mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 05:38:58 +00:00
Conflicts: NEWS RELEASE common configure.ac docs/libs/gst-plugins-bad-libs-sections.txt docs/plugins/gst-plugins-bad-plugins.args docs/plugins/gst-plugins-bad-plugins.hierarchy docs/plugins/gst-plugins-bad-plugins.interfaces docs/plugins/inspect/plugin-adpcmdec.xml docs/plugins/inspect/plugin-adpcmenc.xml docs/plugins/inspect/plugin-assrender.xml docs/plugins/inspect/plugin-audiovisualizers.xml docs/plugins/inspect/plugin-autoconvert.xml docs/plugins/inspect/plugin-bayer.xml docs/plugins/inspect/plugin-bz2.xml docs/plugins/inspect/plugin-camerabin2.xml docs/plugins/inspect/plugin-celt.xml docs/plugins/inspect/plugin-dataurisrc.xml docs/plugins/inspect/plugin-debugutilsbad.xml docs/plugins/inspect/plugin-dtmf.xml docs/plugins/inspect/plugin-dtsdec.xml docs/plugins/inspect/plugin-dvbsuboverlay.xml docs/plugins/inspect/plugin-dvdspu.xml docs/plugins/inspect/plugin-faac.xml docs/plugins/inspect/plugin-faad.xml docs/plugins/inspect/plugin-gsm.xml docs/plugins/inspect/plugin-h264parse.xml docs/plugins/inspect/plugin-mms.xml docs/plugins/inspect/plugin-modplug.xml docs/plugins/inspect/plugin-mpeg2enc.xml docs/plugins/inspect/plugin-mpegdemux2.xml docs/plugins/inspect/plugin-mpegtsdemux.xml docs/plugins/inspect/plugin-mpegvideoparse.xml docs/plugins/inspect/plugin-mplex.xml docs/plugins/inspect/plugin-pcapparse.xml docs/plugins/inspect/plugin-rawparse.xml docs/plugins/inspect/plugin-rtpmux.xml docs/plugins/inspect/plugin-rtpvp8.xml docs/plugins/inspect/plugin-scaletempo.xml docs/plugins/inspect/plugin-schro.xml docs/plugins/inspect/plugin-sdp.xml docs/plugins/inspect/plugin-segmentclip.xml docs/plugins/inspect/plugin-shm.xml docs/plugins/inspect/plugin-videomaxrate.xml docs/plugins/inspect/plugin-videoparsersbad.xml docs/plugins/inspect/plugin-vp8.xml docs/plugins/inspect/plugin-y4mdec.xml ext/celt/gstceltdec.c ext/dts/gstdtsdec.c ext/modplug/gstmodplug.cc ext/opus/gstopusenc.c gst-libs/gst/video/gstbasevideocodec.c gst-libs/gst/video/gstbasevideocodec.h gst-libs/gst/video/gstbasevideodecoder.c gst-libs/gst/video/gstbasevideodecoder.h gst-libs/gst/video/gstbasevideoencoder.c gst-libs/gst/video/gstbasevideoencoder.h gst/adpcmdec/Makefile.am gst/audiovisualizers/gstbaseaudiovisualizer.c gst/h264parse/gsth264parse.c gst/mpegdemux/mpegtsparse.c gst/mpegtsdemux/mpegtsbase.c gst/mpegtsdemux/mpegtspacketizer.c gst/mpegtsdemux/mpegtsparse.c gst/mpegtsdemux/tsdemux.c gst/mpegtsdemux/tsdemux.h gst/mxf/mxfdemux.c gst/rawparse/gstaudioparse.c gst/videoparsers/gsth263parse.c gst/videoparsers/gsth264parse.c sys/d3dvideosink/d3dvideosink.c sys/decklink/gstdecklinksink.cpp sys/dvb/gstdvbsrc.c sys/shm/gstshmsrc.c sys/vdpau/h264/gstvdph264dec.c sys/vdpau/mpeg/gstvdpmpegdec.c tests/examples/opencv/gst_element_print_properties.c win32/common/config.h
368 lines
11 KiB
C
368 lines
11 KiB
C
/* GStreamer
|
|
* Copyright (C) 2011 David A. Schleef <ds@schleef.org>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Library General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2 of the License, or (at your option) any later version.
|
|
*
|
|
* This library is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Library General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Library General Public
|
|
* License along with this library; if not, write to the
|
|
* Free Software Foundation, Inc., 51 Franklin Street, Suite 500,
|
|
* Boston, MA 02110-1335, USA.
|
|
*/
|
|
/**
|
|
* SECTION:element-gstinteraudiosink
|
|
*
|
|
* The interaudiosink element is an audio sink element. It is used
|
|
* in connection with a interaudiosrc element in a different pipeline,
|
|
* similar to intervideosink and intervideosrc.
|
|
*
|
|
* <refsect2>
|
|
* <title>Example launch line</title>
|
|
* |[
|
|
* gst-launch -v audiotestsrc ! queue ! interaudiosink
|
|
* ]|
|
|
*
|
|
* The interaudiosink element cannot be used effectively with gst-launch,
|
|
* as it requires a second pipeline in the application to receive the
|
|
* audio.
|
|
* See the gstintertest.c example in the gst-plugins-bad source code for
|
|
* more details.
|
|
* </refsect2>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include "config.h"
|
|
#endif
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/base/gstbasesink.h>
|
|
#include <gst/audio/audio.h>
|
|
#include "gstinteraudiosink.h"
|
|
#include <string.h>
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_sink_debug_category);
|
|
#define GST_CAT_DEFAULT gst_inter_audio_sink_debug_category
|
|
|
|
/* prototypes */
|
|
|
|
|
|
static void gst_inter_audio_sink_set_property (GObject * object,
|
|
guint property_id, const GValue * value, GParamSpec * pspec);
|
|
static void gst_inter_audio_sink_get_property (GObject * object,
|
|
guint property_id, GValue * value, GParamSpec * pspec);
|
|
static void gst_inter_audio_sink_dispose (GObject * object);
|
|
static void gst_inter_audio_sink_finalize (GObject * object);
|
|
|
|
static GstCaps *gst_inter_audio_sink_get_caps (GstBaseSink * sink);
|
|
static gboolean gst_inter_audio_sink_set_caps (GstBaseSink * sink,
|
|
GstCaps * caps);
|
|
static GstFlowReturn gst_inter_audio_sink_buffer_alloc (GstBaseSink * sink,
|
|
guint64 offset, guint size, GstCaps * caps, GstBuffer ** buf);
|
|
static void gst_inter_audio_sink_get_times (GstBaseSink * sink,
|
|
GstBuffer * buffer, GstClockTime * start, GstClockTime * end);
|
|
static gboolean gst_inter_audio_sink_start (GstBaseSink * sink);
|
|
static gboolean gst_inter_audio_sink_stop (GstBaseSink * sink);
|
|
static gboolean gst_inter_audio_sink_unlock (GstBaseSink * sink);
|
|
static gboolean gst_inter_audio_sink_event (GstBaseSink * sink,
|
|
GstEvent * event);
|
|
static GstFlowReturn gst_inter_audio_sink_preroll (GstBaseSink * sink,
|
|
GstBuffer * buffer);
|
|
static GstFlowReturn gst_inter_audio_sink_render (GstBaseSink * sink,
|
|
GstBuffer * buffer);
|
|
static GstStateChangeReturn gst_inter_audio_sink_async_play (GstBaseSink *
|
|
sink);
|
|
static gboolean gst_inter_audio_sink_activate_pull (GstBaseSink * sink,
|
|
gboolean active);
|
|
static gboolean gst_inter_audio_sink_unlock_stop (GstBaseSink * sink);
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_CHANNEL
|
|
};
|
|
|
|
/* pad templates */
|
|
|
|
static GstStaticPadTemplate gst_inter_audio_sink_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("sink",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw-int, "
|
|
"endianness = (int) BYTE_ORDER, "
|
|
"signed = (boolean) true, "
|
|
"width = (int) 16, "
|
|
"depth = (int) 16, " "rate = (int) 48000, " "channels = (int) 2")
|
|
);
|
|
|
|
|
|
/* class initialization */
|
|
|
|
#define DEBUG_INIT(bla) \
|
|
GST_DEBUG_CATEGORY_INIT (gst_inter_audio_sink_debug_category, "interaudiosink", 0, \
|
|
"debug category for interaudiosink element");
|
|
|
|
GST_BOILERPLATE_FULL (GstInterAudioSink, gst_inter_audio_sink, GstBaseSink,
|
|
GST_TYPE_BASE_SINK, DEBUG_INIT);
|
|
|
|
static void
|
|
gst_inter_audio_sink_base_init (gpointer g_class)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
|
|
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&gst_inter_audio_sink_sink_template));
|
|
|
|
gst_element_class_set_details_simple (element_class,
|
|
"Internal audio sink",
|
|
"Sink/Audio",
|
|
"Virtual audio sink for internal process communication",
|
|
"David Schleef <ds@schleef.org>");
|
|
}
|
|
|
|
static void
|
|
gst_inter_audio_sink_class_init (GstInterAudioSinkClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
|
|
GstBaseSinkClass *base_sink_class = GST_BASE_SINK_CLASS (klass);
|
|
|
|
gobject_class->set_property = gst_inter_audio_sink_set_property;
|
|
gobject_class->get_property = gst_inter_audio_sink_get_property;
|
|
gobject_class->dispose = gst_inter_audio_sink_dispose;
|
|
gobject_class->finalize = gst_inter_audio_sink_finalize;
|
|
base_sink_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_caps);
|
|
base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_set_caps);
|
|
if (0)
|
|
base_sink_class->buffer_alloc =
|
|
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_buffer_alloc);
|
|
base_sink_class->get_times =
|
|
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_get_times);
|
|
base_sink_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_start);
|
|
base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_stop);
|
|
base_sink_class->unlock = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_unlock);
|
|
if (0)
|
|
base_sink_class->event = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_event);
|
|
//if (0)
|
|
base_sink_class->preroll = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_preroll);
|
|
base_sink_class->render = GST_DEBUG_FUNCPTR (gst_inter_audio_sink_render);
|
|
if (0)
|
|
base_sink_class->async_play =
|
|
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_async_play);
|
|
if (0)
|
|
base_sink_class->activate_pull =
|
|
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_activate_pull);
|
|
base_sink_class->unlock_stop =
|
|
GST_DEBUG_FUNCPTR (gst_inter_audio_sink_unlock_stop);
|
|
|
|
#if 0
|
|
g_object_class_install_property (gobject_class, PROP_CHANNEL,
|
|
g_param_spec_string ("channel", "Channel",
|
|
"Channel name to match inter src and sink elements",
|
|
"default", G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
#endif
|
|
}
|
|
|
|
static void
|
|
gst_inter_audio_sink_init (GstInterAudioSink * interaudiosink,
|
|
GstInterAudioSinkClass * interaudiosink_class)
|
|
{
|
|
interaudiosink->channel = g_strdup ("default");
|
|
}
|
|
|
|
void
|
|
gst_inter_audio_sink_set_property (GObject * object, guint property_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
|
|
|
|
switch (property_id) {
|
|
case PROP_CHANNEL:
|
|
g_free (interaudiosink->channel);
|
|
interaudiosink->channel = g_value_dup_string (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
void
|
|
gst_inter_audio_sink_get_property (GObject * object, guint property_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object);
|
|
|
|
switch (property_id) {
|
|
case PROP_CHANNEL:
|
|
g_value_set_string (value, interaudiosink->channel);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
void
|
|
gst_inter_audio_sink_dispose (GObject * object)
|
|
{
|
|
/* GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object); */
|
|
|
|
/* clean up as possible. may be called multiple times */
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
}
|
|
|
|
void
|
|
gst_inter_audio_sink_finalize (GObject * object)
|
|
{
|
|
/* GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (object); */
|
|
|
|
/* clean up object here */
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
|
|
|
|
static GstCaps *
|
|
gst_inter_audio_sink_get_caps (GstBaseSink * sink)
|
|
{
|
|
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
gst_inter_audio_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
|
|
{
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_inter_audio_sink_buffer_alloc (GstBaseSink * sink, guint64 offset,
|
|
guint size, GstCaps * caps, GstBuffer ** buf)
|
|
{
|
|
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
|
|
static void
|
|
gst_inter_audio_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
|
|
GstClockTime * start, GstClockTime * end)
|
|
{
|
|
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
|
|
|
|
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
|
|
*start = GST_BUFFER_TIMESTAMP (buffer);
|
|
if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
|
|
*end = *start + GST_BUFFER_DURATION (buffer);
|
|
} else {
|
|
if (interaudiosink->fps_n > 0) {
|
|
*end = *start +
|
|
gst_util_uint64_scale_int (GST_SECOND, interaudiosink->fps_d,
|
|
interaudiosink->fps_n);
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
}
|
|
|
|
static gboolean
|
|
gst_inter_audio_sink_start (GstBaseSink * sink)
|
|
{
|
|
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
|
|
|
|
GST_DEBUG ("start");
|
|
|
|
interaudiosink->surface = gst_inter_surface_get (interaudiosink->channel);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_inter_audio_sink_stop (GstBaseSink * sink)
|
|
{
|
|
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
|
|
|
|
GST_DEBUG ("stop");
|
|
|
|
g_mutex_lock (interaudiosink->surface->mutex);
|
|
gst_adapter_clear (interaudiosink->surface->audio_adapter);
|
|
g_mutex_unlock (interaudiosink->surface->mutex);
|
|
|
|
gst_inter_surface_unref (interaudiosink->surface);
|
|
interaudiosink->surface = NULL;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_inter_audio_sink_unlock (GstBaseSink * sink)
|
|
{
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_inter_audio_sink_event (GstBaseSink * sink, GstEvent * event)
|
|
{
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_inter_audio_sink_preroll (GstBaseSink * sink, GstBuffer * buffer)
|
|
{
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_inter_audio_sink_render (GstBaseSink * sink, GstBuffer * buffer)
|
|
{
|
|
GstInterAudioSink *interaudiosink = GST_INTER_AUDIO_SINK (sink);
|
|
int n;
|
|
|
|
GST_DEBUG ("render %d", GST_BUFFER_SIZE (buffer));
|
|
|
|
g_mutex_lock (interaudiosink->surface->mutex);
|
|
n = gst_adapter_available (interaudiosink->surface->audio_adapter) / 4;
|
|
if (n > (800 * 2 * 2)) {
|
|
GST_INFO ("flushing 800 samples");
|
|
gst_adapter_flush (interaudiosink->surface->audio_adapter, 800 * 4);
|
|
n -= 800;
|
|
}
|
|
gst_adapter_push (interaudiosink->surface->audio_adapter,
|
|
gst_buffer_ref (buffer));
|
|
g_mutex_unlock (interaudiosink->surface->mutex);
|
|
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_inter_audio_sink_async_play (GstBaseSink * sink)
|
|
{
|
|
|
|
return GST_STATE_CHANGE_SUCCESS;
|
|
}
|
|
|
|
static gboolean
|
|
gst_inter_audio_sink_activate_pull (GstBaseSink * sink, gboolean active)
|
|
{
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_inter_audio_sink_unlock_stop (GstBaseSink * sink)
|
|
{
|
|
|
|
return TRUE;
|
|
}
|