mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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162 lines
6.1 KiB
C++
162 lines
6.1 KiB
C++
/* GStreamer
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* Copyright (C) 2021 Sebastian Dröge <sebastian@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin Street, Suite 500,
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* Boston, MA 02110-1335, USA.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/audio/audio.h>
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#include "gstajacommon.h"
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#include "gstajasrcdemux.h"
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GST_DEBUG_CATEGORY_STATIC(gst_aja_src_demux_debug);
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#define GST_CAT_DEFAULT gst_aja_src_demux_debug
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static GstStaticPadTemplate video_src_template = GST_STATIC_PAD_TEMPLATE(
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"video", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS("video/x-raw"));
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static GstStaticPadTemplate audio_src_template = GST_STATIC_PAD_TEMPLATE(
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"audio", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS("audio/x-raw"));
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE(
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"sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS("video/x-raw"));
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static GstFlowReturn gst_aja_src_demux_sink_chain(GstPad *pad,
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GstObject *parent,
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GstBuffer *buffer);
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static gboolean gst_aja_src_demux_sink_event(GstPad *pad, GstObject *parent,
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GstEvent *event);
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#define parent_class gst_aja_src_demux_parent_class
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G_DEFINE_TYPE(GstAjaSrcDemux, gst_aja_src_demux, GST_TYPE_ELEMENT);
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static void gst_aja_src_demux_class_init(GstAjaSrcDemuxClass *klass) {
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GstElementClass *element_class = GST_ELEMENT_CLASS(klass);
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gst_element_class_add_static_pad_template(element_class, &sink_template);
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gst_element_class_add_static_pad_template(element_class, &video_src_template);
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gst_element_class_add_static_pad_template(element_class, &audio_src_template);
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gst_element_class_set_static_metadata(
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element_class, "AJA audio/video source demuxer", "Audio/Video/Demux",
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"Demuxes audio/video from video buffers",
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"Sebastian Dröge <sebastian@centricular.com>");
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GST_DEBUG_CATEGORY_INIT(gst_aja_src_demux_debug, "ajasrcdemux", 0,
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"AJA source demuxer");
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}
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static void gst_aja_src_demux_init(GstAjaSrcDemux *self) {
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self->sink = gst_pad_new_from_static_template(&sink_template, "sink");
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gst_pad_set_chain_function(self->sink,
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GST_DEBUG_FUNCPTR(gst_aja_src_demux_sink_chain));
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gst_pad_set_event_function(self->sink,
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GST_DEBUG_FUNCPTR(gst_aja_src_demux_sink_event));
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gst_element_add_pad(GST_ELEMENT(self), self->sink);
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self->audio_src =
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gst_pad_new_from_static_template(&audio_src_template, "audio");
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gst_pad_use_fixed_caps(self->audio_src);
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gst_element_add_pad(GST_ELEMENT(self), self->audio_src);
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self->video_src =
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gst_pad_new_from_static_template(&video_src_template, "video");
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gst_pad_use_fixed_caps(self->video_src);
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gst_element_add_pad(GST_ELEMENT(self), self->video_src);
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}
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static GstFlowReturn gst_aja_src_demux_sink_chain(GstPad *pad,
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GstObject *parent,
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GstBuffer *buffer) {
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GstAjaSrcDemux *self = GST_AJA_SRC_DEMUX(parent);
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GstAjaAudioMeta *meta = gst_buffer_get_aja_audio_meta(buffer);
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GstFlowReturn audio_flow_ret = GST_FLOW_OK;
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GstFlowReturn video_flow_ret = GST_FLOW_OK;
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if (meta) {
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GstBuffer *audio_buffer;
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buffer = gst_buffer_make_writable(buffer);
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meta = gst_buffer_get_aja_audio_meta(buffer);
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audio_buffer = gst_buffer_ref(meta->buffer);
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gst_buffer_remove_meta(buffer, GST_META_CAST(meta));
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audio_flow_ret = gst_pad_push(self->audio_src, audio_buffer);
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} else {
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GstEvent *event =
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gst_event_new_gap(GST_BUFFER_PTS(buffer), GST_BUFFER_DURATION(buffer));
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gst_pad_push_event(self->audio_src, event);
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}
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video_flow_ret = gst_pad_push(self->video_src, buffer);
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// Combine flows the way it makes sense
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if (video_flow_ret == GST_FLOW_NOT_LINKED &&
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audio_flow_ret == GST_FLOW_NOT_LINKED)
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return GST_FLOW_NOT_LINKED;
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if (video_flow_ret == GST_FLOW_EOS && audio_flow_ret == GST_FLOW_EOS)
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return GST_FLOW_EOS;
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if (video_flow_ret == GST_FLOW_FLUSHING ||
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video_flow_ret <= GST_FLOW_NOT_NEGOTIATED)
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return video_flow_ret;
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if (audio_flow_ret == GST_FLOW_FLUSHING ||
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audio_flow_ret <= GST_FLOW_NOT_NEGOTIATED)
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return audio_flow_ret;
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return GST_FLOW_OK;
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}
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static gboolean gst_aja_src_demux_sink_event(GstPad *pad, GstObject *parent,
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GstEvent *event) {
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GstAjaSrcDemux *self = GST_AJA_SRC_DEMUX(parent);
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switch (GST_EVENT_TYPE(event)) {
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case GST_EVENT_CAPS: {
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GstCaps *caps;
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GstStructure *s;
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GstAudioInfo audio_info;
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gint audio_channels = 0;
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gst_event_parse_caps(event, &caps);
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s = gst_caps_get_structure(caps, 0);
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gst_structure_get_int(s, "audio-channels", &audio_channels);
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GstCaps *audio_caps, *video_caps;
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gst_audio_info_init(&audio_info);
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gst_audio_info_set_format(&audio_info, GST_AUDIO_FORMAT_S32LE, 48000,
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audio_channels ? audio_channels : 1, NULL);
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audio_caps = gst_audio_info_to_caps(&audio_info);
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gst_pad_set_caps(self->audio_src, audio_caps);
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gst_caps_unref(audio_caps);
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video_caps = gst_caps_ref(caps);
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gst_event_unref(event);
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video_caps = gst_caps_make_writable(video_caps);
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s = gst_caps_get_structure(video_caps, 0);
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gst_structure_remove_field(s, "audio-channels");
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gst_pad_set_caps(self->video_src, video_caps);
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gst_caps_unref(video_caps);
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return TRUE;
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}
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default:
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return gst_pad_event_default(pad, parent, event);
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}
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}
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