
|
|
"Generic/Source", "Allow the application to feed buffers to a pipeline",
|
|
"David Schleef <ds@schleef.org>, Wim Taymans <wim.taymans@gmail.com>");
|
|
|
|
gst_element_class_add_static_pad_template (element_class,
|
|
&gst_app_src_template);
|
|
|
|
element_class->send_event = gst_app_src_send_event;
|
|
|
|
basesrc_class->negotiate = gst_app_src_negotiate;
|
|
basesrc_class->get_caps = gst_app_src_internal_get_caps;
|
|
basesrc_class->create = gst_app_src_create;
|
|
basesrc_class->start = gst_app_src_start;
|
|
basesrc_class->stop = gst_app_src_stop;
|
|
basesrc_class->unlock = gst_app_src_unlock;
|
|
basesrc_class->unlock_stop = gst_app_src_unlock_stop;
|
|
basesrc_class->do_seek = gst_app_src_do_seek;
|
|
basesrc_class->is_seekable = gst_app_src_is_seekable;
|
|
basesrc_class->get_size = gst_app_src_do_get_size;
|
|
basesrc_class->query = gst_app_src_query;
|
|
basesrc_class->event = gst_app_src_event;
|
|
|
|
klass->push_buffer = gst_app_src_push_buffer_action;
|
|
klass->push_buffer_list = gst_app_src_push_buffer_list_action;
|
|
klass->push_sample = gst_app_src_push_sample_action;
|
|
klass->end_of_stream = gst_app_src_end_of_stream;
|
|
}
|
|
|
|
static void
|
|
gst_app_src_init (GstAppSrc * appsrc)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
|
|
priv = appsrc->priv = gst_app_src_get_instance_private (appsrc);
|
|
|
|
g_mutex_init (&priv->mutex);
|
|
g_cond_init (&priv->cond);
|
|
priv->queue = gst_queue_array_new (16);
|
|
priv->wait_status = NOONE_WAITING;
|
|
|
|
priv->size = DEFAULT_PROP_SIZE;
|
|
priv->duration = DEFAULT_PROP_DURATION;
|
|
priv->stream_type = DEFAULT_PROP_STREAM_TYPE;
|
|
priv->max_bytes = DEFAULT_PROP_MAX_BYTES;
|
|
priv->format = DEFAULT_PROP_FORMAT;
|
|
priv->block = DEFAULT_PROP_BLOCK;
|
|
priv->min_latency = DEFAULT_PROP_MIN_LATENCY;
|
|
priv->max_latency = DEFAULT_PROP_MAX_LATENCY;
|
|
priv->emit_signals = DEFAULT_PROP_EMIT_SIGNALS;
|
|
priv->min_percent = DEFAULT_PROP_MIN_PERCENT;
|
|
priv->handle_segment_change = DEFAULT_PROP_HANDLE_SEGMENT_CHANGE;
|
|
|
|
gst_base_src_set_live (GST_BASE_SRC (appsrc), DEFAULT_PROP_IS_LIVE);
|
|
}
|
|
|
|
/* Must be called with priv->mutex */
|
|
static void
|
|
gst_app_src_flush_queued (GstAppSrc * src, gboolean retain_last_caps)
|
|
{
|
|
GstMiniObject *obj;
|
|
GstAppSrcPrivate *priv = src->priv;
|
|
GstCaps *requeue_caps = NULL;
|
|
|
|
while (!gst_queue_array_is_empty (priv->queue)) {
|
|
obj = gst_queue_array_pop_head (priv->queue);
|
|
if (obj) {
|
|
if (GST_IS_CAPS (obj) && retain_last_caps) {
|
|
gst_caps_replace (&requeue_caps, GST_CAPS_CAST (obj));
|
|
}
|
|
gst_mini_object_unref (obj);
|
|
}
|
|
}
|
|
|
|
if (requeue_caps) {
|
|
gst_queue_array_push_tail (priv->queue, requeue_caps);
|
|
}
|
|
|
|
priv->queued_bytes = 0;
|
|
}
|
|
|
|
static void
|
|
gst_app_src_dispose (GObject * obj)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (obj);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
Callbacks *callbacks = NULL;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
if (priv->current_caps) {
|
|
gst_caps_unref (priv->current_caps);
|
|
priv->current_caps = NULL;
|
|
}
|
|
if (priv->last_caps) {
|
|
gst_caps_unref (priv->last_caps);
|
|
priv->last_caps = NULL;
|
|
}
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
if (priv->callbacks)
|
|
callbacks = g_steal_pointer (&priv->callbacks);
|
|
gst_app_src_flush_queued (appsrc, FALSE);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
g_clear_pointer (&callbacks, callbacks_unref);
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (obj);
|
|
}
|
|
|
|
static void
|
|
gst_app_src_finalize (GObject * obj)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (obj);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
g_mutex_clear (&priv->mutex);
|
|
g_cond_clear (&priv->cond);
|
|
gst_queue_array_free (priv->queue);
|
|
|
|
g_free (priv->uri);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (obj);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_app_src_internal_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC (bsrc);
|
|
GstCaps *caps;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
if ((caps = appsrc->priv->current_caps))
|
|
gst_caps_ref (caps);
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
if (filter) {
|
|
if (caps) {
|
|
GstCaps *intersection =
|
|
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (caps);
|
|
caps = intersection;
|
|
} else {
|
|
caps = gst_caps_ref (filter);
|
|
}
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (appsrc, "caps: %" GST_PTR_FORMAT, caps);
|
|
return caps;
|
|
}
|
|
|
|
static void
|
|
gst_app_src_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (object);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_CAPS:
|
|
gst_app_src_set_caps (appsrc, gst_value_get_caps (value));
|
|
break;
|
|
case PROP_SIZE:
|
|
gst_app_src_set_size (appsrc, g_value_get_int64 (value));
|
|
break;
|
|
case PROP_STREAM_TYPE:
|
|
gst_app_src_set_stream_type (appsrc, g_value_get_enum (value));
|
|
break;
|
|
case PROP_MAX_BYTES:
|
|
gst_app_src_set_max_bytes (appsrc, g_value_get_uint64 (value));
|
|
break;
|
|
case PROP_FORMAT:
|
|
priv->format = g_value_get_enum (value);
|
|
break;
|
|
case PROP_BLOCK:
|
|
priv->block = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_IS_LIVE:
|
|
gst_base_src_set_live (GST_BASE_SRC (appsrc),
|
|
g_value_get_boolean (value));
|
|
break;
|
|
case PROP_MIN_LATENCY:
|
|
gst_app_src_set_latencies (appsrc, TRUE, g_value_get_int64 (value),
|
|
FALSE, -1);
|
|
break;
|
|
case PROP_MAX_LATENCY:
|
|
gst_app_src_set_latencies (appsrc, FALSE, -1, TRUE,
|
|
g_value_get_int64 (value));
|
|
break;
|
|
case PROP_EMIT_SIGNALS:
|
|
gst_app_src_set_emit_signals (appsrc, g_value_get_boolean (value));
|
|
break;
|
|
case PROP_MIN_PERCENT:
|
|
priv->min_percent = g_value_get_uint (value);
|
|
break;
|
|
case PROP_DURATION:
|
|
gst_app_src_set_duration (appsrc, g_value_get_uint64 (value));
|
|
break;
|
|
case PROP_HANDLE_SEGMENT_CHANGE:
|
|
priv->handle_segment_change = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_app_src_get_property (GObject * object, guint prop_id, GValue * value,
|
|
GParamSpec * pspec)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (object);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_CAPS:
|
|
g_value_take_boxed (value, gst_app_src_get_caps (appsrc));
|
|
break;
|
|
case PROP_SIZE:
|
|
g_value_set_int64 (value, gst_app_src_get_size (appsrc));
|
|
break;
|
|
case PROP_STREAM_TYPE:
|
|
g_value_set_enum (value, gst_app_src_get_stream_type (appsrc));
|
|
break;
|
|
case PROP_MAX_BYTES:
|
|
g_value_set_uint64 (value, gst_app_src_get_max_bytes (appsrc));
|
|
break;
|
|
case PROP_FORMAT:
|
|
g_value_set_enum (value, priv->format);
|
|
break;
|
|
case PROP_BLOCK:
|
|
g_value_set_boolean (value, priv->block);
|
|
break;
|
|
case PROP_IS_LIVE:
|
|
g_value_set_boolean (value, gst_base_src_is_live (GST_BASE_SRC (appsrc)));
|
|
break;
|
|
case PROP_MIN_LATENCY:
|
|
{
|
|
guint64 min = 0;
|
|
|
|
gst_app_src_get_latency (appsrc, &min, NULL);
|
|
g_value_set_int64 (value, min);
|
|
break;
|
|
}
|
|
case PROP_MAX_LATENCY:
|
|
{
|
|
guint64 max = 0;
|
|
|
|
gst_app_src_get_latency (appsrc, NULL, &max);
|
|
g_value_set_int64 (value, max);
|
|
break;
|
|
}
|
|
case PROP_EMIT_SIGNALS:
|
|
g_value_set_boolean (value, gst_app_src_get_emit_signals (appsrc));
|
|
break;
|
|
case PROP_MIN_PERCENT:
|
|
g_value_set_uint (value, priv->min_percent);
|
|
break;
|
|
case PROP_CURRENT_LEVEL_BYTES:
|
|
g_value_set_uint64 (value, gst_app_src_get_current_level_bytes (appsrc));
|
|
break;
|
|
case PROP_DURATION:
|
|
g_value_set_uint64 (value, gst_app_src_get_duration (appsrc));
|
|
break;
|
|
case PROP_HANDLE_SEGMENT_CHANGE:
|
|
g_value_set_boolean (value, priv->handle_segment_change);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_send_event (GstElement * element, GstEvent * event)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (element);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
g_mutex_lock (&priv->mutex);
|
|
gst_app_src_flush_queued (appsrc, TRUE);
|
|
g_mutex_unlock (&priv->mutex);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS, send_event, (element,
|
|
event), FALSE);
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_unlock (GstBaseSrc * bsrc)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (bsrc);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
GST_DEBUG_OBJECT (appsrc, "unlock start");
|
|
priv->flushing = TRUE;
|
|
g_cond_broadcast (&priv->cond);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_unlock_stop (GstBaseSrc * bsrc)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (bsrc);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
GST_DEBUG_OBJECT (appsrc, "unlock stop");
|
|
priv->flushing = FALSE;
|
|
g_cond_broadcast (&priv->cond);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_start (GstBaseSrc * bsrc)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (bsrc);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
GST_DEBUG_OBJECT (appsrc, "starting");
|
|
priv->started = TRUE;
|
|
/* set the offset to -1 so that we always do a first seek. This is only used
|
|
* in random-access mode. */
|
|
priv->offset = -1;
|
|
priv->flushing = FALSE;
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
gst_base_src_set_format (bsrc, priv->format);
|
|
gst_segment_init (&priv->last_segment, priv->format);
|
|
gst_segment_init (&priv->current_segment, priv->format);
|
|
priv->pending_custom_segment = FALSE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_stop (GstBaseSrc * bsrc)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (bsrc);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
GST_DEBUG_OBJECT (appsrc, "stopping");
|
|
priv->is_eos = FALSE;
|
|
priv->flushing = TRUE;
|
|
priv->started = FALSE;
|
|
gst_app_src_flush_queued (appsrc, TRUE);
|
|
g_cond_broadcast (&priv->cond);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_is_seekable (GstBaseSrc * src)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (src);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
gboolean res = FALSE;
|
|
|
|
switch (priv->stream_type) {
|
|
case GST_APP_STREAM_TYPE_STREAM:
|
|
break;
|
|
case GST_APP_STREAM_TYPE_SEEKABLE:
|
|
case GST_APP_STREAM_TYPE_RANDOM_ACCESS:
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_do_get_size (GstBaseSrc * src, guint64 * size)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (src);
|
|
|
|
*size = gst_app_src_get_size (appsrc);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_query (GstBaseSrc * src, GstQuery * query)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (src);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
gboolean res;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
GstClockTime min, max;
|
|
gboolean live;
|
|
|
|
/* Query the parent class for the defaults */
|
|
res = gst_base_src_query_latency (src, &live, &min, &max);
|
|
|
|
/* overwrite with our values when we need to */
|
|
g_mutex_lock (&priv->mutex);
|
|
if (priv->min_latency != -1) {
|
|
min = priv->min_latency;
|
|
max = priv->max_latency;
|
|
}
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
gst_query_set_latency (query, live, min, max);
|
|
break;
|
|
}
|
|
case GST_QUERY_SCHEDULING:
|
|
{
|
|
gst_query_set_scheduling (query, GST_SCHEDULING_FLAG_SEEKABLE, 1, -1, 0);
|
|
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH);
|
|
|
|
switch (priv->stream_type) {
|
|
case GST_APP_STREAM_TYPE_STREAM:
|
|
case GST_APP_STREAM_TYPE_SEEKABLE:
|
|
break;
|
|
case GST_APP_STREAM_TYPE_RANDOM_ACCESS:
|
|
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PULL);
|
|
break;
|
|
}
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
case GST_QUERY_DURATION:
|
|
{
|
|
GstFormat format;
|
|
gst_query_parse_duration (query, &format, NULL);
|
|
if (format == GST_FORMAT_BYTES) {
|
|
gst_query_set_duration (query, format, priv->size);
|
|
res = TRUE;
|
|
} else if (format == GST_FORMAT_TIME) {
|
|
if (priv->duration != GST_CLOCK_TIME_NONE) {
|
|
gst_query_set_duration (query, format, priv->duration);
|
|
res = TRUE;
|
|
} else {
|
|
res = FALSE;
|
|
}
|
|
} else {
|
|
res = FALSE;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = GST_BASE_SRC_CLASS (parent_class)->query (src, query);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/* will be called in push mode */
|
|
static gboolean
|
|
gst_app_src_do_seek (GstBaseSrc * src, GstSegment * segment)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (src);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
gint64 desired_position;
|
|
gboolean res = FALSE;
|
|
gboolean emit;
|
|
Callbacks *callbacks = NULL;
|
|
|
|
desired_position = segment->position;
|
|
|
|
/* no need to try to seek in streaming mode */
|
|
if (priv->stream_type == GST_APP_STREAM_TYPE_STREAM)
|
|
return TRUE;
|
|
|
|
GST_DEBUG_OBJECT (appsrc, "seeking to %" G_GINT64_FORMAT ", format %s",
|
|
desired_position, gst_format_get_name (segment->format));
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
emit = priv->emit_signals;
|
|
if (priv->callbacks)
|
|
callbacks = callbacks_ref (priv->callbacks);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
if (callbacks && callbacks->callbacks.seek_data) {
|
|
res =
|
|
callbacks->callbacks.seek_data (appsrc, desired_position,
|
|
callbacks->user_data);
|
|
} else if (emit) {
|
|
g_signal_emit (appsrc, gst_app_src_signals[SIGNAL_SEEK_DATA], 0,
|
|
desired_position, &res);
|
|
}
|
|
|
|
g_clear_pointer (&callbacks, callbacks_unref);
|
|
|
|
if (res) {
|
|
GST_DEBUG_OBJECT (appsrc, "flushing queue");
|
|
g_mutex_lock (&priv->mutex);
|
|
gst_app_src_flush_queued (appsrc, TRUE);
|
|
gst_segment_copy_into (segment, &priv->last_segment);
|
|
gst_segment_copy_into (segment, &priv->current_segment);
|
|
priv->pending_custom_segment = FALSE;
|
|
g_mutex_unlock (&priv->mutex);
|
|
priv->is_eos = FALSE;
|
|
} else {
|
|
GST_WARNING_OBJECT (appsrc, "seek failed");
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
/* must be called with the appsrc mutex */
|
|
static gboolean
|
|
gst_app_src_emit_seek (GstAppSrc * appsrc, guint64 offset)
|
|
{
|
|
gboolean res = FALSE;
|
|
gboolean emit;
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
Callbacks *callbacks = NULL;
|
|
|
|
emit = priv->emit_signals;
|
|
if (priv->callbacks)
|
|
callbacks = callbacks_ref (priv->callbacks);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
GST_DEBUG_OBJECT (appsrc,
|
|
"we are at %" G_GINT64_FORMAT ", seek to %" G_GINT64_FORMAT,
|
|
priv->offset, offset);
|
|
|
|
if (callbacks && callbacks->callbacks.seek_data)
|
|
res = callbacks->callbacks.seek_data (appsrc, offset, callbacks->user_data);
|
|
else if (emit)
|
|
g_signal_emit (appsrc, gst_app_src_signals[SIGNAL_SEEK_DATA], 0,
|
|
offset, &res);
|
|
|
|
g_clear_pointer (&callbacks, callbacks_unref);
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
|
|
return res;
|
|
}
|
|
|
|
/* must be called with the appsrc mutex. After this call things can be
|
|
* flushing */
|
|
static void
|
|
gst_app_src_emit_need_data (GstAppSrc * appsrc, guint size)
|
|
{
|
|
gboolean emit;
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
Callbacks *callbacks = NULL;
|
|
|
|
emit = priv->emit_signals;
|
|
if (priv->callbacks)
|
|
callbacks = callbacks_ref (priv->callbacks);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
/* we have no data, we need some. We fire the signal with the size hint. */
|
|
if (callbacks && callbacks->callbacks.need_data)
|
|
callbacks->callbacks.need_data (appsrc, size, callbacks->user_data);
|
|
else if (emit)
|
|
g_signal_emit (appsrc, gst_app_src_signals[SIGNAL_NEED_DATA], 0, size,
|
|
NULL);
|
|
|
|
g_clear_pointer (&callbacks, callbacks_unref);
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
/* we can be flushing now because we released the lock */
|
|
}
|
|
|
|
/* must be called with the appsrc mutex */
|
|
static gboolean
|
|
gst_app_src_do_negotiate (GstBaseSrc * basesrc)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (basesrc);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
gboolean result;
|
|
GstCaps *caps;
|
|
|
|
GST_OBJECT_LOCK (basesrc);
|
|
caps = priv->current_caps ? gst_caps_ref (priv->current_caps) : NULL;
|
|
GST_OBJECT_UNLOCK (basesrc);
|
|
|
|
/* Avoid deadlock by unlocking mutex
|
|
* otherwise we get deadlock between this and stream lock */
|
|
g_mutex_unlock (&priv->mutex);
|
|
if (caps) {
|
|
result = gst_base_src_set_caps (basesrc, caps);
|
|
gst_caps_unref (caps);
|
|
} else {
|
|
result = GST_BASE_SRC_CLASS (parent_class)->negotiate (basesrc);
|
|
}
|
|
g_mutex_lock (&priv->mutex);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_negotiate (GstBaseSrc * basesrc)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (basesrc);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
gboolean result;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
result = gst_app_src_do_negotiate (basesrc);
|
|
g_mutex_unlock (&priv->mutex);
|
|
return result;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_app_src_create (GstBaseSrc * bsrc, guint64 offset, guint size,
|
|
GstBuffer ** buf)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (bsrc);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
GstFlowReturn ret;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
if (G_UNLIKELY (priv->size != bsrc->segment.duration &&
|
|
bsrc->segment.format == GST_FORMAT_BYTES)) {
|
|
GST_DEBUG_OBJECT (appsrc,
|
|
"Size changed from %" G_GINT64_FORMAT " to %" G_GINT64_FORMAT,
|
|
bsrc->segment.duration, priv->size);
|
|
bsrc->segment.duration = priv->size;
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
gst_element_post_message (GST_ELEMENT (appsrc),
|
|
gst_message_new_duration_changed (GST_OBJECT (appsrc)));
|
|
} else if (G_UNLIKELY (priv->duration != bsrc->segment.duration &&
|
|
bsrc->segment.format == GST_FORMAT_TIME)) {
|
|
GST_DEBUG_OBJECT (appsrc,
|
|
"Duration changed from %" GST_TIME_FORMAT " to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (bsrc->segment.duration), GST_TIME_ARGS (priv->duration));
|
|
bsrc->segment.duration = priv->duration;
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
gst_element_post_message (GST_ELEMENT (appsrc),
|
|
gst_message_new_duration_changed (GST_OBJECT (appsrc)));
|
|
} else {
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
}
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
/* check flushing first */
|
|
if (G_UNLIKELY (priv->flushing))
|
|
goto flushing;
|
|
|
|
if (priv->stream_type == GST_APP_STREAM_TYPE_RANDOM_ACCESS) {
|
|
/* if we are dealing with a random-access stream, issue a seek if the offset
|
|
* changed. */
|
|
if (G_UNLIKELY (priv->offset != offset)) {
|
|
gboolean res;
|
|
|
|
/* do the seek */
|
|
res = gst_app_src_emit_seek (appsrc, offset);
|
|
|
|
if (G_UNLIKELY (!res))
|
|
/* failing to seek is fatal */
|
|
goto seek_error;
|
|
|
|
priv->offset = offset;
|
|
priv->is_eos = FALSE;
|
|
}
|
|
}
|
|
|
|
while (TRUE) {
|
|
/* return data as long as we have some */
|
|
if (!gst_queue_array_is_empty (priv->queue)) {
|
|
guint buf_size;
|
|
GstMiniObject *obj = gst_queue_array_pop_head (priv->queue);
|
|
|
|
if (GST_IS_CAPS (obj)) {
|
|
GstCaps *next_caps = GST_CAPS (obj);
|
|
gboolean caps_changed = TRUE;
|
|
|
|
if (next_caps && priv->current_caps)
|
|
caps_changed = !gst_caps_is_equal (next_caps, priv->current_caps);
|
|
else
|
|
caps_changed = (next_caps != priv->current_caps);
|
|
|
|
gst_caps_replace (&priv->current_caps, next_caps);
|
|
|
|
if (next_caps) {
|
|
gst_caps_unref (next_caps);
|
|
}
|
|
|
|
if (caps_changed)
|
|
gst_app_src_do_negotiate (bsrc);
|
|
|
|
/* Lock has released so now may need
|
|
*- flushing
|
|
*- new caps change
|
|
*- check queue has data */
|
|
if (G_UNLIKELY (priv->flushing))
|
|
goto flushing;
|
|
|
|
/* Continue checks caps and queue */
|
|
continue;
|
|
}
|
|
|
|
if (GST_IS_BUFFER (obj)) {
|
|
*buf = GST_BUFFER (obj);
|
|
buf_size = gst_buffer_get_size (*buf);
|
|
GST_LOG_OBJECT (appsrc, "have buffer %p of size %u", *buf, buf_size);
|
|
} else if (GST_IS_BUFFER_LIST (obj)) {
|
|
GstBufferList *buffer_list;
|
|
|
|
buffer_list = GST_BUFFER_LIST (obj);
|
|
|
|
buf_size = gst_buffer_list_calculate_size (buffer_list);
|
|
|
|
GST_LOG_OBJECT (appsrc, "have buffer list %p of size %u, %u buffers",
|
|
buffer_list, buf_size, gst_buffer_list_length (buffer_list));
|
|
|
|
gst_base_src_submit_buffer_list (bsrc, buffer_list);
|
|
*buf = NULL;
|
|
} else if (GST_IS_EVENT (obj)) {
|
|
GstEvent *event = GST_EVENT (obj);
|
|
const GstSegment *segment = NULL;
|
|
|
|
gst_event_parse_segment (event, &segment);
|
|
g_assert (segment != NULL);
|
|
|
|
if (!gst_segment_is_equal (&priv->current_segment, segment)) {
|
|
GST_DEBUG_OBJECT (appsrc,
|
|
"Update new segment %" GST_PTR_FORMAT, event);
|
|
if (!gst_base_src_new_segment (bsrc, segment)) {
|
|
GST_ERROR_OBJECT (appsrc,
|
|
"Couldn't set new segment %" GST_PTR_FORMAT, event);
|
|
gst_event_unref (event);
|
|
goto invalid_segment;
|
|
}
|
|
gst_segment_copy_into (segment, &priv->current_segment);
|
|
}
|
|
|
|
gst_event_unref (event);
|
|
continue;
|
|
} else {
|
|
g_assert_not_reached ();
|
|
}
|
|
|
|
priv->queued_bytes -= buf_size;
|
|
|
|
/* only update the offset when in random_access mode */
|
|
if (priv->stream_type == GST_APP_STREAM_TYPE_RANDOM_ACCESS)
|
|
priv->offset += buf_size;
|
|
|
|
/* signal that we removed an item */
|
|
if ((priv->wait_status & APP_WAITING))
|
|
g_cond_broadcast (&priv->cond);
|
|
|
|
/* see if we go lower than the min-percent */
|
|
if (priv->min_percent && priv->max_bytes) {
|
|
if (priv->queued_bytes * 100 / priv->max_bytes <= priv->min_percent)
|
|
/* ignore flushing state, we got a buffer and we will return it now.
|
|
* Errors will be handled in the next round */
|
|
gst_app_src_emit_need_data (appsrc, size);
|
|
}
|
|
ret = GST_FLOW_OK;
|
|
break;
|
|
} else {
|
|
gst_app_src_emit_need_data (appsrc, size);
|
|
|
|
/* we can be flushing now because we released the lock above */
|
|
if (G_UNLIKELY (priv->flushing))
|
|
goto flushing;
|
|
|
|
/* if we have a buffer now, continue the loop and try to return it. In
|
|
* random-access mode (where a buffer is normally pushed in the above
|
|
* signal) we can still be empty because the pushed buffer got flushed or
|
|
* when the application pushes the requested buffer later, we support both
|
|
* possibilities. */
|
|
if (!gst_queue_array_is_empty (priv->queue))
|
|
continue;
|
|
|
|
/* no buffer yet, maybe we are EOS, if not, block for more data. */
|
|
}
|
|
|
|
/* check EOS */
|
|
if (G_UNLIKELY (priv->is_eos))
|
|
goto eos;
|
|
|
|
/* nothing to return, wait a while for new data or flushing. */
|
|
priv->wait_status |= STREAM_WAITING;
|
|
g_cond_wait (&priv->cond, &priv->mutex);
|
|
priv->wait_status &= ~STREAM_WAITING;
|
|
}
|
|
g_mutex_unlock (&priv->mutex);
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (appsrc, "we are flushing");
|
|
g_mutex_unlock (&priv->mutex);
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
eos:
|
|
{
|
|
GST_DEBUG_OBJECT (appsrc, "we are EOS");
|
|
g_mutex_unlock (&priv->mutex);
|
|
return GST_FLOW_EOS;
|
|
}
|
|
seek_error:
|
|
{
|
|
g_mutex_unlock (&priv->mutex);
|
|
GST_ELEMENT_ERROR (appsrc, RESOURCE, READ, ("failed to seek"),
|
|
GST_ERROR_SYSTEM);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
invalid_segment:
|
|
{
|
|
g_mutex_unlock (&priv->mutex);
|
|
GST_ELEMENT_ERROR (appsrc, LIBRARY, SETTINGS,
|
|
(NULL), ("Failed to configure the provided input segment."));
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
/* external API */
|
|
|
|
/**
|
|
* gst_app_src_set_caps:
|
|
* @appsrc: a #GstAppSrc
|
|
* @caps: caps to set
|
|
*
|
|
* Set the capabilities on the appsrc element. This function takes
|
|
* a copy of the caps structure. After calling this method, the source will
|
|
* only produce caps that match @caps. @caps must be fixed and the caps on the
|
|
* buffers must match the caps or left NULL.
|
|
*/
|
|
void
|
|
gst_app_src_set_caps (GstAppSrc * appsrc, const GstCaps * caps)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
gboolean caps_changed;
|
|
|
|
g_return_if_fail (GST_IS_APP_SRC (appsrc));
|
|
|
|
priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
if (caps && priv->last_caps)
|
|
caps_changed = !gst_caps_is_equal (caps, priv->last_caps);
|
|
else
|
|
caps_changed = (caps != priv->last_caps);
|
|
|
|
if (caps_changed) {
|
|
GstCaps *new_caps;
|
|
gpointer t;
|
|
|
|
new_caps = caps ? gst_caps_copy (caps) : NULL;
|
|
GST_DEBUG_OBJECT (appsrc, "setting caps to %" GST_PTR_FORMAT, caps);
|
|
|
|
while ((t = gst_queue_array_peek_tail (priv->queue)) && GST_IS_CAPS (t)) {
|
|
gst_caps_unref (gst_queue_array_pop_tail (priv->queue));
|
|
}
|
|
gst_queue_array_push_tail (priv->queue, new_caps);
|
|
gst_caps_replace (&priv->last_caps, new_caps);
|
|
|
|
if ((priv->wait_status & STREAM_WAITING))
|
|
g_cond_broadcast (&priv->cond);
|
|
}
|
|
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
g_mutex_unlock (&priv->mutex);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_get_caps:
|
|
* @appsrc: a #GstAppSrc
|
|
*
|
|
* Get the configured caps on @appsrc.
|
|
*
|
|
* Returns: the #GstCaps produced by the source. gst_caps_unref() after usage.
|
|
*/
|
|
GstCaps *
|
|
gst_app_src_get_caps (GstAppSrc * appsrc)
|
|
{
|
|
|
|
GstCaps *caps;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), NULL);
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
if ((caps = appsrc->priv->last_caps))
|
|
gst_caps_ref (caps);
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
return caps;
|
|
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_set_size:
|
|
* @appsrc: a #GstAppSrc
|
|
* @size: the size to set
|
|
*
|
|
* Set the size of the stream in bytes. A value of -1 means that the size is
|
|
* not known.
|
|
*/
|
|
void
|
|
gst_app_src_set_size (GstAppSrc * appsrc, gint64 size)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_APP_SRC (appsrc));
|
|
|
|
priv = appsrc->priv;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
GST_DEBUG_OBJECT (appsrc, "setting size of %" G_GINT64_FORMAT, size);
|
|
priv->size = size;
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_get_size:
|
|
* @appsrc: a #GstAppSrc
|
|
*
|
|
* Get the size of the stream in bytes. A value of -1 means that the size is
|
|
* not known.
|
|
*
|
|
* Returns: the size of the stream previously set with gst_app_src_set_size();
|
|
*/
|
|
gint64
|
|
gst_app_src_get_size (GstAppSrc * appsrc)
|
|
{
|
|
gint64 size;
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), -1);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
size = priv->size;
|
|
GST_DEBUG_OBJECT (appsrc, "getting size of %" G_GINT64_FORMAT, size);
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
return size;
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_set_duration:
|
|
* @appsrc: a #GstAppSrc
|
|
* @duration: the duration to set
|
|
*
|
|
* Set the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is
|
|
* not known.
|
|
*
|
|
* Since: 1.10
|
|
*/
|
|
void
|
|
gst_app_src_set_duration (GstAppSrc * appsrc, GstClockTime duration)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_APP_SRC (appsrc));
|
|
|
|
priv = appsrc->priv;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
GST_DEBUG_OBJECT (appsrc, "setting duration of %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (duration));
|
|
priv->duration = duration;
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_get_duration:
|
|
* @appsrc: a #GstAppSrc
|
|
*
|
|
* Get the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is
|
|
* not known.
|
|
*
|
|
* Returns: the duration of the stream previously set with gst_app_src_set_duration();
|
|
*
|
|
* Since: 1.10
|
|
*/
|
|
GstClockTime
|
|
gst_app_src_get_duration (GstAppSrc * appsrc)
|
|
{
|
|
GstClockTime duration;
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), GST_CLOCK_TIME_NONE);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
duration = priv->duration;
|
|
GST_DEBUG_OBJECT (appsrc, "getting duration of %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (duration));
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
return duration;
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_set_stream_type:
|
|
* @appsrc: a #GstAppSrc
|
|
* @type: the new state
|
|
*
|
|
* Set the stream type on @appsrc. For seekable streams, the "seek" signal must
|
|
* be connected to.
|
|
*
|
|
* A stream_type stream
|
|
*/
|
|
void
|
|
gst_app_src_set_stream_type (GstAppSrc * appsrc, GstAppStreamType type)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_APP_SRC (appsrc));
|
|
|
|
priv = appsrc->priv;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
GST_DEBUG_OBJECT (appsrc, "setting stream_type of %d", type);
|
|
priv->stream_type = type;
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_get_stream_type:
|
|
* @appsrc: a #GstAppSrc
|
|
*
|
|
* Get the stream type. Control the stream type of @appsrc
|
|
* with gst_app_src_set_stream_type().
|
|
*
|
|
* Returns: the stream type.
|
|
*/
|
|
GstAppStreamType
|
|
gst_app_src_get_stream_type (GstAppSrc * appsrc)
|
|
{
|
|
gboolean stream_type;
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), FALSE);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
stream_type = priv->stream_type;
|
|
GST_DEBUG_OBJECT (appsrc, "getting stream_type of %d", stream_type);
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
return stream_type;
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_set_max_bytes:
|
|
* @appsrc: a #GstAppSrc
|
|
* @max: the maximum number of bytes to queue
|
|
*
|
|
* Set the maximum amount of bytes that can be queued in @appsrc.
|
|
* After the maximum amount of bytes are queued, @appsrc will emit the
|
|
* "enough-data" signal.
|
|
*/
|
|
void
|
|
gst_app_src_set_max_bytes (GstAppSrc * appsrc, guint64 max)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_APP_SRC (appsrc));
|
|
|
|
priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
if (max != priv->max_bytes) {
|
|
GST_DEBUG_OBJECT (appsrc, "setting max-bytes to %" G_GUINT64_FORMAT, max);
|
|
priv->max_bytes = max;
|
|
/* signal the change */
|
|
g_cond_broadcast (&priv->cond);
|
|
}
|
|
g_mutex_unlock (&priv->mutex);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_get_max_bytes:
|
|
* @appsrc: a #GstAppSrc
|
|
*
|
|
* Get the maximum amount of bytes that can be queued in @appsrc.
|
|
*
|
|
* Returns: The maximum amount of bytes that can be queued.
|
|
*/
|
|
guint64
|
|
gst_app_src_get_max_bytes (GstAppSrc * appsrc)
|
|
{
|
|
guint64 result;
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), 0);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
result = priv->max_bytes;
|
|
GST_DEBUG_OBJECT (appsrc, "getting max-bytes of %" G_GUINT64_FORMAT, result);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_get_current_level_bytes:
|
|
* @appsrc: a #GstAppSrc
|
|
*
|
|
* Get the number of currently queued bytes inside @appsrc.
|
|
*
|
|
* Returns: The number of currently queued bytes.
|
|
*
|
|
* Since: 1.2
|
|
*/
|
|
guint64
|
|
gst_app_src_get_current_level_bytes (GstAppSrc * appsrc)
|
|
{
|
|
gint64 queued;
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), -1);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
GST_OBJECT_LOCK (appsrc);
|
|
queued = priv->queued_bytes;
|
|
GST_DEBUG_OBJECT (appsrc, "current level bytes is %" G_GUINT64_FORMAT,
|
|
queued);
|
|
GST_OBJECT_UNLOCK (appsrc);
|
|
|
|
return queued;
|
|
}
|
|
|
|
static void
|
|
gst_app_src_set_latencies (GstAppSrc * appsrc, gboolean do_min, guint64 min,
|
|
gboolean do_max, guint64 max)
|
|
{
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
gboolean changed = FALSE;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
if (do_min && priv->min_latency != min) {
|
|
priv->min_latency = min;
|
|
changed = TRUE;
|
|
}
|
|
if (do_max && priv->max_latency != max) {
|
|
priv->max_latency = max;
|
|
changed = TRUE;
|
|
}
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
if (changed) {
|
|
GST_DEBUG_OBJECT (appsrc, "posting latency changed");
|
|
gst_element_post_message (GST_ELEMENT_CAST (appsrc),
|
|
gst_message_new_latency (GST_OBJECT_CAST (appsrc)));
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_set_latency:
|
|
* @appsrc: a #GstAppSrc
|
|
* @min: the min latency
|
|
* @max: the max latency
|
|
*
|
|
* Configure the @min and @max latency in @src. If @min is set to -1, the
|
|
* default latency calculations for pseudo-live sources will be used.
|
|
*/
|
|
void
|
|
gst_app_src_set_latency (GstAppSrc * appsrc, guint64 min, guint64 max)
|
|
{
|
|
gst_app_src_set_latencies (appsrc, TRUE, min, TRUE, max);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_get_latency:
|
|
* @appsrc: a #GstAppSrc
|
|
* @min: (out): the min latency
|
|
* @max: (out): the max latency
|
|
*
|
|
* Retrieve the min and max latencies in @min and @max respectively.
|
|
*/
|
|
void
|
|
gst_app_src_get_latency (GstAppSrc * appsrc, guint64 * min, guint64 * max)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_APP_SRC (appsrc));
|
|
|
|
priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
if (min)
|
|
*min = priv->min_latency;
|
|
if (max)
|
|
*max = priv->max_latency;
|
|
g_mutex_unlock (&priv->mutex);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_set_emit_signals:
|
|
* @appsrc: a #GstAppSrc
|
|
* @emit: the new state
|
|
*
|
|
* Make appsrc emit the "new-preroll" and "new-buffer" signals. This option is
|
|
* by default disabled because signal emission is expensive and unneeded when
|
|
* the application prefers to operate in pull mode.
|
|
*/
|
|
void
|
|
gst_app_src_set_emit_signals (GstAppSrc * appsrc, gboolean emit)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_APP_SRC (appsrc));
|
|
|
|
priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
priv->emit_signals = emit;
|
|
g_mutex_unlock (&priv->mutex);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_get_emit_signals:
|
|
* @appsrc: a #GstAppSrc
|
|
*
|
|
* Check if appsrc will emit the "new-preroll" and "new-buffer" signals.
|
|
*
|
|
* Returns: %TRUE if @appsrc is emitting the "new-preroll" and "new-buffer"
|
|
* signals.
|
|
*/
|
|
gboolean
|
|
gst_app_src_get_emit_signals (GstAppSrc * appsrc)
|
|
{
|
|
gboolean result;
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), FALSE);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
result = priv->emit_signals;
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_app_src_push_internal (GstAppSrc * appsrc, GstBuffer * buffer,
|
|
GstBufferList * buflist, gboolean steal_ref)
|
|
{
|
|
gboolean first = TRUE;
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), GST_FLOW_ERROR);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
if (buffer != NULL)
|
|
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
|
else
|
|
g_return_val_if_fail (GST_IS_BUFFER_LIST (buflist), GST_FLOW_ERROR);
|
|
|
|
if (buflist != NULL) {
|
|
if (gst_buffer_list_length (buflist) == 0)
|
|
return GST_FLOW_OK;
|
|
|
|
buffer = gst_buffer_list_get (buflist, 0);
|
|
}
|
|
|
|
if (GST_BUFFER_DTS (buffer) == GST_CLOCK_TIME_NONE &&
|
|
GST_BUFFER_PTS (buffer) == GST_CLOCK_TIME_NONE &&
|
|
gst_base_src_get_do_timestamp (GST_BASE_SRC_CAST (appsrc))) {
|
|
GstClock *clock;
|
|
|
|
clock = gst_element_get_clock (GST_ELEMENT_CAST (appsrc));
|
|
if (clock) {
|
|
GstClockTime now;
|
|
GstClockTime base_time =
|
|
gst_element_get_base_time (GST_ELEMENT_CAST (appsrc));
|
|
|
|
now = gst_clock_get_time (clock);
|
|
if (now > base_time)
|
|
now -= base_time;
|
|
else
|
|
now = 0;
|
|
gst_object_unref (clock);
|
|
|
|
if (buflist == NULL) {
|
|
if (!steal_ref) {
|
|
buffer = gst_buffer_copy (buffer);
|
|
steal_ref = TRUE;
|
|
} else {
|
|
buffer = gst_buffer_make_writable (buffer);
|
|
}
|
|
} else {
|
|
if (!steal_ref) {
|
|
buflist = gst_buffer_list_copy (buflist);
|
|
steal_ref = TRUE;
|
|
} else {
|
|
buflist = gst_buffer_list_make_writable (buflist);
|
|
}
|
|
buffer = gst_buffer_list_get_writable (buflist, 0);
|
|
}
|
|
|
|
GST_BUFFER_PTS (buffer) = now;
|
|
GST_BUFFER_DTS (buffer) = now;
|
|
} else {
|
|
GST_WARNING_OBJECT (appsrc,
|
|
"do-timestamp=TRUE but buffers are provided before "
|
|
"reaching the PLAYING state and having a clock. Timestamps will "
|
|
"not be accurate!");
|
|
}
|
|
}
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
|
|
while (TRUE) {
|
|
/* can't accept buffers when we are flushing or EOS */
|
|
if (priv->flushing)
|
|
goto flushing;
|
|
|
|
if (priv->is_eos)
|
|
goto eos;
|
|
|
|
if (priv->max_bytes && priv->queued_bytes >= priv->max_bytes) {
|
|
GST_DEBUG_OBJECT (appsrc,
|
|
"queue filled (%" G_GUINT64_FORMAT " >= %" G_GUINT64_FORMAT ")",
|
|
priv->queued_bytes, priv->max_bytes);
|
|
|
|
if (first) {
|
|
Callbacks *callbacks = NULL;
|
|
gboolean emit;
|
|
|
|
emit = priv->emit_signals;
|
|
if (priv->callbacks)
|
|
callbacks = callbacks_ref (priv->callbacks);
|
|
/* only signal on the first push */
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
if (callbacks && callbacks->callbacks.enough_data)
|
|
callbacks->callbacks.enough_data (appsrc, callbacks->user_data);
|
|
else if (emit)
|
|
g_signal_emit (appsrc, gst_app_src_signals[SIGNAL_ENOUGH_DATA], 0,
|
|
NULL);
|
|
|
|
g_clear_pointer (&callbacks, callbacks_unref);
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
/* continue to check for flushing/eos after releasing the lock */
|
|
first = FALSE;
|
|
continue;
|
|
}
|
|
if (priv->block) {
|
|
GST_DEBUG_OBJECT (appsrc, "waiting for free space");
|
|
/* we are filled, wait until a buffer gets popped or when we
|
|
* flush. */
|
|
priv->wait_status |= APP_WAITING;
|
|
g_cond_wait (&priv->cond, &priv->mutex);
|
|
priv->wait_status &= ~APP_WAITING;
|
|
} else {
|
|
/* no need to wait for free space, we just pump more data into the
|
|
* queue hoping that the caller reacts to the enough-data signal and
|
|
* stops pushing buffers. */
|
|
break;
|
|
}
|
|
} else
|
|
break;
|
|
}
|
|
|
|
if (priv->pending_custom_segment) {
|
|
GstEvent *event = gst_event_new_segment (&priv->last_segment);
|
|
|
|
GST_DEBUG_OBJECT (appsrc, "enqueue new segment %" GST_PTR_FORMAT, event);
|
|
gst_queue_array_push_tail (priv->queue, event);
|
|
priv->pending_custom_segment = FALSE;
|
|
}
|
|
|
|
if (buflist != NULL) {
|
|
GST_DEBUG_OBJECT (appsrc, "queueing buffer list %p", buflist);
|
|
if (!steal_ref)
|
|
gst_buffer_list_ref (buflist);
|
|
gst_queue_array_push_tail (priv->queue, buflist);
|
|
priv->queued_bytes += gst_buffer_list_calculate_size (buflist);
|
|
} else {
|
|
GST_DEBUG_OBJECT (appsrc, "queueing buffer %p", buffer);
|
|
if (!steal_ref)
|
|
gst_buffer_ref (buffer);
|
|
gst_queue_array_push_tail (priv->queue, buffer);
|
|
priv->queued_bytes += gst_buffer_get_size (buffer);
|
|
}
|
|
|
|
if ((priv->wait_status & STREAM_WAITING))
|
|
g_cond_broadcast (&priv->cond);
|
|
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
GST_DEBUG_OBJECT (appsrc, "refuse buffer %p, we are flushing", buffer);
|
|
if (steal_ref)
|
|
gst_buffer_unref (buffer);
|
|
g_mutex_unlock (&priv->mutex);
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
eos:
|
|
{
|
|
GST_DEBUG_OBJECT (appsrc, "refuse buffer %p, we are EOS", buffer);
|
|
if (steal_ref)
|
|
gst_buffer_unref (buffer);
|
|
g_mutex_unlock (&priv->mutex);
|
|
return GST_FLOW_EOS;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_app_src_push_buffer_full (GstAppSrc * appsrc, GstBuffer * buffer,
|
|
gboolean steal_ref)
|
|
{
|
|
return gst_app_src_push_internal (appsrc, buffer, NULL, steal_ref);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_app_src_push_sample_internal (GstAppSrc * appsrc, GstSample * sample)
|
|
{
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
GstBufferList *buffer_list;
|
|
GstBuffer *buffer;
|
|
GstCaps *caps;
|
|
|
|
g_return_val_if_fail (GST_IS_SAMPLE (sample), GST_FLOW_ERROR);
|
|
|
|
caps = gst_sample_get_caps (sample);
|
|
if (caps != NULL) {
|
|
gst_app_src_set_caps (appsrc, caps);
|
|
} else {
|
|
GST_WARNING_OBJECT (appsrc, "received sample without caps");
|
|
}
|
|
|
|
if (priv->handle_segment_change && priv->format == GST_FORMAT_TIME) {
|
|
GstSegment *segment = gst_sample_get_segment (sample);
|
|
|
|
if (segment->format != GST_FORMAT_TIME) {
|
|
GST_LOG_OBJECT (appsrc, "format %s is not supported",
|
|
gst_format_get_name (segment->format));
|
|
goto handle_buffer;
|
|
}
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
if (gst_segment_is_equal (&priv->last_segment, segment)) {
|
|
GST_LOG_OBJECT (appsrc, "segment wasn't changed");
|
|
g_mutex_unlock (&priv->mutex);
|
|
goto handle_buffer;
|
|
}
|
|
|
|
/* will be pushed to queue with next buffer/buffer-list */
|
|
gst_segment_copy_into (segment, &priv->last_segment);
|
|
priv->pending_custom_segment = TRUE;
|
|
g_mutex_unlock (&priv->mutex);
|
|
}
|
|
|
|
handle_buffer:
|
|
|
|
buffer = gst_sample_get_buffer (sample);
|
|
if (buffer != NULL)
|
|
return gst_app_src_push_buffer_full (appsrc, buffer, FALSE);
|
|
|
|
buffer_list = gst_sample_get_buffer_list (sample);
|
|
if (buffer_list != NULL)
|
|
return gst_app_src_push_internal (appsrc, NULL, buffer_list, FALSE);
|
|
|
|
GST_WARNING_OBJECT (appsrc, "received sample without buffer or buffer list");
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_push_buffer:
|
|
* @appsrc: a #GstAppSrc
|
|
* @buffer: (transfer full): a #GstBuffer to push
|
|
*
|
|
* Adds a buffer to the queue of buffers that the appsrc element will
|
|
* push to its source pad. This function takes ownership of the buffer.
|
|
*
|
|
* When the block property is TRUE, this function can block until free
|
|
* space becomes available in the queue.
|
|
*
|
|
* Returns: #GST_FLOW_OK when the buffer was successfully queued.
|
|
* #GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
|
|
* #GST_FLOW_EOS when EOS occurred.
|
|
*/
|
|
GstFlowReturn
|
|
gst_app_src_push_buffer (GstAppSrc * appsrc, GstBuffer * buffer)
|
|
{
|
|
return gst_app_src_push_buffer_full (appsrc, buffer, TRUE);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_push_buffer_list:
|
|
* @appsrc: a #GstAppSrc
|
|
* @buffer_list: (transfer full): a #GstBufferList to push
|
|
*
|
|
* Adds a buffer list to the queue of buffers and buffer lists that the
|
|
* appsrc element will push to its source pad. This function takes ownership
|
|
* of @buffer_list.
|
|
*
|
|
* When the block property is TRUE, this function can block until free
|
|
* space becomes available in the queue.
|
|
*
|
|
* Returns: #GST_FLOW_OK when the buffer list was successfully queued.
|
|
* #GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
|
|
* #GST_FLOW_EOS when EOS occurred.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
GstFlowReturn
|
|
gst_app_src_push_buffer_list (GstAppSrc * appsrc, GstBufferList * buffer_list)
|
|
{
|
|
return gst_app_src_push_internal (appsrc, NULL, buffer_list, TRUE);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_push_sample:
|
|
* @appsrc: a #GstAppSrc
|
|
* @sample: (transfer none): a #GstSample from which buffer and caps may be
|
|
* extracted
|
|
*
|
|
* Extract a buffer from the provided sample and adds it to the queue of
|
|
* buffers that the appsrc element will push to its source pad. Any
|
|
* previous caps that were set on appsrc will be replaced by the caps
|
|
* associated with the sample if not equal.
|
|
*
|
|
* This function does not take ownership of the
|
|
* sample so the sample needs to be unreffed after calling this function.
|
|
*
|
|
* When the block property is TRUE, this function can block until free
|
|
* space becomes available in the queue.
|
|
*
|
|
* Returns: #GST_FLOW_OK when the buffer was successfully queued.
|
|
* #GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
|
|
* #GST_FLOW_EOS when EOS occurred.
|
|
*
|
|
* Since: 1.6
|
|
*
|
|
*/
|
|
GstFlowReturn
|
|
gst_app_src_push_sample (GstAppSrc * appsrc, GstSample * sample)
|
|
{
|
|
return gst_app_src_push_sample_internal (appsrc, sample);
|
|
}
|
|
|
|
/* push a buffer without stealing the ref of the buffer. This is used for the
|
|
* action signal. */
|
|
static GstFlowReturn
|
|
gst_app_src_push_buffer_action (GstAppSrc * appsrc, GstBuffer * buffer)
|
|
{
|
|
return gst_app_src_push_buffer_full (appsrc, buffer, FALSE);
|
|
}
|
|
|
|
/* push a buffer list without stealing the ref of the buffer list. This is
|
|
* used for the action signal. */
|
|
static GstFlowReturn
|
|
gst_app_src_push_buffer_list_action (GstAppSrc * appsrc,
|
|
GstBufferList * buffer_list)
|
|
{
|
|
return gst_app_src_push_internal (appsrc, NULL, buffer_list, FALSE);
|
|
}
|
|
|
|
/* push a sample without stealing the ref. This is used for the
|
|
* action signal. */
|
|
static GstFlowReturn
|
|
gst_app_src_push_sample_action (GstAppSrc * appsrc, GstSample * sample)
|
|
{
|
|
return gst_app_src_push_sample_internal (appsrc, sample);
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_end_of_stream:
|
|
* @appsrc: a #GstAppSrc
|
|
*
|
|
* Indicates to the appsrc element that the last buffer queued in the
|
|
* element is the last buffer of the stream.
|
|
*
|
|
* Returns: #GST_FLOW_OK when the EOS was successfully queued.
|
|
* #GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
|
|
*/
|
|
GstFlowReturn
|
|
gst_app_src_end_of_stream (GstAppSrc * appsrc)
|
|
{
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_val_if_fail (GST_IS_APP_SRC (appsrc), GST_FLOW_ERROR);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
/* can't accept buffers when we are flushing. We can accept them when we are
|
|
* EOS although it will not do anything. */
|
|
if (priv->flushing)
|
|
goto flushing;
|
|
|
|
GST_DEBUG_OBJECT (appsrc, "sending EOS");
|
|
priv->is_eos = TRUE;
|
|
g_cond_broadcast (&priv->cond);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
return GST_FLOW_OK;
|
|
|
|
/* ERRORS */
|
|
flushing:
|
|
{
|
|
g_mutex_unlock (&priv->mutex);
|
|
GST_DEBUG_OBJECT (appsrc, "refuse EOS, we are flushing");
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* gst_app_src_set_callbacks: (skip)
|
|
* @appsrc: a #GstAppSrc
|
|
* @callbacks: the callbacks
|
|
* @user_data: a user_data argument for the callbacks
|
|
* @notify: a destroy notify function
|
|
*
|
|
* Set callbacks which will be executed when data is needed, enough data has
|
|
* been collected or when a seek should be performed.
|
|
* This is an alternative to using the signals, it has lower overhead and is thus
|
|
* less expensive, but also less flexible.
|
|
*
|
|
* If callbacks are installed, no signals will be emitted for performance
|
|
* reasons.
|
|
*
|
|
* Before 1.16.3 it was not possible to change the callbacks in a thread-safe
|
|
* way.
|
|
*/
|
|
void
|
|
gst_app_src_set_callbacks (GstAppSrc * appsrc,
|
|
GstAppSrcCallbacks * callbacks, gpointer user_data, GDestroyNotify notify)
|
|
{
|
|
Callbacks *old_callbacks, *new_callbacks = NULL;
|
|
GstAppSrcPrivate *priv;
|
|
|
|
g_return_if_fail (GST_IS_APP_SRC (appsrc));
|
|
g_return_if_fail (callbacks != NULL);
|
|
|
|
priv = appsrc->priv;
|
|
|
|
if (callbacks) {
|
|
new_callbacks = g_new0 (Callbacks, 1);
|
|
new_callbacks->callbacks = *callbacks;
|
|
new_callbacks->user_data = user_data;
|
|
new_callbacks->destroy_notify = notify;
|
|
new_callbacks->ref_count = 1;
|
|
}
|
|
|
|
g_mutex_lock (&priv->mutex);
|
|
old_callbacks = g_steal_pointer (&priv->callbacks);
|
|
priv->callbacks = g_steal_pointer (&new_callbacks);
|
|
g_mutex_unlock (&priv->mutex);
|
|
|
|
g_clear_pointer (&old_callbacks, callbacks_unref);
|
|
}
|
|
|
|
/*** GSTURIHANDLER INTERFACE *************************************************/
|
|
|
|
static GstURIType
|
|
gst_app_src_uri_get_type (GType type)
|
|
{
|
|
return GST_URI_SRC;
|
|
}
|
|
|
|
static const gchar *const *
|
|
gst_app_src_uri_get_protocols (GType type)
|
|
{
|
|
static const gchar *protocols[] = { "appsrc", NULL };
|
|
|
|
return protocols;
|
|
}
|
|
|
|
static gchar *
|
|
gst_app_src_uri_get_uri (GstURIHandler * handler)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC (handler);
|
|
|
|
return appsrc->priv->uri ? g_strdup (appsrc->priv->uri) : NULL;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
|
|
GError ** error)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC (handler);
|
|
|
|
g_free (appsrc->priv->uri);
|
|
appsrc->priv->uri = uri ? g_strdup (uri) : NULL;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_app_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
|
|
{
|
|
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
|
|
|
|
iface->get_type = gst_app_src_uri_get_type;
|
|
iface->get_protocols = gst_app_src_uri_get_protocols;
|
|
iface->get_uri = gst_app_src_uri_get_uri;
|
|
iface->set_uri = gst_app_src_uri_set_uri;
|
|
}
|
|
|
|
static gboolean
|
|
gst_app_src_event (GstBaseSrc * src, GstEvent * event)
|
|
{
|
|
GstAppSrc *appsrc = GST_APP_SRC_CAST (src);
|
|
GstAppSrcPrivate *priv = appsrc->priv;
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_FLUSH_STOP:
|
|
g_mutex_lock (&priv->mutex);
|
|
priv->is_eos = FALSE;
|
|
g_mutex_unlock (&priv->mutex);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return GST_BASE_SRC_CLASS (parent_class)->event (src, event);
|
|
}
|