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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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782 lines
23 KiB
C++
782 lines
23 KiB
C++
/* GStreamer
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* Copyright (C) 2011 David Schleef <ds@entropywave.com>
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* Copyright (C) 2014 Sebastian Dröge <sebastian@centricular.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 Street, Suite 500,
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* Boston, MA 02110-1335, USA.
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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 "gstdecklinkaudiosrc.h"
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#include "gstdecklinkvideosrc.h"
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_src_debug);
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#define GST_CAT_DEFAULT gst_decklink_audio_src_debug
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#define DEFAULT_CONNECTION (GST_DECKLINK_AUDIO_CONNECTION_AUTO)
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#define DEFAULT_BUFFER_SIZE (5)
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#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
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#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
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enum
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{
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PROP_0,
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PROP_CONNECTION,
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PROP_DEVICE_NUMBER,
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PROP_ALIGNMENT_THRESHOLD,
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PROP_DISCONT_WAIT,
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PROP_BUFFER_SIZE
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};
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static GstStaticPadTemplate sink_template = 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
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("audio/x-raw, format={S16LE,S32LE}, channels=2, rate=48000, "
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"layout=interleaved")
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);
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typedef struct
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{
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IDeckLinkAudioInputPacket *packet;
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GstClockTime capture_time;
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} CapturePacket;
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static void
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capture_packet_free (void *data)
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{
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CapturePacket *packet = (CapturePacket *) data;
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packet->packet->Release ();
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g_free (packet);
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}
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typedef struct
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{
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IDeckLinkAudioInputPacket *packet;
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IDeckLinkInput *input;
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} AudioPacket;
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static void
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audio_packet_free (void *data)
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{
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AudioPacket *packet = (AudioPacket *) data;
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packet->packet->Release ();
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packet->input->Release ();
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g_free (packet);
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}
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static void gst_decklink_audio_src_set_property (GObject * object,
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guint property_id, const GValue * value, GParamSpec * pspec);
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static void gst_decklink_audio_src_get_property (GObject * object,
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guint property_id, GValue * value, GParamSpec * pspec);
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static void gst_decklink_audio_src_finalize (GObject * object);
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static GstStateChangeReturn
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gst_decklink_audio_src_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_decklink_audio_src_set_caps (GstBaseSrc * bsrc,
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GstCaps * caps);
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static GstCaps *gst_decklink_audio_src_get_caps (GstBaseSrc * bsrc,
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GstCaps * filter);
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static gboolean gst_decklink_audio_src_unlock (GstBaseSrc * bsrc);
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static gboolean gst_decklink_audio_src_unlock_stop (GstBaseSrc * bsrc);
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static gboolean gst_decklink_audio_src_query (GstBaseSrc * bsrc,
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GstQuery * query);
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static GstFlowReturn gst_decklink_audio_src_create (GstPushSrc * psrc,
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GstBuffer ** buffer);
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static gboolean gst_decklink_audio_src_open (GstDecklinkAudioSrc * self);
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static gboolean gst_decklink_audio_src_close (GstDecklinkAudioSrc * self);
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#define parent_class gst_decklink_audio_src_parent_class
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G_DEFINE_TYPE (GstDecklinkAudioSrc, gst_decklink_audio_src, GST_TYPE_PUSH_SRC);
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static void
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gst_decklink_audio_src_class_init (GstDecklinkAudioSrcClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseSrcClass *basesrc_class = GST_BASE_SRC_CLASS (klass);
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GstPushSrcClass *pushsrc_class = GST_PUSH_SRC_CLASS (klass);
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gobject_class->set_property = gst_decklink_audio_src_set_property;
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gobject_class->get_property = gst_decklink_audio_src_get_property;
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gobject_class->finalize = gst_decklink_audio_src_finalize;
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element_class->change_state =
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GST_DEBUG_FUNCPTR (gst_decklink_audio_src_change_state);
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basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_get_caps);
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basesrc_class->set_caps = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_set_caps);
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basesrc_class->query = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_query);
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basesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_unlock);
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basesrc_class->unlock_stop =
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GST_DEBUG_FUNCPTR (gst_decklink_audio_src_unlock_stop);
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pushsrc_class->create = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_create);
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g_object_class_install_property (gobject_class, PROP_CONNECTION,
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g_param_spec_enum ("connection", "Connection",
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"Audio input connection to use",
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GST_TYPE_DECKLINK_AUDIO_CONNECTION, DEFAULT_CONNECTION,
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(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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G_PARAM_CONSTRUCT)));
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g_object_class_install_property (gobject_class, PROP_DEVICE_NUMBER,
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g_param_spec_int ("device-number", "Device number",
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"Output device instance to use", 0, G_MAXINT, 0,
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(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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G_PARAM_CONSTRUCT)));
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g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
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g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
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"Timestamp alignment threshold in nanoseconds", 0,
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G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
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(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
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g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT,
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g_param_spec_uint64 ("discont-wait", "Discont Wait",
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"Window of time in nanoseconds to wait before "
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"creating a discontinuity", 0,
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G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT,
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(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
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g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
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g_param_spec_uint ("buffer-size", "Buffer Size",
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"Size of internal buffer in number of video frames", 1,
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G_MAXINT, DEFAULT_BUFFER_SIZE,
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(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_static_metadata (element_class, "Decklink Audio Source",
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"Audio/Src", "Decklink Source", "David Schleef <ds@entropywave.com>, "
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"Sebastian Dröge <sebastian@centricular.com>");
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GST_DEBUG_CATEGORY_INIT (gst_decklink_audio_src_debug, "decklinkaudiosrc",
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0, "debug category for decklinkaudiosrc element");
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}
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static void
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gst_decklink_audio_src_init (GstDecklinkAudioSrc * self)
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{
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self->device_number = 0;
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self->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
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self->discont_wait = DEFAULT_DISCONT_WAIT;
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self->buffer_size = DEFAULT_BUFFER_SIZE;
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gst_base_src_set_live (GST_BASE_SRC (self), TRUE);
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gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME);
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g_mutex_init (&self->lock);
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g_cond_init (&self->cond);
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g_queue_init (&self->current_packets);
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}
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void
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gst_decklink_audio_src_set_property (GObject * object, guint property_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
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switch (property_id) {
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case PROP_CONNECTION:
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self->connection =
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(GstDecklinkAudioConnectionEnum) g_value_get_enum (value);
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break;
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case PROP_DEVICE_NUMBER:
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self->device_number = g_value_get_int (value);
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break;
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case PROP_ALIGNMENT_THRESHOLD:
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self->alignment_threshold = g_value_get_uint64 (value);
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break;
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case PROP_DISCONT_WAIT:
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self->discont_wait = g_value_get_uint64 (value);
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break;
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case PROP_BUFFER_SIZE:
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self->buffer_size = g_value_get_uint (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
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break;
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}
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}
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void
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gst_decklink_audio_src_get_property (GObject * object, guint property_id,
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GValue * value, GParamSpec * pspec)
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{
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GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
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switch (property_id) {
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case PROP_CONNECTION:
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g_value_set_enum (value, self->connection);
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break;
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case PROP_DEVICE_NUMBER:
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g_value_set_int (value, self->device_number);
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break;
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case PROP_ALIGNMENT_THRESHOLD:
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g_value_set_uint64 (value, self->alignment_threshold);
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break;
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case PROP_DISCONT_WAIT:
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g_value_set_uint64 (value, self->discont_wait);
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break;
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case PROP_BUFFER_SIZE:
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g_value_set_uint (value, self->buffer_size);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
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break;
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}
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}
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void
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gst_decklink_audio_src_finalize (GObject * object)
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{
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GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (object);
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g_mutex_clear (&self->lock);
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g_cond_clear (&self->cond);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_decklink_audio_src_set_caps (GstBaseSrc * bsrc, GstCaps * caps)
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{
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GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
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BMDAudioSampleType sample_depth;
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GstCaps *current_caps;
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HRESULT ret;
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BMDAudioConnection conn = (BMDAudioConnection) - 1;
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GST_DEBUG_OBJECT (self, "Setting caps %" GST_PTR_FORMAT, caps);
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if ((current_caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (bsrc)))) {
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GST_DEBUG_OBJECT (self, "Pad already has caps %" GST_PTR_FORMAT, caps);
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if (!gst_caps_is_equal (caps, current_caps)) {
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GST_ERROR_OBJECT (self, "New caps are not equal to old caps");
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gst_caps_unref (current_caps);
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return FALSE;
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} else {
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gst_caps_unref (current_caps);
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return TRUE;
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}
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}
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if (!gst_audio_info_from_caps (&self->info, caps))
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return FALSE;
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if (self->info.finfo->format == GST_AUDIO_FORMAT_S16LE) {
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sample_depth = bmdAudioSampleType16bitInteger;
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} else {
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sample_depth = bmdAudioSampleType32bitInteger;
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}
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switch (self->connection) {
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case GST_DECKLINK_AUDIO_CONNECTION_AUTO:{
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GstElement *videosrc = NULL;
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GstDecklinkConnectionEnum vconn;
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// Try to get the connection from the videosrc and try
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// to select a sensible audio connection based on that
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g_mutex_lock (&self->input->lock);
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if (self->input->videosrc)
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videosrc = GST_ELEMENT_CAST (gst_object_ref (self->input->videosrc));
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g_mutex_unlock (&self->input->lock);
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if (videosrc) {
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g_object_get (videosrc, "connection", &vconn, NULL);
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gst_object_unref (videosrc);
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switch (vconn) {
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case GST_DECKLINK_CONNECTION_SDI:
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conn = bmdAudioConnectionEmbedded;
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break;
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case GST_DECKLINK_CONNECTION_HDMI:
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conn = bmdAudioConnectionEmbedded;
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break;
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case GST_DECKLINK_CONNECTION_OPTICAL_SDI:
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conn = bmdAudioConnectionEmbedded;
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break;
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case GST_DECKLINK_CONNECTION_COMPONENT:
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conn = bmdAudioConnectionAnalog;
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break;
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case GST_DECKLINK_CONNECTION_COMPOSITE:
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conn = bmdAudioConnectionAnalog;
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break;
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case GST_DECKLINK_CONNECTION_SVIDEO:
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conn = bmdAudioConnectionAnalog;
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break;
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default:
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// Use default
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break;
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}
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}
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break;
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}
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case GST_DECKLINK_AUDIO_CONNECTION_EMBEDDED:
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conn = bmdAudioConnectionEmbedded;
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break;
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case GST_DECKLINK_AUDIO_CONNECTION_AES_EBU:
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conn = bmdAudioConnectionAESEBU;
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break;
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case GST_DECKLINK_AUDIO_CONNECTION_ANALOG:
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conn = bmdAudioConnectionAnalog;
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break;
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case GST_DECKLINK_AUDIO_CONNECTION_ANALOG_XLR:
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conn = bmdAudioConnectionAnalogXLR;
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break;
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case GST_DECKLINK_AUDIO_CONNECTION_ANALOG_RCA:
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conn = bmdAudioConnectionAnalogRCA;
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break;
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default:
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g_assert_not_reached ();
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break;
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}
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if (conn != (BMDAudioConnection) - 1) {
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ret =
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self->input->config->SetInt (bmdDeckLinkConfigAudioInputConnection,
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conn);
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if (ret != S_OK) {
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GST_ERROR ("set configuration (audio input connection)");
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return FALSE;
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}
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}
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ret = self->input->input->EnableAudioInput (bmdAudioSampleRate48kHz,
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sample_depth, 2);
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if (ret != S_OK) {
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GST_WARNING_OBJECT (self, "Failed to enable audio input");
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return FALSE;
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}
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g_mutex_lock (&self->input->lock);
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self->input->audio_enabled = TRUE;
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if (self->input->start_streams && self->input->videosrc)
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self->input->start_streams (self->input->videosrc);
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g_mutex_unlock (&self->input->lock);
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return TRUE;
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}
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static GstCaps *
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gst_decklink_audio_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
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{
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GstCaps *caps;
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// We don't support renegotiation
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caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (bsrc));
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if (!caps)
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caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (bsrc));
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if (filter) {
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GstCaps *tmp =
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gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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caps = tmp;
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}
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return caps;
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}
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static void
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gst_decklink_audio_src_got_packet (GstElement * element,
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IDeckLinkAudioInputPacket * packet, GstClockTime capture_time)
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{
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GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
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GstDecklinkVideoSrc *videosrc = NULL;
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GST_LOG_OBJECT (self, "Got audio packet at %" GST_TIME_FORMAT,
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GST_TIME_ARGS (capture_time));
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g_mutex_lock (&self->input->lock);
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if (self->input->videosrc)
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videosrc =
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GST_DECKLINK_VIDEO_SRC_CAST (gst_object_ref (self->input->videosrc));
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g_mutex_unlock (&self->input->lock);
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if (videosrc) {
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gst_decklink_video_src_convert_to_external_clock (videosrc, &capture_time,
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NULL);
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gst_object_unref (videosrc);
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}
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g_mutex_lock (&self->lock);
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if (!self->flushing) {
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CapturePacket *p;
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while (g_queue_get_length (&self->current_packets) >= self->buffer_size) {
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p = (CapturePacket *) g_queue_pop_head (&self->current_packets);
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GST_WARNING_OBJECT (self, "Dropping old packet at %" GST_TIME_FORMAT,
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GST_TIME_ARGS (p->capture_time));
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capture_packet_free (p);
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}
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p = (CapturePacket *) g_malloc0 (sizeof (CapturePacket));
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p->packet = packet;
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p->capture_time = capture_time;
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packet->AddRef ();
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g_queue_push_tail (&self->current_packets, p);
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g_cond_signal (&self->cond);
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}
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g_mutex_unlock (&self->lock);
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}
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static GstFlowReturn
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gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
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{
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GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
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GstFlowReturn flow_ret = GST_FLOW_OK;
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const guint8 *data;
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glong sample_count;
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gsize data_size;
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CapturePacket *p;
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AudioPacket *ap;
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GstClockTime timestamp, duration;
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GstClockTime start_time, end_time;
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guint64 start_offset, end_offset;
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gboolean discont = FALSE;
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g_mutex_lock (&self->lock);
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while (g_queue_is_empty (&self->current_packets) && !self->flushing) {
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g_cond_wait (&self->cond, &self->lock);
|
|
}
|
|
|
|
p = (CapturePacket *) g_queue_pop_head (&self->current_packets);
|
|
g_mutex_unlock (&self->lock);
|
|
|
|
if (self->flushing) {
|
|
if (p)
|
|
capture_packet_free (p);
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
|
|
p->packet->GetBytes ((gpointer *) & data);
|
|
sample_count = p->packet->GetSampleFrameCount ();
|
|
data_size = self->info.bpf * sample_count;
|
|
|
|
ap = (AudioPacket *) g_malloc0 (sizeof (AudioPacket));
|
|
|
|
*buffer =
|
|
gst_buffer_new_wrapped_full ((GstMemoryFlags) GST_MEMORY_FLAG_READONLY,
|
|
(gpointer) data, data_size, 0, data_size, ap,
|
|
(GDestroyNotify) audio_packet_free);
|
|
|
|
ap->packet = p->packet;
|
|
p->packet->AddRef ();
|
|
ap->input = self->input->input;
|
|
ap->input->AddRef ();
|
|
|
|
duration =
|
|
gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate);
|
|
// Our capture time is the end timestamp, subtract the
|
|
// duration to get the start timestamp
|
|
if (p->capture_time >= duration)
|
|
timestamp = p->capture_time - duration;
|
|
else
|
|
timestamp = 0;
|
|
|
|
// Jitter and discontinuity handling, based on audiobasesrc
|
|
start_time = timestamp;
|
|
end_time = p->capture_time;
|
|
|
|
// Convert to the sample numbers
|
|
end_offset = gst_util_uint64_scale (end_time, self->info.rate, GST_SECOND);
|
|
if (end_offset >= (guint64) sample_count)
|
|
start_offset = end_offset - sample_count;
|
|
else
|
|
start_offset = 0;
|
|
|
|
if (self->next_offset == (guint64) - 1) {
|
|
discont = TRUE;
|
|
} else {
|
|
guint64 diff, max_sample_diff;
|
|
|
|
// Check discont
|
|
if (start_offset <= self->next_offset)
|
|
diff = self->next_offset - start_offset;
|
|
else
|
|
diff = start_offset - self->next_offset;
|
|
|
|
max_sample_diff =
|
|
gst_util_uint64_scale_int (self->alignment_threshold, self->info.rate,
|
|
GST_SECOND);
|
|
|
|
// Discont!
|
|
if (G_UNLIKELY (diff >= max_sample_diff)) {
|
|
if (self->discont_wait > 0) {
|
|
if (self->discont_time == GST_CLOCK_TIME_NONE) {
|
|
self->discont_time = start_time;
|
|
} else if (start_time - self->discont_time >= self->discont_wait) {
|
|
discont = TRUE;
|
|
self->discont_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
} else {
|
|
discont = TRUE;
|
|
}
|
|
} else if (G_UNLIKELY (self->discont_time != GST_CLOCK_TIME_NONE)) {
|
|
// we have had a discont, but are now back on track!
|
|
self->discont_time = GST_CLOCK_TIME_NONE;
|
|
}
|
|
}
|
|
|
|
if (discont) {
|
|
// Have discont, need resync and use the capture timestamps
|
|
if (self->next_offset != (guint64) - 1)
|
|
GST_INFO_OBJECT (self, "Have discont. Expected %"
|
|
G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
|
|
self->next_offset, start_offset);
|
|
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
|
|
self->next_offset = end_offset;
|
|
} else {
|
|
// No discont, just keep counting
|
|
self->discont_time = GST_CLOCK_TIME_NONE;
|
|
timestamp =
|
|
gst_util_uint64_scale (self->next_offset, GST_SECOND, self->info.rate);
|
|
self->next_offset += sample_count;
|
|
duration =
|
|
gst_util_uint64_scale (self->next_offset, GST_SECOND,
|
|
self->info.rate) - timestamp;
|
|
}
|
|
|
|
GST_BUFFER_TIMESTAMP (*buffer) = timestamp;
|
|
GST_BUFFER_DURATION (*buffer) = duration;
|
|
|
|
capture_packet_free (p);
|
|
|
|
return flow_ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_decklink_audio_src_query (GstBaseSrc * bsrc, GstQuery * query)
|
|
{
|
|
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
|
|
gboolean ret = TRUE;
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:{
|
|
if (self->input) {
|
|
g_mutex_lock (&self->input->lock);
|
|
if (self->input->mode) {
|
|
GstClockTime min, max;
|
|
|
|
min =
|
|
gst_util_uint64_scale_ceil (GST_SECOND, self->input->mode->fps_d,
|
|
self->input->mode->fps_n);
|
|
max = self->buffer_size * min;
|
|
|
|
gst_query_set_latency (query, TRUE, min, max);
|
|
ret = TRUE;
|
|
} else {
|
|
ret = FALSE;
|
|
}
|
|
g_mutex_unlock (&self->input->lock);
|
|
} else {
|
|
ret = FALSE;
|
|
}
|
|
|
|
break;
|
|
}
|
|
default:
|
|
ret = GST_BASE_SRC_CLASS (parent_class)->query (bsrc, query);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_decklink_audio_src_unlock (GstBaseSrc * bsrc)
|
|
{
|
|
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
|
|
|
|
g_mutex_lock (&self->lock);
|
|
self->flushing = TRUE;
|
|
g_cond_signal (&self->cond);
|
|
g_mutex_unlock (&self->lock);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_decklink_audio_src_unlock_stop (GstBaseSrc * bsrc)
|
|
{
|
|
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
|
|
|
|
g_mutex_lock (&self->lock);
|
|
self->flushing = FALSE;
|
|
g_queue_foreach (&self->current_packets, (GFunc) capture_packet_free, NULL);
|
|
g_queue_clear (&self->current_packets);
|
|
g_mutex_unlock (&self->lock);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_decklink_audio_src_open (GstDecklinkAudioSrc * self)
|
|
{
|
|
GST_DEBUG_OBJECT (self, "Starting");
|
|
|
|
self->input =
|
|
gst_decklink_acquire_nth_input (self->device_number,
|
|
GST_ELEMENT_CAST (self), TRUE);
|
|
if (!self->input) {
|
|
GST_ERROR_OBJECT (self, "Failed to acquire input");
|
|
return FALSE;
|
|
}
|
|
|
|
g_mutex_lock (&self->input->lock);
|
|
self->input->got_audio_packet = gst_decklink_audio_src_got_packet;
|
|
g_mutex_unlock (&self->input->lock);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_decklink_audio_src_close (GstDecklinkAudioSrc * self)
|
|
{
|
|
|
|
GST_DEBUG_OBJECT (self, "Stopping");
|
|
|
|
if (self->input) {
|
|
g_mutex_lock (&self->input->lock);
|
|
self->input->got_audio_packet = NULL;
|
|
self->input->audio_enabled = FALSE;
|
|
if (self->input->start_streams && self->input->videosrc)
|
|
self->input->start_streams (self->input->videosrc);
|
|
g_mutex_unlock (&self->input->lock);
|
|
|
|
self->input->input->DisableAudioInput ();
|
|
gst_decklink_release_nth_input (self->device_number,
|
|
GST_ELEMENT_CAST (self), TRUE);
|
|
self->input = NULL;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
#if 0
|
|
static gboolean
|
|
in_same_pipeline (GstElement * a, GstElement * b)
|
|
{
|
|
GstObject *root = NULL, *tmp;
|
|
gboolean ret = FALSE;
|
|
|
|
tmp = gst_object_get_parent (GST_OBJECT_CAST (a));
|
|
while (tmp != NULL) {
|
|
if (root)
|
|
gst_object_unref (root);
|
|
root = tmp;
|
|
tmp = gst_object_get_parent (root);
|
|
}
|
|
|
|
ret = root && gst_object_has_ancestor (GST_OBJECT_CAST (b), root);
|
|
|
|
if (root)
|
|
gst_object_unref (root);
|
|
|
|
return ret;
|
|
}
|
|
#endif
|
|
|
|
static GstStateChangeReturn
|
|
gst_decklink_audio_src_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
|
|
GstStateChangeReturn ret;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (!gst_decklink_audio_src_open (self)) {
|
|
ret = GST_STATE_CHANGE_FAILURE;
|
|
goto out;
|
|
}
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:{
|
|
GstElement *videosrc = NULL;
|
|
|
|
// Check if there is a video src for this input too and if it
|
|
// is actually in the same pipeline
|
|
g_mutex_lock (&self->input->lock);
|
|
if (self->input->videosrc)
|
|
videosrc = GST_ELEMENT_CAST (gst_object_ref (self->input->videosrc));
|
|
g_mutex_unlock (&self->input->lock);
|
|
|
|
if (!videosrc) {
|
|
GST_ELEMENT_ERROR (self, STREAM, FAILED,
|
|
(NULL), ("Audio src needs a video src for its operation"));
|
|
ret = GST_STATE_CHANGE_FAILURE;
|
|
goto out;
|
|
}
|
|
// FIXME: This causes deadlocks sometimes
|
|
#if 0
|
|
else if (!in_same_pipeline (GST_ELEMENT_CAST (self), videosrc)) {
|
|
GST_ELEMENT_ERROR (self, STREAM, FAILED,
|
|
(NULL),
|
|
("Audio src and video src need to be in the same pipeline"));
|
|
ret = GST_STATE_CHANGE_FAILURE;
|
|
gst_object_unref (videosrc);
|
|
goto out;
|
|
}
|
|
#endif
|
|
|
|
if (videosrc)
|
|
gst_object_unref (videosrc);
|
|
|
|
self->flushing = FALSE;
|
|
self->next_offset = -1;
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
return ret;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
g_queue_foreach (&self->current_packets, (GFunc) capture_packet_free,
|
|
NULL);
|
|
g_queue_clear (&self->current_packets);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_decklink_audio_src_close (self);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
out:
|
|
|
|
return ret;
|
|
}
|