mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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1116 lines
35 KiB
C++
1116 lines
35 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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/**
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* SECTION:element-decklinkaudiosrc
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* @short_description: Inputs Audio from a BlackMagic DeckLink Device
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* @see_also: decklinkvideosrc
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*
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* Capture Video and Audio from a BlackMagic DeckLink Device. Can only be used
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* in conjunction with decklinkvideosink.
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*
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* ## Sample pipeline
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* |[
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* gst-launch-1.0 \
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* decklinkvideosrc device-number=0 mode=1080p25 ! autovideosink \
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* decklinkaudiosrc device-number=0 ! autoaudiosink
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* ]|
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* Capturing 1080p25 video and audio from the SDI-In of Card 0. Devices are numbered
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* starting with 0.
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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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#define DEFAULT_CHANNELS (GST_DECKLINK_AUDIO_CHANNELS_2)
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#ifndef ABSDIFF
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#define ABSDIFF(x, y) ( (x) > (y) ? ((x) - (y)) : ((y) - (x)) )
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#endif
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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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PROP_CHANNELS,
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PROP_HW_SERIAL_NUMBER
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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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"audio/x-raw, format={S16LE,S32LE}, channels={8,16}, channel-mask=(bitmask)0, 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 timestamp;
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GstClockTime stream_timestamp;
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GstClockTime stream_duration;
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GstClockTime hardware_timestamp;
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GstClockTime hardware_duration;
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gboolean no_signal;
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} CapturePacket;
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static void
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capture_packet_clear (CapturePacket * packet)
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{
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packet->packet->Release ();
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memset (packet, 0, sizeof (*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_unlock (GstBaseSrc * bsrc);
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static gboolean gst_decklink_audio_src_unlock_stop (GstBaseSrc * bsrc);
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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_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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static gboolean gst_decklink_audio_src_stop (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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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (decklinkaudiosrc, "decklinkaudiosrc", GST_RANK_NONE,
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GST_TYPE_DECKLINK_AUDIO_SRC, decklink_element_init (plugin));
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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->query = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_query);
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basesrc_class->negotiate = NULL;
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basesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_decklink_audio_src_get_caps);
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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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g_object_class_install_property (gobject_class, PROP_CHANNELS,
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g_param_spec_enum ("channels", "Channels",
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"Audio channels",
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GST_TYPE_DECKLINK_AUDIO_CHANNELS, DEFAULT_CHANNELS,
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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_HW_SERIAL_NUMBER,
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g_param_spec_string ("hw-serial-number", "Hardware serial number",
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"The serial number (hardware ID) of the Decklink card",
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NULL, (GParamFlags) (G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)));
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gst_element_class_add_static_pad_template (element_class, &sink_template);
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gst_element_class_set_static_metadata (element_class, "Decklink Audio Source",
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"Audio/Source/Hardware", "Decklink Source",
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"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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self->channels = DEFAULT_CHANNELS;
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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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gst_pad_use_fixed_caps (GST_BASE_SRC_PAD (self));
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g_mutex_init (&self->lock);
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g_cond_init (&self->cond);
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self->current_packets =
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gst_queue_array_new_for_struct (sizeof (CapturePacket),
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DEFAULT_BUFFER_SIZE);
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self->skipped_last = 0;
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self->skip_from_timestamp = GST_CLOCK_TIME_NONE;
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self->skip_to_timestamp = GST_CLOCK_TIME_NONE;
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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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case PROP_CHANNELS:
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self->channels = (GstDecklinkAudioChannelsEnum) g_value_get_enum (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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case PROP_CHANNELS:
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g_value_set_enum (value, self->channels);
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break;
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case PROP_HW_SERIAL_NUMBER:
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if (self->input)
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g_value_set_string (value, self->input->hw_serial_number);
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else
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g_value_set_string (value, NULL);
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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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if (self->current_packets) {
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while (gst_queue_array_get_length (self->current_packets) > 0) {
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CapturePacket *tmp = (CapturePacket *)
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gst_queue_array_pop_head_struct (self->current_packets);
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capture_packet_clear (tmp);
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}
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gst_queue_array_free (self->current_packets);
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self->current_packets = NULL;
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}
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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_start (GstDecklinkAudioSrc * self)
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{
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BMDAudioSampleType sample_depth;
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HRESULT ret;
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BMDAudioConnection conn = (BMDAudioConnection) - 1;
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GstCaps *allowed_caps, *caps;
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g_mutex_lock (&self->input->lock);
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if (self->input->audio_enabled) {
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g_mutex_unlock (&self->input->lock);
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return TRUE;
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}
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g_mutex_unlock (&self->input->lock);
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/* Negotiate the format / sample depth with downstream */
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allowed_caps = gst_pad_get_allowed_caps (GST_BASE_SRC_PAD (self));
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if (!allowed_caps)
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allowed_caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (self));
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sample_depth = bmdAudioSampleType32bitInteger;
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if (!gst_caps_is_empty (allowed_caps)) {
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GstStructure *s;
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allowed_caps = gst_caps_simplify (allowed_caps);
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s = gst_caps_get_structure (allowed_caps, 0);
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/* If it's not a string then both formats are supported */
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if (gst_structure_has_field_typed (s, "format", G_TYPE_STRING)) {
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const gchar *format = gst_structure_get_string (s, "format");
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if (g_str_equal (format, "S16LE")) {
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sample_depth = bmdAudioSampleType16bitInteger;
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}
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}
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}
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gst_caps_unref (allowed_caps);
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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): 0x%08lx",
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(unsigned long) ret);
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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, self->channels_found);
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if (ret != S_OK) {
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GST_WARNING_OBJECT (self, "Failed to enable audio input: 0x%08lx",
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(unsigned long) ret);
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return FALSE;
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}
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gst_audio_info_set_format (&self->info,
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sample_depth ==
|
|
bmdAudioSampleType16bitInteger ? GST_AUDIO_FORMAT_S16LE :
|
|
GST_AUDIO_FORMAT_S32LE, 48000, self->channels_found, NULL);
|
|
|
|
g_mutex_lock (&self->input->lock);
|
|
self->input->audio_enabled = TRUE;
|
|
if (self->input->start_streams && self->input->videosrc)
|
|
self->input->start_streams (self->input->videosrc);
|
|
g_mutex_unlock (&self->input->lock);
|
|
|
|
caps = gst_audio_info_to_caps (&self->info);
|
|
if (!gst_base_src_set_caps (GST_BASE_SRC (self), caps)) {
|
|
gst_caps_unref (caps);
|
|
GST_WARNING_OBJECT (self, "Failed to set caps");
|
|
return FALSE;
|
|
}
|
|
gst_caps_unref (caps);
|
|
|
|
self->skipped_last = 0;
|
|
self->skip_from_timestamp = GST_CLOCK_TIME_NONE;
|
|
self->skip_to_timestamp = GST_CLOCK_TIME_NONE;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_decklink_audio_src_got_packet (GstElement * element,
|
|
IDeckLinkAudioInputPacket * packet, GstClockTime capture_time,
|
|
GstClockTime stream_time, GstClockTime stream_duration,
|
|
GstClockTime hardware_time, GstClockTime hardware_duration,
|
|
gboolean no_signal)
|
|
{
|
|
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
|
|
GstClockTime timestamp = GST_CLOCK_TIME_NONE;
|
|
|
|
GST_LOG_OBJECT (self,
|
|
"Got audio packet at %" GST_TIME_FORMAT " / %" GST_TIME_FORMAT
|
|
", no signal %d", GST_TIME_ARGS (capture_time),
|
|
GST_TIME_ARGS (stream_time), no_signal);
|
|
|
|
g_mutex_lock (&self->input->lock);
|
|
if (self->input->videosrc) {
|
|
GstDecklinkVideoSrc *videosrc =
|
|
GST_DECKLINK_VIDEO_SRC_CAST (gst_object_ref (self->input->videosrc));
|
|
|
|
if (videosrc->drop_no_signal_frames && no_signal) {
|
|
g_mutex_unlock (&self->input->lock);
|
|
return;
|
|
}
|
|
|
|
if (videosrc->first_time == GST_CLOCK_TIME_NONE)
|
|
videosrc->first_time = stream_time;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (videosrc->first_time) &&
|
|
GST_CLOCK_TIME_IS_VALID (stream_time) &&
|
|
videosrc->skip_first_time > 0 &&
|
|
stream_time - videosrc->first_time < videosrc->skip_first_time) {
|
|
GST_DEBUG_OBJECT (self,
|
|
"Skipping frame as requested: %" GST_TIME_FORMAT " < %"
|
|
GST_TIME_FORMAT, GST_TIME_ARGS (stream_time),
|
|
GST_TIME_ARGS (videosrc->skip_first_time + videosrc->first_time));
|
|
g_mutex_unlock (&self->input->lock);
|
|
return;
|
|
}
|
|
|
|
if (videosrc->output_stream_time)
|
|
timestamp = stream_time;
|
|
else if (GST_CLOCK_TIME_IS_VALID (stream_time))
|
|
timestamp = gst_clock_adjust_with_calibration (NULL, stream_time,
|
|
videosrc->current_time_mapping.xbase,
|
|
videosrc->current_time_mapping.b, videosrc->current_time_mapping.num,
|
|
videosrc->current_time_mapping.den);
|
|
} else {
|
|
timestamp = capture_time;
|
|
}
|
|
g_mutex_unlock (&self->input->lock);
|
|
|
|
GST_LOG_OBJECT (self, "Converted times to %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (timestamp));
|
|
|
|
g_mutex_lock (&self->lock);
|
|
if (!self->flushing) {
|
|
CapturePacket p;
|
|
guint skipped_packets = 0;
|
|
|
|
while (gst_queue_array_get_length (self->current_packets) >=
|
|
self->buffer_size) {
|
|
CapturePacket *tmp = (CapturePacket *)
|
|
gst_queue_array_pop_head_struct (self->current_packets);
|
|
if (skipped_packets == 0 && self->skipped_last == 0)
|
|
self->skip_from_timestamp = tmp->timestamp;
|
|
skipped_packets++;
|
|
self->skip_to_timestamp = tmp->timestamp;
|
|
capture_packet_clear (tmp);
|
|
}
|
|
|
|
if (self->skipped_last == 0 && skipped_packets > 0) {
|
|
GST_WARNING_OBJECT (self, "Starting to drop audio packets");
|
|
}
|
|
|
|
if (skipped_packets == 0 && self->skipped_last > 0) {
|
|
GST_ELEMENT_WARNING_WITH_DETAILS (self,
|
|
STREAM, FAILED,
|
|
("Dropped %u old packets from %" GST_TIME_FORMAT " to %"
|
|
GST_TIME_FORMAT, self->skipped_last,
|
|
GST_TIME_ARGS (self->skip_from_timestamp),
|
|
GST_TIME_ARGS (self->skip_to_timestamp)),
|
|
(NULL),
|
|
("dropped", G_TYPE_UINT, self->skipped_last,
|
|
"from", G_TYPE_UINT64, self->skip_from_timestamp,
|
|
"to", G_TYPE_UINT64, self->skip_to_timestamp, NULL));
|
|
self->skipped_last = 0;
|
|
}
|
|
self->skipped_last += skipped_packets;
|
|
|
|
memset (&p, 0, sizeof (p));
|
|
p.packet = packet;
|
|
p.timestamp = timestamp;
|
|
p.stream_timestamp = stream_time;
|
|
p.stream_duration = stream_duration;
|
|
p.hardware_timestamp = hardware_time;
|
|
p.hardware_duration = hardware_duration;
|
|
p.no_signal = no_signal;
|
|
packet->AddRef ();
|
|
gst_queue_array_push_tail_struct (self->current_packets, &p);
|
|
g_cond_signal (&self->cond);
|
|
}
|
|
g_mutex_unlock (&self->lock);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
|
|
{
|
|
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
|
|
GstFlowReturn flow_ret = GST_FLOW_OK;
|
|
const guint8 *data;
|
|
glong sample_count;
|
|
gsize data_size;
|
|
CapturePacket p;
|
|
AudioPacket *ap;
|
|
GstClockTime timestamp, duration;
|
|
GstClockTime start_time, end_time;
|
|
guint64 start_offset, end_offset;
|
|
gboolean discont = FALSE;
|
|
static GstStaticCaps stream_reference =
|
|
GST_STATIC_CAPS ("timestamp/x-decklink-stream");
|
|
static GstStaticCaps hardware_reference =
|
|
GST_STATIC_CAPS ("timestamp/x-decklink-hardware");
|
|
|
|
if (!gst_decklink_audio_src_start (self)) {
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
}
|
|
|
|
retry:
|
|
g_mutex_lock (&self->lock);
|
|
while (gst_queue_array_is_empty (self->current_packets) && !self->flushing) {
|
|
g_cond_wait (&self->cond, &self->lock);
|
|
}
|
|
|
|
if (self->flushing) {
|
|
GST_DEBUG_OBJECT (self, "Flushing");
|
|
g_mutex_unlock (&self->lock);
|
|
return GST_FLOW_FLUSHING;
|
|
}
|
|
|
|
p = *(CapturePacket *)
|
|
gst_queue_array_pop_head_struct (self->current_packets);
|
|
g_mutex_unlock (&self->lock);
|
|
|
|
p.packet->GetBytes ((gpointer *) & data);
|
|
sample_count = p.packet->GetSampleFrameCount ();
|
|
data_size = self->info.bpf * sample_count;
|
|
|
|
timestamp = p.timestamp;
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
if (self->next_offset == (guint64) - 1) {
|
|
GST_DEBUG_OBJECT (self,
|
|
"Got packet without timestamp before initial "
|
|
"timestamp after discont - dropping");
|
|
capture_packet_clear (&p);
|
|
goto retry;
|
|
} else {
|
|
GST_INFO_OBJECT (self, "Unknown timestamp value");
|
|
|
|
/* Likely the case where IDeckLinkInputCallback::VideoInputFrameArrived()
|
|
* didn't provide video frame, so no reference video stream timestamp
|
|
* is available. It can happen as per SDK documentation
|
|
* under the following circumstances:
|
|
* - On Intensity Pro with progressive NTSC only, every video frame will
|
|
* have two audio packets.
|
|
* - With 3:2 pulldown there are five audio packets for each four
|
|
* video frames.
|
|
* - If video processing is not fast enough, audio will still be delivered
|
|
*
|
|
* Assume there was no packet drop from previous valid packet, and use
|
|
* previously calculated expected timestamp here */
|
|
timestamp = gst_util_uint64_scale (self->next_offset,
|
|
GST_SECOND, self->info.rate);
|
|
}
|
|
}
|
|
|
|
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 ();
|
|
|
|
// Jitter and discontinuity handling, based on audiobasesrc
|
|
start_time = timestamp;
|
|
|
|
// Convert to the sample numbers
|
|
start_offset =
|
|
gst_util_uint64_scale (start_time, self->info.rate, GST_SECOND);
|
|
// Convert back to round down to a sample multiple and get rid of rounding errors
|
|
start_time = gst_util_uint64_scale (start_offset, GST_SECOND, self->info.rate);
|
|
|
|
end_offset = start_offset + sample_count;
|
|
end_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
|
|
self->info.rate);
|
|
|
|
duration = end_time - start_time;
|
|
|
|
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 (self->alignment_threshold > 0
|
|
&& self->alignment_threshold != GST_CLOCK_TIME_NONE
|
|
&& 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;
|
|
// Got a discont and adjusted, reset the discont_time marker.
|
|
self->discont_time = GST_CLOCK_TIME_NONE;
|
|
} else if (self->alignment_threshold == 0) {
|
|
// Don't align, just pass through timestamps
|
|
} else {
|
|
// No discont, just keep counting
|
|
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;
|
|
}
|
|
|
|
// Detect gaps in stream time
|
|
self->processed += sample_count;
|
|
if (self->expected_stream_time != GST_CLOCK_TIME_NONE
|
|
&& p.stream_timestamp == GST_CLOCK_TIME_NONE) {
|
|
/* We missed a frame. Extrapolate the timestamps */
|
|
p.stream_timestamp = self->expected_stream_time;
|
|
p.stream_duration =
|
|
gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate);
|
|
}
|
|
if (self->last_hardware_time != GST_CLOCK_TIME_NONE
|
|
&& p.hardware_timestamp == GST_CLOCK_TIME_NONE) {
|
|
/* This should always happen when the previous one also does, but let's
|
|
* have two separate checks just in case */
|
|
GstClockTime start_hw_offset, end_hw_offset;
|
|
start_hw_offset =
|
|
gst_util_uint64_scale (self->last_hardware_time, self->info.rate,
|
|
GST_SECOND);
|
|
end_hw_offset = start_hw_offset + sample_count;
|
|
p.hardware_timestamp =
|
|
gst_util_uint64_scale_int (end_hw_offset, GST_SECOND, self->info.rate);
|
|
/* Will be the same as the stream duration - reuse it */
|
|
p.hardware_duration = p.stream_duration;
|
|
}
|
|
|
|
if (p.stream_timestamp != GST_CLOCK_TIME_NONE) {
|
|
GstClockTime start_stream_time, end_stream_time;
|
|
|
|
start_stream_time = p.stream_timestamp;
|
|
|
|
start_offset =
|
|
gst_util_uint64_scale (start_stream_time, self->info.rate, GST_SECOND);
|
|
|
|
end_offset = start_offset + sample_count;
|
|
end_stream_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
|
|
self->info.rate);
|
|
|
|
/* With drop-frame we have gaps of 1 sample every now and then (rounding
|
|
* errors because of the samples-per-frame pattern which is not 100%
|
|
* accurate), and due to rounding errors in the calculations these can be
|
|
* 2>x>1 */
|
|
if (self->expected_stream_time != GST_CLOCK_TIME_NONE &&
|
|
ABSDIFF (self->expected_stream_time, p.stream_timestamp) >
|
|
gst_util_uint64_scale (2, GST_SECOND, self->info.rate)) {
|
|
GstMessage *msg;
|
|
GstClockTime running_time;
|
|
|
|
self->dropped +=
|
|
gst_util_uint64_scale (ABSDIFF (self->expected_stream_time,
|
|
p.stream_timestamp), self->info.rate, GST_SECOND);
|
|
running_time =
|
|
gst_segment_to_running_time (&GST_BASE_SRC (self)->segment,
|
|
GST_FORMAT_TIME, timestamp);
|
|
|
|
msg =
|
|
gst_message_new_qos (GST_OBJECT (self), TRUE, running_time,
|
|
p.stream_timestamp, timestamp, duration);
|
|
gst_message_set_qos_stats (msg, GST_FORMAT_DEFAULT, self->processed,
|
|
self->dropped);
|
|
gst_element_post_message (GST_ELEMENT (self), msg);
|
|
}
|
|
self->expected_stream_time = end_stream_time;
|
|
}
|
|
self->last_hardware_time = p.hardware_timestamp;
|
|
|
|
if (p.no_signal)
|
|
GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_GAP);
|
|
GST_BUFFER_TIMESTAMP (*buffer) = timestamp;
|
|
GST_BUFFER_DURATION (*buffer) = duration;
|
|
|
|
gst_buffer_add_reference_timestamp_meta (*buffer,
|
|
gst_static_caps_get (&stream_reference), p.stream_timestamp,
|
|
p.stream_duration);
|
|
gst_buffer_add_reference_timestamp_meta (*buffer,
|
|
gst_static_caps_get (&hardware_reference), p.hardware_timestamp,
|
|
p.hardware_duration);
|
|
|
|
GST_DEBUG_OBJECT (self,
|
|
"Outputting buffer %p with timestamp %" GST_TIME_FORMAT " and duration %"
|
|
GST_TIME_FORMAT, *buffer, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (*buffer)),
|
|
GST_TIME_ARGS (GST_BUFFER_DURATION (*buffer)));
|
|
|
|
capture_packet_clear (&p);
|
|
|
|
return flow_ret;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_decklink_audio_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
|
|
{
|
|
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
|
|
GstCaps *caps, *template_caps;
|
|
const GstStructure *s;
|
|
gint channels;
|
|
|
|
channels = self->channels;
|
|
if (channels == 0)
|
|
channels = self->channels_found;
|
|
|
|
template_caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (bsrc));
|
|
if (channels == 0) {
|
|
caps = template_caps;
|
|
} else {
|
|
if (channels > 2)
|
|
s = gst_caps_get_structure (template_caps, 1);
|
|
else
|
|
s = gst_caps_get_structure (template_caps, 0);
|
|
|
|
caps = gst_caps_new_full (gst_structure_copy (s), NULL);
|
|
gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
|
|
gst_caps_unref (template_caps);
|
|
}
|
|
|
|
if (filter) {
|
|
GstCaps *tmp =
|
|
gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (caps);
|
|
caps = tmp;
|
|
}
|
|
|
|
return caps;
|
|
}
|
|
|
|
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;
|
|
while (gst_queue_array_get_length (self->current_packets) > 0) {
|
|
CapturePacket *tmp = (CapturePacket *)
|
|
gst_queue_array_pop_head_struct (self->current_packets);
|
|
capture_packet_clear (tmp);
|
|
}
|
|
g_mutex_unlock (&self->lock);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_decklink_audio_src_open (GstDecklinkAudioSrc * self)
|
|
{
|
|
GST_DEBUG_OBJECT (self, "Opening");
|
|
|
|
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_object_notify (G_OBJECT (self), "hw-serial-number");
|
|
|
|
g_mutex_lock (&self->input->lock);
|
|
if (self->channels > 0) {
|
|
self->channels_found = self->channels;
|
|
} else {
|
|
if (self->input->attributes) {
|
|
int64_t channels_found;
|
|
|
|
HRESULT ret = self->input->attributes->GetInt
|
|
(BMDDeckLinkMaximumAudioChannels, &channels_found);
|
|
self->channels_found = channels_found;
|
|
|
|
/* Sometimes the card may report an invalid number of channels. In
|
|
* that case, we should (empirically) use 8. */
|
|
if (ret != S_OK ||
|
|
self->channels_found == 0 || g_enum_get_value ((GEnumClass *)
|
|
g_type_class_peek (GST_TYPE_DECKLINK_AUDIO_CHANNELS),
|
|
self->channels_found)
|
|
== NULL) {
|
|
self->channels_found = GST_DECKLINK_AUDIO_CHANNELS_8;
|
|
}
|
|
}
|
|
}
|
|
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, "Closing");
|
|
|
|
if (self->input) {
|
|
g_mutex_lock (&self->input->lock);
|
|
self->input->got_audio_packet = NULL;
|
|
g_mutex_unlock (&self->input->lock);
|
|
|
|
gst_decklink_release_nth_input (self->device_number,
|
|
GST_ELEMENT_CAST (self), TRUE);
|
|
self->input = NULL;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_decklink_audio_src_stop (GstDecklinkAudioSrc * self)
|
|
{
|
|
GST_DEBUG_OBJECT (self, "Stopping");
|
|
|
|
while (gst_queue_array_get_length (self->current_packets) > 0) {
|
|
CapturePacket *tmp = (CapturePacket *)
|
|
gst_queue_array_pop_head_struct (self->current_packets);
|
|
capture_packet_clear (tmp);
|
|
}
|
|
|
|
if (self->input && self->input->audio_enabled) {
|
|
g_mutex_lock (&self->input->lock);
|
|
self->input->audio_enabled = FALSE;
|
|
g_mutex_unlock (&self->input->lock);
|
|
|
|
self->input->input->DisableAudioInput ();
|
|
}
|
|
|
|
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:
|
|
self->processed = 0;
|
|
self->dropped = 0;
|
|
self->expected_stream_time = GST_CLOCK_TIME_NONE;
|
|
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:
|
|
gst_decklink_audio_src_stop (self);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_decklink_audio_src_close (self);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
out:
|
|
|
|
return ret;
|
|
}
|