Update
This commit is contained in:
Akkariin Meiko
@@ -0,0 +1,927 @@
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/* 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-decklinkaudiosink
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* @short_description: Outputs Audio to a BlackMagic DeckLink Device
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* @see_also: decklinkvideosink
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*
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* Playout Video and Audio to 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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* videotestsrc ! decklinkvideosink device-number=0 mode=1080p25 \
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* audiotestsrc ! decklinkaudiosink device-number=0
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* ]|
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* Playout a 1080p25 test-video with a test-audio signal to the SDI-Out of Card 0.
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* Devices are numbered 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 "gstdecklinkaudiosink.h"
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#include "gstdecklinkvideosink.h"
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (gst_decklink_audio_sink_debug);
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#define GST_CAT_DEFAULT gst_decklink_audio_sink_debug
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#define DEFAULT_DEVICE_NUMBER (0)
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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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// Microseconds for audiobasesink compatibility...
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#define DEFAULT_BUFFER_TIME (50 * GST_MSECOND / 1000)
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enum
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{
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PROP_0,
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PROP_DEVICE_NUMBER,
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PROP_HW_SERIAL_NUMBER,
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PROP_ALIGNMENT_THRESHOLD,
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PROP_DISCONT_WAIT,
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PROP_BUFFER_TIME,
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};
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static void gst_decklink_audio_sink_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_sink_get_property (GObject * object,
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guint property_id, GValue * value, GParamSpec * pspec);
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static void gst_decklink_audio_sink_finalize (GObject * object);
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static GstStateChangeReturn
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gst_decklink_audio_sink_change_state (GstElement * element,
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GstStateChange transition);
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static GstClock *gst_decklink_audio_sink_provide_clock (GstElement * element);
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static GstCaps *gst_decklink_audio_sink_get_caps (GstBaseSink * bsink,
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GstCaps * filter);
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static gboolean gst_decklink_audio_sink_set_caps (GstBaseSink * bsink,
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GstCaps * caps);
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static GstFlowReturn gst_decklink_audio_sink_render (GstBaseSink * bsink,
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GstBuffer * buffer);
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static gboolean gst_decklink_audio_sink_open (GstBaseSink * bsink);
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static gboolean gst_decklink_audio_sink_close (GstBaseSink * bsink);
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static gboolean gst_decklink_audio_sink_stop (GstDecklinkAudioSink * self);
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static gboolean gst_decklink_audio_sink_unlock_stop (GstBaseSink * bsink);
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static void gst_decklink_audio_sink_get_times (GstBaseSink * bsink,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static gboolean gst_decklink_audio_sink_query (GstBaseSink * bsink,
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GstQuery * query);
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static gboolean gst_decklink_audio_sink_event (GstBaseSink * bsink,
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GstEvent * event);
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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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, 8, 16}, rate=48000, "
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"layout=interleaved")
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);
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#define parent_class gst_decklink_audio_sink_parent_class
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G_DEFINE_TYPE (GstDecklinkAudioSink, gst_decklink_audio_sink,
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GST_TYPE_BASE_SINK);
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GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (decklinkaudiosink, "decklinkaudiosink", GST_RANK_NONE,
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GST_TYPE_DECKLINK_AUDIO_SINK, decklink_element_init (plugin));
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static void
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gst_decklink_audio_sink_class_init (GstDecklinkAudioSinkClass * 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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GstBaseSinkClass *basesink_class = GST_BASE_SINK_CLASS (klass);
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gobject_class->set_property = gst_decklink_audio_sink_set_property;
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gobject_class->get_property = gst_decklink_audio_sink_get_property;
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gobject_class->finalize = gst_decklink_audio_sink_finalize;
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element_class->change_state =
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GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_change_state);
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element_class->provide_clock =
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GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_provide_clock);
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basesink_class->get_caps =
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GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_get_caps);
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basesink_class->set_caps =
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GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_set_caps);
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basesink_class->render = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_render);
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// FIXME: These are misnamed in basesink!
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basesink_class->start = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_open);
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basesink_class->stop = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_close);
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basesink_class->unlock_stop =
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GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_unlock_stop);
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basesink_class->get_times =
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GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_get_times);
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basesink_class->query = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_query);
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basesink_class->event = GST_DEBUG_FUNCPTR (gst_decklink_audio_sink_event);
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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, DEFAULT_DEVICE_NUMBER,
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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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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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GST_PARAM_MUTABLE_READY)));
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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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GST_PARAM_MUTABLE_READY)));
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g_object_class_install_property (gobject_class, PROP_BUFFER_TIME,
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g_param_spec_uint64 ("buffer-time", "Buffer Time",
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"Size of audio buffer in microseconds, this is the minimum latency that the sink reports",
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0, G_MAXUINT64, DEFAULT_BUFFER_TIME,
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(GParamFlags) (G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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GST_PARAM_MUTABLE_READY)));
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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 Sink",
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"Audio/Sink/Hardware", "Decklink Sink",
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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_sink_debug, "decklinkaudiosink",
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0, "debug category for decklinkaudiosink element");
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}
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static void
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gst_decklink_audio_sink_init (GstDecklinkAudioSink * self)
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{
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self->device_number = DEFAULT_DEVICE_NUMBER;
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self->stream_align =
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gst_audio_stream_align_new (48000, DEFAULT_ALIGNMENT_THRESHOLD,
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DEFAULT_DISCONT_WAIT);
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self->buffer_time = DEFAULT_BUFFER_TIME * 1000;
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gst_base_sink_set_max_lateness (GST_BASE_SINK_CAST (self), 20 * GST_MSECOND);
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}
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void
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gst_decklink_audio_sink_set_property (GObject * object, guint property_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object);
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switch (property_id) {
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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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GST_OBJECT_LOCK (self);
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gst_audio_stream_align_set_alignment_threshold (self->stream_align,
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g_value_get_uint64 (value));
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GST_OBJECT_UNLOCK (self);
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break;
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case PROP_DISCONT_WAIT:
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GST_OBJECT_LOCK (self);
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gst_audio_stream_align_set_discont_wait (self->stream_align,
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g_value_get_uint64 (value));
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GST_OBJECT_UNLOCK (self);
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break;
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case PROP_BUFFER_TIME:
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GST_OBJECT_LOCK (self);
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self->buffer_time = g_value_get_uint64 (value) * 1000;
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GST_OBJECT_UNLOCK (self);
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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_sink_get_property (GObject * object, guint property_id,
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GValue * value, GParamSpec * pspec)
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{
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GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object);
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switch (property_id) {
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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_HW_SERIAL_NUMBER:
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if (self->output)
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g_value_set_string (value, self->output->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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case PROP_ALIGNMENT_THRESHOLD:
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GST_OBJECT_LOCK (self);
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g_value_set_uint64 (value,
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gst_audio_stream_align_get_alignment_threshold (self->stream_align));
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GST_OBJECT_UNLOCK (self);
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break;
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case PROP_DISCONT_WAIT:
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GST_OBJECT_LOCK (self);
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g_value_set_uint64 (value,
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gst_audio_stream_align_get_discont_wait (self->stream_align));
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GST_OBJECT_UNLOCK (self);
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break;
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case PROP_BUFFER_TIME:
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GST_OBJECT_LOCK (self);
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g_value_set_uint64 (value, self->buffer_time / 1000);
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GST_OBJECT_UNLOCK (self);
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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_sink_finalize (GObject * object)
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{
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GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (object);
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if (self->stream_align) {
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gst_audio_stream_align_free (self->stream_align);
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self->stream_align = 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_sink_set_caps (GstBaseSink * bsink, GstCaps * caps)
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{
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GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
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HRESULT ret;
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BMDAudioSampleType sample_depth;
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GstAudioInfo info;
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GST_DEBUG_OBJECT (self, "Setting caps %" GST_PTR_FORMAT, caps);
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if (!gst_audio_info_from_caps (&info, caps))
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return FALSE;
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if (self->output->audio_enabled
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&& (self->info.finfo->format != info.finfo->format
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|| self->info.channels != info.channels)) {
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GST_ERROR_OBJECT (self, "Reconfiguration not supported");
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return FALSE;
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} else if (self->output->audio_enabled) {
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return TRUE;
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}
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if (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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ret = self->output->output->EnableAudioOutput (bmdAudioSampleRate48kHz,
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sample_depth, info.channels, bmdAudioOutputStreamContinuous);
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if (ret != S_OK) {
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GST_WARNING_OBJECT (self, "Failed to enable audio output 0x%08lx",
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(unsigned long) ret);
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return FALSE;
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}
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self->output->audio_enabled = TRUE;
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self->info = info;
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// Create a new resampler as needed
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if (self->resampler)
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gst_audio_resampler_free (self->resampler);
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self->resampler = NULL;
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return TRUE;
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}
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static GstCaps *
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gst_decklink_audio_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
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{
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GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
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GstCaps *caps;
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if ((caps = gst_pad_get_current_caps (GST_BASE_SINK_PAD (bsink))))
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return caps;
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caps = gst_pad_get_pad_template_caps (GST_BASE_SINK_PAD (bsink));
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GST_OBJECT_LOCK (self);
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if (self->output && self->output->attributes) {
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int64_t max_channels = 0;
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HRESULT ret;
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GstStructure *s;
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GValue arr = G_VALUE_INIT;
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GValue v = G_VALUE_INIT;
|
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ret =
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self->output->attributes->GetInt (BMDDeckLinkMaximumAudioChannels,
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&max_channels);
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/* 2 should always be supported */
|
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if (ret != S_OK) {
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max_channels = 2;
|
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}
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|
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caps = gst_caps_make_writable (caps);
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s = gst_caps_get_structure (caps, 0);
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|
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g_value_init (&arr, GST_TYPE_LIST);
|
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g_value_init (&v, G_TYPE_INT);
|
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if (max_channels >= 16) {
|
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g_value_set_int (&v, 16);
|
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gst_value_list_append_value (&arr, &v);
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}
|
||||
if (max_channels >= 8) {
|
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g_value_set_int (&v, 8);
|
||||
gst_value_list_append_value (&arr, &v);
|
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}
|
||||
g_value_set_int (&v, 2);
|
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gst_value_list_append_value (&arr, &v);
|
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|
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gst_structure_set_value (s, "channels", &arr);
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g_value_unset (&v);
|
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g_value_unset (&arr);
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}
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GST_OBJECT_UNLOCK (self);
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||||
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if (filter) {
|
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GstCaps *intersection =
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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 = intersection;
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}
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return caps;
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}
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static gboolean
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gst_decklink_audio_sink_query (GstBaseSink * bsink, GstQuery * query)
|
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{
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GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK (bsink);
|
||||
gboolean res = FALSE;
|
||||
|
||||
switch (GST_QUERY_TYPE (query)) {
|
||||
case GST_QUERY_LATENCY:
|
||||
{
|
||||
gboolean live, us_live;
|
||||
GstClockTime min_l, max_l;
|
||||
|
||||
GST_DEBUG_OBJECT (self, "latency query");
|
||||
|
||||
/* ask parent first, it will do an upstream query for us. */
|
||||
if ((res =
|
||||
gst_base_sink_query_latency (GST_BASE_SINK_CAST (self), &live,
|
||||
&us_live, &min_l, &max_l))) {
|
||||
GstClockTime base_latency, min_latency, max_latency;
|
||||
|
||||
/* we and upstream are both live, adjust the min_latency */
|
||||
if (live && us_live) {
|
||||
GST_OBJECT_LOCK (self);
|
||||
if (!self->info.rate) {
|
||||
GST_OBJECT_UNLOCK (self);
|
||||
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"we are not negotiated, can't report latency yet");
|
||||
res = FALSE;
|
||||
goto done;
|
||||
}
|
||||
|
||||
base_latency = self->buffer_time * 1000;
|
||||
GST_OBJECT_UNLOCK (self);
|
||||
|
||||
/* we cannot go lower than the buffer size and the min peer latency */
|
||||
min_latency = base_latency + min_l;
|
||||
/* the max latency is the max of the peer, we can delay an infinite
|
||||
* amount of time. */
|
||||
max_latency =
|
||||
(max_l ==
|
||||
GST_CLOCK_TIME_NONE) ? GST_CLOCK_TIME_NONE : (base_latency +
|
||||
max_l);
|
||||
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"peer min %" GST_TIME_FORMAT ", our min latency: %"
|
||||
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
|
||||
GST_TIME_ARGS (min_latency));
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"peer max %" GST_TIME_FORMAT ", our max latency: %"
|
||||
GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
|
||||
GST_TIME_ARGS (max_latency));
|
||||
} else {
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"peer or we are not live, don't care about latency");
|
||||
min_latency = min_l;
|
||||
max_latency = max_l;
|
||||
}
|
||||
gst_query_set_latency (query, live, min_latency, max_latency);
|
||||
}
|
||||
break;
|
||||
}
|
||||
default:
|
||||
res = GST_BASE_SINK_CLASS (parent_class)->query (bsink, query);
|
||||
break;
|
||||
}
|
||||
|
||||
done:
|
||||
return res;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_decklink_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
|
||||
{
|
||||
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
|
||||
|
||||
if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT) {
|
||||
const GstSegment *new_segment;
|
||||
|
||||
gst_event_parse_segment (event, &new_segment);
|
||||
|
||||
if (ABS (new_segment->rate) != 1.0) {
|
||||
guint out_rate = self->info.rate / ABS (new_segment->rate);
|
||||
|
||||
if (self->resampler && (self->resampler_out_rate != out_rate
|
||||
|| self->resampler_in_rate != (guint) self->info.rate))
|
||||
gst_audio_resampler_update (self->resampler, self->info.rate, out_rate,
|
||||
NULL);
|
||||
else if (!self->resampler)
|
||||
self->resampler =
|
||||
gst_audio_resampler_new (GST_AUDIO_RESAMPLER_METHOD_LINEAR,
|
||||
GST_AUDIO_RESAMPLER_FLAG_NONE, self->info.finfo->format,
|
||||
self->info.channels, self->info.rate, out_rate, NULL);
|
||||
|
||||
self->resampler_in_rate = self->info.rate;
|
||||
self->resampler_out_rate = out_rate;
|
||||
} else if (self->resampler) {
|
||||
gst_audio_resampler_free (self->resampler);
|
||||
self->resampler = NULL;
|
||||
}
|
||||
|
||||
if (new_segment->rate < 0)
|
||||
gst_audio_stream_align_set_rate (self->stream_align, -48000);
|
||||
}
|
||||
|
||||
return GST_BASE_SINK_CLASS (parent_class)->event (bsink, event);
|
||||
}
|
||||
|
||||
static GstFlowReturn
|
||||
gst_decklink_audio_sink_render (GstBaseSink * bsink, GstBuffer * buffer)
|
||||
{
|
||||
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
|
||||
GstDecklinkVideoSink *video_sink;
|
||||
GstFlowReturn flow_ret;
|
||||
HRESULT ret;
|
||||
GstClockTime timestamp, duration;
|
||||
GstClockTime running_time, running_time_duration;
|
||||
GstClockTime schedule_time, schedule_time_duration;
|
||||
GstClockTime latency, render_delay;
|
||||
GstClockTimeDiff ts_offset;
|
||||
GstMapInfo map_info;
|
||||
const guint8 *data;
|
||||
gsize len, written_all;
|
||||
gboolean discont;
|
||||
|
||||
GST_DEBUG_OBJECT (self, "Rendering buffer %p", buffer);
|
||||
|
||||
// FIXME: Handle no timestamps
|
||||
if (!GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
|
||||
return GST_FLOW_ERROR;
|
||||
}
|
||||
|
||||
if (GST_BASE_SINK_CAST (self)->flushing) {
|
||||
return GST_FLOW_FLUSHING;
|
||||
}
|
||||
// If we're called before output is actually started, start pre-rolling
|
||||
if (!self->output->started) {
|
||||
self->output->output->BeginAudioPreroll ();
|
||||
}
|
||||
|
||||
video_sink =
|
||||
GST_DECKLINK_VIDEO_SINK (gst_object_ref (self->output->videosink));
|
||||
|
||||
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
||||
duration = GST_BUFFER_DURATION (buffer);
|
||||
discont = gst_audio_stream_align_process (self->stream_align,
|
||||
GST_BUFFER_IS_DISCONT (buffer), timestamp,
|
||||
gst_buffer_get_size (buffer) / self->info.bpf, ×tamp, &duration,
|
||||
NULL);
|
||||
|
||||
if (discont && self->resampler)
|
||||
gst_audio_resampler_reset (self->resampler);
|
||||
|
||||
if (GST_BASE_SINK_CAST (self)->segment.rate < 0.0) {
|
||||
GstMapInfo out_map;
|
||||
gint out_frames = gst_buffer_get_size (buffer) / self->info.bpf;
|
||||
|
||||
buffer = gst_buffer_make_writable (gst_buffer_ref (buffer));
|
||||
|
||||
gst_buffer_map (buffer, &out_map, GST_MAP_READWRITE);
|
||||
if (self->info.finfo->format == GST_AUDIO_FORMAT_S16) {
|
||||
gint16 *swap_data = (gint16 *) out_map.data;
|
||||
gint16 *swap_data_end =
|
||||
swap_data + (out_frames - 1) * self->info.channels;
|
||||
gint16 swap_tmp[16];
|
||||
|
||||
while (out_frames > 0) {
|
||||
memcpy (&swap_tmp, swap_data, self->info.bpf);
|
||||
memcpy (swap_data, swap_data_end, self->info.bpf);
|
||||
memcpy (swap_data_end, &swap_tmp, self->info.bpf);
|
||||
|
||||
swap_data += self->info.channels;
|
||||
swap_data_end -= self->info.channels;
|
||||
|
||||
out_frames -= 2;
|
||||
}
|
||||
} else {
|
||||
gint32 *swap_data = (gint32 *) out_map.data;
|
||||
gint32 *swap_data_end =
|
||||
swap_data + (out_frames - 1) * self->info.channels;
|
||||
gint32 swap_tmp[16];
|
||||
|
||||
while (out_frames > 0) {
|
||||
memcpy (&swap_tmp, swap_data, self->info.bpf);
|
||||
memcpy (swap_data, swap_data_end, self->info.bpf);
|
||||
memcpy (swap_data_end, &swap_tmp, self->info.bpf);
|
||||
|
||||
swap_data += self->info.channels;
|
||||
swap_data_end -= self->info.channels;
|
||||
|
||||
out_frames -= 2;
|
||||
}
|
||||
}
|
||||
gst_buffer_unmap (buffer, &out_map);
|
||||
} else {
|
||||
gst_buffer_ref (buffer);
|
||||
}
|
||||
|
||||
if (self->resampler) {
|
||||
gint in_frames = gst_buffer_get_size (buffer) / self->info.bpf;
|
||||
gint out_frames =
|
||||
gst_audio_resampler_get_out_frames (self->resampler, in_frames);
|
||||
GstBuffer *out_buf = gst_buffer_new_and_alloc (out_frames * self->info.bpf);
|
||||
GstMapInfo out_map;
|
||||
|
||||
gst_buffer_map (buffer, &map_info, GST_MAP_READ);
|
||||
gst_buffer_map (out_buf, &out_map, GST_MAP_READWRITE);
|
||||
|
||||
gst_audio_resampler_resample (self->resampler, (gpointer *) & map_info.data,
|
||||
in_frames, (gpointer *) & out_map.data, out_frames);
|
||||
|
||||
gst_buffer_unmap (out_buf, &out_map);
|
||||
gst_buffer_unmap (buffer, &map_info);
|
||||
buffer = out_buf;
|
||||
}
|
||||
|
||||
gst_buffer_map (buffer, &map_info, GST_MAP_READ);
|
||||
data = map_info.data;
|
||||
len = map_info.size / self->info.bpf;
|
||||
written_all = 0;
|
||||
|
||||
do {
|
||||
GstClockTime timestamp_now =
|
||||
timestamp + gst_util_uint64_scale (written_all, GST_SECOND,
|
||||
self->info.rate);
|
||||
guint32 buffered_samples;
|
||||
GstClockTime buffered_time;
|
||||
guint32 written = 0;
|
||||
GstClock *clock;
|
||||
GstClockTimeDiff clock_ahead;
|
||||
|
||||
if (GST_BASE_SINK_CAST (self)->flushing) {
|
||||
flow_ret = GST_FLOW_FLUSHING;
|
||||
break;
|
||||
}
|
||||
|
||||
running_time =
|
||||
gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment,
|
||||
GST_FORMAT_TIME, timestamp_now);
|
||||
running_time_duration =
|
||||
gst_segment_to_running_time (&GST_BASE_SINK_CAST (self)->segment,
|
||||
GST_FORMAT_TIME, timestamp_now + duration) - running_time;
|
||||
|
||||
/* See gst_base_sink_adjust_time() */
|
||||
latency = gst_base_sink_get_latency (bsink);
|
||||
render_delay = gst_base_sink_get_render_delay (bsink);
|
||||
ts_offset = gst_base_sink_get_ts_offset (bsink);
|
||||
running_time += latency;
|
||||
|
||||
if (ts_offset < 0) {
|
||||
ts_offset = -ts_offset;
|
||||
if ((GstClockTime) ts_offset < running_time)
|
||||
running_time -= ts_offset;
|
||||
else
|
||||
running_time = 0;
|
||||
} else {
|
||||
running_time += ts_offset;
|
||||
}
|
||||
|
||||
if (running_time > render_delay)
|
||||
running_time -= render_delay;
|
||||
else
|
||||
running_time = 0;
|
||||
|
||||
clock = gst_element_get_clock (GST_ELEMENT_CAST (self));
|
||||
clock_ahead = 0;
|
||||
if (clock) {
|
||||
GstClockTime clock_now = gst_clock_get_time (clock);
|
||||
GstClockTime base_time =
|
||||
gst_element_get_base_time (GST_ELEMENT_CAST (self));
|
||||
gst_object_unref (clock);
|
||||
clock = NULL;
|
||||
|
||||
if (clock_now != GST_CLOCK_TIME_NONE && base_time != GST_CLOCK_TIME_NONE) {
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"Clock time %" GST_TIME_FORMAT ", base time %" GST_TIME_FORMAT
|
||||
", target running time %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (clock_now), GST_TIME_ARGS (base_time),
|
||||
GST_TIME_ARGS (running_time));
|
||||
if (clock_now > base_time)
|
||||
clock_now -= base_time;
|
||||
else
|
||||
clock_now = 0;
|
||||
|
||||
clock_ahead = running_time - clock_now;
|
||||
}
|
||||
}
|
||||
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"Ahead %" GST_STIME_FORMAT " of the clock running time",
|
||||
GST_STIME_ARGS (clock_ahead));
|
||||
|
||||
if (self->output->
|
||||
output->GetBufferedAudioSampleFrameCount (&buffered_samples) != S_OK)
|
||||
buffered_samples = 0;
|
||||
|
||||
buffered_time =
|
||||
gst_util_uint64_scale (buffered_samples, GST_SECOND, self->info.rate);
|
||||
buffered_time /= ABS (GST_BASE_SINK_CAST (self)->segment.rate);
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"Buffered %" GST_TIME_FORMAT " in the driver (%u samples)",
|
||||
GST_TIME_ARGS (buffered_time), buffered_samples);
|
||||
|
||||
{
|
||||
GstClockTimeDiff buffered_ahead_of_clock_ahead =
|
||||
GST_CLOCK_DIFF (clock_ahead, buffered_time);
|
||||
|
||||
GST_DEBUG_OBJECT (self, "driver is %" GST_STIME_FORMAT " ahead of the "
|
||||
"expected clock", GST_STIME_ARGS (buffered_ahead_of_clock_ahead));
|
||||
/* we don't want to store too much data in the driver as decklink
|
||||
* doesn't seem to actually use our provided timestamps to perform its
|
||||
* own synchronisation. It seems to count samples instead. */
|
||||
/* FIXME: do we need to split buffers? */
|
||||
if (buffered_ahead_of_clock_ahead > 0 &&
|
||||
buffered_ahead_of_clock_ahead >
|
||||
gst_base_sink_get_max_lateness (bsink)) {
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"Dropping buffer that is %" GST_STIME_FORMAT " too late",
|
||||
GST_STIME_ARGS (buffered_ahead_of_clock_ahead));
|
||||
if (self->resampler)
|
||||
gst_audio_resampler_reset (self->resampler);
|
||||
flow_ret = GST_FLOW_OK;
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
// We start waiting once we have more than buffer-time buffered
|
||||
if (((GstClockTime) clock_ahead) > self->buffer_time) {
|
||||
GstClockReturn clock_ret;
|
||||
GstClockTime wait_time = running_time;
|
||||
|
||||
GST_DEBUG_OBJECT (self,
|
||||
"Buffered enough, wait for preroll or the clock or flushing. "
|
||||
"Configured buffer time: %" GST_TIME_FORMAT,
|
||||
GST_TIME_ARGS (self->buffer_time));
|
||||
|
||||
if (wait_time < self->buffer_time)
|
||||
wait_time = 0;
|
||||
else
|
||||
wait_time -= self->buffer_time;
|
||||
|
||||
flow_ret =
|
||||
gst_base_sink_do_preroll (GST_BASE_SINK_CAST (self),
|
||||
GST_MINI_OBJECT_CAST (buffer));
|
||||
if (flow_ret != GST_FLOW_OK)
|
||||
break;
|
||||
|
||||
clock_ret =
|
||||
gst_base_sink_wait_clock (GST_BASE_SINK_CAST (self), wait_time, NULL);
|
||||
if (GST_BASE_SINK_CAST (self)->flushing) {
|
||||
flow_ret = GST_FLOW_FLUSHING;
|
||||
break;
|
||||
}
|
||||
// Rerun the whole loop again
|
||||
if (clock_ret == GST_CLOCK_UNSCHEDULED)
|
||||
continue;
|
||||
}
|
||||
|
||||
schedule_time = running_time;
|
||||
schedule_time_duration = running_time_duration;
|
||||
|
||||
gst_decklink_video_sink_convert_to_internal_clock (video_sink,
|
||||
&schedule_time, &schedule_time_duration);
|
||||
|
||||
GST_LOG_OBJECT (self, "Scheduling audio samples at %" GST_TIME_FORMAT
|
||||
" with duration %" GST_TIME_FORMAT, GST_TIME_ARGS (schedule_time),
|
||||
GST_TIME_ARGS (schedule_time_duration));
|
||||
|
||||
ret = self->output->output->ScheduleAudioSamples ((void *) data, len,
|
||||
schedule_time, GST_SECOND, &written);
|
||||
if (ret != S_OK) {
|
||||
bool is_running = true;
|
||||
self->output->output->IsScheduledPlaybackRunning (&is_running);
|
||||
|
||||
if (is_running && !GST_BASE_SINK_CAST (self)->flushing
|
||||
&& self->output->started) {
|
||||
GST_ELEMENT_ERROR (self, STREAM, FAILED, (NULL),
|
||||
("Failed to schedule frame: 0x%08lx", (unsigned long) ret));
|
||||
flow_ret = GST_FLOW_ERROR;
|
||||
break;
|
||||
} else {
|
||||
// Ignore the error and go out of the loop here, we're shutting down
|
||||
// or are not started yet and there's nothing we can do at this point
|
||||
GST_INFO_OBJECT (self,
|
||||
"Ignoring scheduling error 0x%08x because we're not started yet"
|
||||
" or not anymore", (guint) ret);
|
||||
flow_ret = GST_FLOW_OK;
|
||||
break;
|
||||
}
|
||||
}
|
||||
|
||||
len -= written;
|
||||
data += written * self->info.bpf;
|
||||
if (self->resampler)
|
||||
written_all += written * ABS (GST_BASE_SINK_CAST (self)->segment.rate);
|
||||
else
|
||||
written_all += written;
|
||||
|
||||
flow_ret = GST_FLOW_OK;
|
||||
} while (len > 0);
|
||||
|
||||
gst_buffer_unmap (buffer, &map_info);
|
||||
gst_buffer_unref (buffer);
|
||||
|
||||
GST_DEBUG_OBJECT (self, "Returning %s", gst_flow_get_name (flow_ret));
|
||||
|
||||
return flow_ret;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_decklink_audio_sink_open (GstBaseSink * bsink)
|
||||
{
|
||||
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
|
||||
|
||||
GST_DEBUG_OBJECT (self, "Starting");
|
||||
|
||||
self->output =
|
||||
gst_decklink_acquire_nth_output (self->device_number,
|
||||
GST_ELEMENT_CAST (self), TRUE);
|
||||
if (!self->output) {
|
||||
GST_ERROR_OBJECT (self, "Failed to acquire output");
|
||||
return FALSE;
|
||||
}
|
||||
|
||||
g_object_notify (G_OBJECT (self), "hw-serial-number");
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_decklink_audio_sink_close (GstBaseSink * bsink)
|
||||
{
|
||||
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (bsink);
|
||||
|
||||
GST_DEBUG_OBJECT (self, "Closing");
|
||||
|
||||
if (self->output) {
|
||||
g_mutex_lock (&self->output->lock);
|
||||
self->output->mode = NULL;
|
||||
self->output->audio_enabled = FALSE;
|
||||
if (self->output->start_scheduled_playback && self->output->videosink)
|
||||
self->output->start_scheduled_playback (self->output->videosink);
|
||||
g_mutex_unlock (&self->output->lock);
|
||||
|
||||
self->output->output->DisableAudioOutput ();
|
||||
gst_decklink_release_nth_output (self->device_number,
|
||||
GST_ELEMENT_CAST (self), TRUE);
|
||||
self->output = NULL;
|
||||
}
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_decklink_audio_sink_stop (GstDecklinkAudioSink * self)
|
||||
{
|
||||
GST_DEBUG_OBJECT (self, "Stopping");
|
||||
|
||||
if (self->output && self->output->audio_enabled) {
|
||||
g_mutex_lock (&self->output->lock);
|
||||
self->output->audio_enabled = FALSE;
|
||||
g_mutex_unlock (&self->output->lock);
|
||||
|
||||
self->output->output->DisableAudioOutput ();
|
||||
}
|
||||
|
||||
if (self->resampler) {
|
||||
gst_audio_resampler_free (self->resampler);
|
||||
self->resampler = NULL;
|
||||
}
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static gboolean
|
||||
gst_decklink_audio_sink_unlock_stop (GstBaseSink * bsink)
|
||||
{
|
||||
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK (bsink);
|
||||
|
||||
if (self->output) {
|
||||
self->output->output->FlushBufferedAudioSamples ();
|
||||
}
|
||||
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_decklink_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
|
||||
GstClockTime * start, GstClockTime * end)
|
||||
{
|
||||
/* our clock sync is a bit too much for the base class to handle so
|
||||
* we implement it ourselves. */
|
||||
*start = GST_CLOCK_TIME_NONE;
|
||||
*end = GST_CLOCK_TIME_NONE;
|
||||
}
|
||||
|
||||
static GstStateChangeReturn
|
||||
gst_decklink_audio_sink_change_state (GstElement * element,
|
||||
GstStateChange transition)
|
||||
{
|
||||
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element);
|
||||
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
||||
|
||||
switch (transition) {
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
GST_OBJECT_LOCK (self);
|
||||
gst_audio_stream_align_mark_discont (self->stream_align);
|
||||
GST_OBJECT_UNLOCK (self);
|
||||
|
||||
g_mutex_lock (&self->output->lock);
|
||||
if (self->output->start_scheduled_playback)
|
||||
self->output->start_scheduled_playback (self->output->videosink);
|
||||
g_mutex_unlock (&self->output->lock);
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
gst_decklink_audio_sink_stop (self);
|
||||
break;
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||
if (ret == GST_STATE_CHANGE_FAILURE)
|
||||
return ret;
|
||||
|
||||
switch (transition) {
|
||||
default:
|
||||
break;
|
||||
}
|
||||
|
||||
return ret;
|
||||
}
|
||||
|
||||
static GstClock *
|
||||
gst_decklink_audio_sink_provide_clock (GstElement * element)
|
||||
{
|
||||
GstDecklinkAudioSink *self = GST_DECKLINK_AUDIO_SINK_CAST (element);
|
||||
|
||||
if (!self->output)
|
||||
return NULL;
|
||||
|
||||
return GST_CLOCK_CAST (gst_object_ref (self->output->clock));
|
||||
}
|
||||
Reference in New Issue
Block a user