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406 lines
13 KiB
C
406 lines
13 KiB
C
/* GStreamer
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* Copyright (C) 2011 David A. Schleef <ds@schleef.org>
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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-interaudiosink
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* @title: gstinteraudiosink
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*
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* The interaudiosink element is an audio sink element. It is used
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* in connection with a interaudiosrc element in a different pipeline,
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* similar to intervideosink and intervideosrc.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 -v audiotestsrc ! queue ! interaudiosink
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* ]|
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*
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* The interaudiosink element cannot be used effectively with gst-launch-1.0,
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* as it requires a second pipeline in the application to receive the
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* audio.
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* See the gstintertest.c example in the gst-plugins-bad source code for
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* more details.
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*
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/base/gstbasesink.h>
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#include <gst/audio/audio.h>
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#include "gstinteraudiosink.h"
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_sink_debug_category);
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#define GST_CAT_DEFAULT gst_inter_audio_sink_debug_category
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/* prototypes */
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static void gst_inter_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_inter_audio_sink_get_property (GObject * object,
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guint property_id, GValue * value, GParamSpec * pspec);
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static void gst_inter_audio_sink_finalize (GObject * object);
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static void gst_inter_audio_sink_get_times (GstBaseSink * sink,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
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static gboolean gst_inter_audio_sink_start (GstBaseSink * sink);
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static gboolean gst_inter_audio_sink_stop (GstBaseSink * sink);
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static gboolean gst_inter_audio_sink_set_caps (GstBaseSink * sink,
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GstCaps * caps);
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static gboolean gst_inter_audio_sink_event (GstBaseSink * sink,
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GstEvent * event);
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static GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
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GstBuffer * buffer);
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static gboolean gst_inter_audio_sink_query (GstBaseSink * sink,
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GstQuery * query);
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enum
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{
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PROP_0,
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PROP_CHANNEL
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};
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#define DEFAULT_CHANNEL ("default")
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/* pad templates */
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static GstStaticPadTemplate gst_inter_audio_sink_sink_template =
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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 (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL))
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);
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/* class initialization */
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#define parent_class gst_inter_audio_sink_parent_class
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G_DEFINE_TYPE (GstInterAudioSink, gst_inter_audio_sink, GST_TYPE_BASE_SINK);
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GST_ELEMENT_REGISTER_DEFINE (interaudiosink, "interaudiosink",
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GST_RANK_NONE, GST_TYPE_INTER_AUDIO_SINK);
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static void
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gst_inter_audio_sink_class_init (GstInterAudioSinkClass * 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 *base_sink_class = GST_BASE_SINK_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (gst_inter_audio_sink_debug_category,
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"interaudiosink", 0, "debug category for interaudiosink element");
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gst_element_class_add_static_pad_template (element_class,
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&gst_inter_audio_sink_sink_template);
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gst_element_class_set_static_metadata (element_class,
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"Internal audio sink",
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"Sink/Audio",
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"Virtual audio sink for internal process communication",
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"David Schleef <ds@schleef.org>");
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gobject_class->set_property = gst_inter_audio_sink_set_property;
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gobject_class->get_property = gst_inter_audio_sink_get_property;
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gobject_class->finalize = gst_inter_audio_sink_finalize;
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base_sink_class->get_times =
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GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_times);
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base_sink_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_start);
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base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_stop);
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base_sink_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_event);
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base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_set_caps);
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base_sink_class->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
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base_sink_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_query);
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g_object_class_install_property (gobject_class, PROP_CHANNEL,
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g_param_spec_string ("channel", "Channel",
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"Channel name to match inter src and sink elements",
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DEFAULT_CHANNEL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_inter_audio_sink_init (GstInterAudioSink * interaudiosink)
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{
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interaudiosink->channel = g_strdup (DEFAULT_CHANNEL);
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interaudiosink->input_adapter = gst_adapter_new ();
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}
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void
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gst_inter_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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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
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switch (property_id) {
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case PROP_CHANNEL:
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g_free (interaudiosink->channel);
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interaudiosink->channel = g_value_dup_string (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_inter_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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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
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switch (property_id) {
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case PROP_CHANNEL:
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g_value_set_string (value, interaudiosink->channel);
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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_inter_audio_sink_finalize (GObject * object)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
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/* clean up object here */
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g_free (interaudiosink->channel);
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gst_object_unref (interaudiosink->input_adapter);
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G_OBJECT_CLASS (gst_inter_audio_sink_parent_class)->finalize (object);
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}
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static void
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gst_inter_audio_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
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GstClockTime * start, GstClockTime * end)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
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*start = GST_BUFFER_TIMESTAMP (buffer);
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if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
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*end = *start + GST_BUFFER_DURATION (buffer);
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} else {
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if (interaudiosink->info.rate > 0) {
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*end = *start +
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gst_util_uint64_scale_int (gst_buffer_get_size (buffer), GST_SECOND,
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interaudiosink->info.rate * interaudiosink->info.bpf);
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}
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}
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}
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}
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static gboolean
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gst_inter_audio_sink_start (GstBaseSink * sink)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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GST_DEBUG_OBJECT (interaudiosink, "start");
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interaudiosink->surface = gst_inter_surface_get (interaudiosink->channel);
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g_mutex_lock (&interaudiosink->surface->mutex);
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memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
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/* We want to write latency-time before syncing has happened */
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/* FIXME: The other side can change this value when it starts */
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gst_base_sink_set_render_delay (sink,
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interaudiosink->surface->audio_latency_time);
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g_mutex_unlock (&interaudiosink->surface->mutex);
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return TRUE;
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}
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static gboolean
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gst_inter_audio_sink_stop (GstBaseSink * sink)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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GST_DEBUG_OBJECT (interaudiosink, "stop");
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g_mutex_lock (&interaudiosink->surface->mutex);
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gst_adapter_clear (interaudiosink->surface->audio_adapter);
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memset (&interaudiosink->surface->audio_info, 0, sizeof (GstAudioInfo));
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g_mutex_unlock (&interaudiosink->surface->mutex);
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gst_inter_surface_unref (interaudiosink->surface);
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interaudiosink->surface = NULL;
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gst_adapter_clear (interaudiosink->input_adapter);
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return TRUE;
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}
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static gboolean
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gst_inter_audio_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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GstAudioInfo info;
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if (!gst_audio_info_from_caps (&info, caps)) {
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GST_ERROR_OBJECT (sink, "Failed to parse caps %" GST_PTR_FORMAT, caps);
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return FALSE;
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}
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g_mutex_lock (&interaudiosink->surface->mutex);
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interaudiosink->surface->audio_info = info;
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interaudiosink->info = info;
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/* TODO: Ideally we would drain the source here */
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gst_adapter_clear (interaudiosink->surface->audio_adapter);
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g_mutex_unlock (&interaudiosink->surface->mutex);
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return TRUE;
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}
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static gboolean
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gst_inter_audio_sink_event (GstBaseSink * sink, GstEvent * event)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:{
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GstBuffer *tmp;
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guint n;
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if ((n = gst_adapter_available (interaudiosink->input_adapter)) > 0) {
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g_mutex_lock (&interaudiosink->surface->mutex);
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tmp = gst_adapter_take_buffer (interaudiosink->input_adapter, n);
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gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
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g_mutex_unlock (&interaudiosink->surface->mutex);
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}
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break;
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}
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default:
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break;
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}
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return GST_BASE_SINK_CLASS (parent_class)->event (sink, event);
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}
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static GstFlowReturn
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gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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guint n, bpf;
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guint64 period_time, buffer_time;
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guint64 period_samples, buffer_samples;
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GST_DEBUG_OBJECT (interaudiosink, "render %" G_GSIZE_FORMAT,
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gst_buffer_get_size (buffer));
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bpf = interaudiosink->info.bpf;
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g_mutex_lock (&interaudiosink->surface->mutex);
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buffer_time = interaudiosink->surface->audio_buffer_time;
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period_time = interaudiosink->surface->audio_period_time;
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if (buffer_time < period_time) {
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GST_ERROR_OBJECT (interaudiosink,
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"Buffer time smaller than period time (%" GST_TIME_FORMAT " < %"
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GST_TIME_FORMAT ")", GST_TIME_ARGS (buffer_time),
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GST_TIME_ARGS (period_time));
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g_mutex_unlock (&interaudiosink->surface->mutex);
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return GST_FLOW_ERROR;
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}
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buffer_samples =
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gst_util_uint64_scale (buffer_time, interaudiosink->info.rate,
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GST_SECOND);
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period_samples =
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gst_util_uint64_scale (period_time, interaudiosink->info.rate,
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GST_SECOND);
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n = gst_adapter_available (interaudiosink->surface->audio_adapter) / bpf;
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while (n > buffer_samples) {
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GST_DEBUG_OBJECT (interaudiosink, "flushing %" GST_TIME_FORMAT,
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GST_TIME_ARGS (period_time));
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gst_adapter_flush (interaudiosink->surface->audio_adapter,
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period_samples * bpf);
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n -= period_samples;
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}
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n = gst_adapter_available (interaudiosink->input_adapter);
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if (period_samples * bpf > gst_buffer_get_size (buffer) + n) {
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gst_adapter_push (interaudiosink->input_adapter, gst_buffer_ref (buffer));
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} else {
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GstBuffer *tmp;
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if (n > 0) {
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tmp = gst_adapter_take_buffer (interaudiosink->input_adapter, n);
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gst_adapter_push (interaudiosink->surface->audio_adapter, tmp);
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}
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gst_adapter_push (interaudiosink->surface->audio_adapter,
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gst_buffer_ref (buffer));
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}
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g_mutex_unlock (&interaudiosink->surface->mutex);
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return GST_FLOW_OK;
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}
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static gboolean
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gst_inter_audio_sink_query (GstBaseSink * sink, GstQuery * query)
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{
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GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
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gboolean ret;
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GST_DEBUG_OBJECT (sink, "query");
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_LATENCY:{
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gboolean live, us_live;
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GstClockTime min_l, max_l;
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GST_DEBUG_OBJECT (sink, "latency query");
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if ((ret =
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gst_base_sink_query_latency (GST_BASE_SINK_CAST (sink), &live,
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&us_live, &min_l, &max_l))) {
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GstClockTime base_latency, min_latency, max_latency;
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/* we and upstream are both live, adjust the min_latency */
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if (live && us_live) {
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/* FIXME: The other side can change this value when it starts */
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base_latency = interaudiosink->surface->audio_latency_time;
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/* we cannot go lower than the buffer size and the min peer latency */
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min_latency = base_latency + min_l;
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/* the max latency is the max of the peer, we can delay an infinite
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* amount of time. */
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max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
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GST_DEBUG_OBJECT (sink,
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"peer min %" GST_TIME_FORMAT ", our min latency: %"
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GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
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GST_TIME_ARGS (min_latency));
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GST_DEBUG_OBJECT (sink,
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"peer max %" GST_TIME_FORMAT ", our max latency: %"
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GST_TIME_FORMAT, GST_TIME_ARGS (max_l),
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GST_TIME_ARGS (max_latency));
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} else {
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GST_DEBUG_OBJECT (sink,
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"peer or we are not live, don't care about latency");
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min_latency = min_l;
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max_latency = max_l;
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}
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gst_query_set_latency (query, live, min_latency, max_latency);
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}
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break;
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}
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default:
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ret =
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GST_BASE_SINK_CLASS (gst_inter_audio_sink_parent_class)->query (sink,
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query);
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break;
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}
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return ret;
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}
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