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891be51105
Split plugin into features including dynamic types which can be indiviually registered during a static build. More details here: https://gitlab.freedesktop.org/gstreamer/gst-build/-/merge_requests/199 https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/661 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2110>
489 lines
16 KiB
C
489 lines
16 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-interaudiosrc
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* @title: gstinteraudiosrc
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*
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* The interaudiosrc element is an audio source element. It is used
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* in connection with a interaudiosink element in a different pipeline.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 -v interaudiosrc ! queue ! autoaudiosink
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* ]|
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*
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* The interaudiosrc element cannot be used effectively with gst-launch-1.0,
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* as it requires a second pipeline in the application to send 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 "gstinteraudiosrc.h"
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#include <gst/gst.h>
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#include <gst/base/gstbasesrc.h>
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#include <gst/audio/audio.h>
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (gst_inter_audio_src_debug_category);
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#define GST_CAT_DEFAULT gst_inter_audio_src_debug_category
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/* prototypes */
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static void gst_inter_audio_src_set_property (GObject * object,
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guint property_id, const GValue * value, GParamSpec * pspec);
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static void gst_inter_audio_src_get_property (GObject * object,
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guint property_id, GValue * value, GParamSpec * pspec);
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static void gst_inter_audio_src_finalize (GObject * object);
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static GstCaps *gst_inter_audio_src_get_caps (GstBaseSrc * src,
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GstCaps * filter);
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static gboolean gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps);
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static gboolean gst_inter_audio_src_start (GstBaseSrc * src);
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static gboolean gst_inter_audio_src_stop (GstBaseSrc * src);
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static void
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gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
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GstClockTime * start, GstClockTime * end);
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static GstFlowReturn
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gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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GstBuffer ** buf);
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static gboolean gst_inter_audio_src_query (GstBaseSrc * src, GstQuery * query);
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static GstCaps *gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps);
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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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PROP_BUFFER_TIME,
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PROP_LATENCY_TIME,
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PROP_PERIOD_TIME
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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_src_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL)
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", layout = (string) interleaved")
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);
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/* class initialization */
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#define parent_class gst_inter_audio_src_parent_class
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G_DEFINE_TYPE (GstInterAudioSrc, gst_inter_audio_src, GST_TYPE_BASE_SRC);
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GST_ELEMENT_REGISTER_DEFINE (interaudiosrc, "interaudiosrc",
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GST_RANK_NONE, GST_TYPE_INTER_AUDIO_SRC);
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static void
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gst_inter_audio_src_class_init (GstInterAudioSrcClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstBaseSrcClass *base_src_class = GST_BASE_SRC_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (gst_inter_audio_src_debug_category, "interaudiosrc",
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0, "debug category for interaudiosrc element");
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gst_element_class_add_static_pad_template (element_class,
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&gst_inter_audio_src_src_template);
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gst_element_class_set_static_metadata (element_class,
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"Internal audio source",
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"Source/Audio",
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"Virtual audio source for internal process communication",
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"David Schleef <ds@schleef.org>");
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gobject_class->set_property = gst_inter_audio_src_set_property;
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gobject_class->get_property = gst_inter_audio_src_get_property;
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gobject_class->finalize = gst_inter_audio_src_finalize;
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base_src_class->get_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_caps);
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base_src_class->set_caps = GST_DEBUG_FUNCPTR (gst_inter_audio_src_set_caps);
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base_src_class->start = GST_DEBUG_FUNCPTR (gst_inter_audio_src_start);
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base_src_class->stop = GST_DEBUG_FUNCPTR (gst_inter_audio_src_stop);
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base_src_class->get_times = GST_DEBUG_FUNCPTR (gst_inter_audio_src_get_times);
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base_src_class->create = GST_DEBUG_FUNCPTR (gst_inter_audio_src_create);
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base_src_class->query = GST_DEBUG_FUNCPTR (gst_inter_audio_src_query);
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base_src_class->fixate = GST_DEBUG_FUNCPTR (gst_inter_audio_src_fixate);
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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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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", 1, G_MAXUINT64, DEFAULT_AUDIO_BUFFER_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_LATENCY_TIME,
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g_param_spec_uint64 ("latency-time", "Latency Time",
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"Latency as reported by the source",
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1, G_MAXUINT64, DEFAULT_AUDIO_LATENCY_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PERIOD_TIME,
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g_param_spec_uint64 ("period-time", "Period Time",
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"The minimum amount of data to read in each iteration",
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1, G_MAXUINT64, DEFAULT_AUDIO_PERIOD_TIME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_inter_audio_src_init (GstInterAudioSrc * interaudiosrc)
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{
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gst_base_src_set_format (GST_BASE_SRC (interaudiosrc), GST_FORMAT_TIME);
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gst_base_src_set_live (GST_BASE_SRC (interaudiosrc), TRUE);
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gst_base_src_set_blocksize (GST_BASE_SRC (interaudiosrc), -1);
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interaudiosrc->channel = g_strdup (DEFAULT_CHANNEL);
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interaudiosrc->buffer_time = DEFAULT_AUDIO_BUFFER_TIME;
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interaudiosrc->latency_time = DEFAULT_AUDIO_LATENCY_TIME;
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interaudiosrc->period_time = DEFAULT_AUDIO_PERIOD_TIME;
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}
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void
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gst_inter_audio_src_set_property (GObject * object, guint property_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
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switch (property_id) {
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case PROP_CHANNEL:
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g_free (interaudiosrc->channel);
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interaudiosrc->channel = g_value_dup_string (value);
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break;
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case PROP_BUFFER_TIME:
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interaudiosrc->buffer_time = g_value_get_uint64 (value);
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break;
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case PROP_LATENCY_TIME:
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interaudiosrc->latency_time = g_value_get_uint64 (value);
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break;
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case PROP_PERIOD_TIME:
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interaudiosrc->period_time = g_value_get_uint64 (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_src_get_property (GObject * object, guint property_id,
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GValue * value, GParamSpec * pspec)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
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switch (property_id) {
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case PROP_CHANNEL:
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g_value_set_string (value, interaudiosrc->channel);
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break;
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case PROP_BUFFER_TIME:
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g_value_set_uint64 (value, interaudiosrc->buffer_time);
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break;
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case PROP_LATENCY_TIME:
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g_value_set_uint64 (value, interaudiosrc->latency_time);
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break;
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case PROP_PERIOD_TIME:
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g_value_set_uint64 (value, interaudiosrc->period_time);
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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_src_finalize (GObject * object)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (object);
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/* clean up object here */
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g_free (interaudiosrc->channel);
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G_OBJECT_CLASS (gst_inter_audio_src_parent_class)->finalize (object);
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}
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static GstCaps *
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gst_inter_audio_src_get_caps (GstBaseSrc * src, GstCaps * filter)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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GstCaps *caps;
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GST_DEBUG_OBJECT (interaudiosrc, "get_caps");
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if (!interaudiosrc->surface)
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return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter);
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g_mutex_lock (&interaudiosrc->surface->mutex);
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if (interaudiosrc->surface->audio_info.finfo) {
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caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
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if (filter) {
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GstCaps *tmp;
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tmp = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
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gst_caps_unref (caps);
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caps = tmp;
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}
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} else {
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caps = NULL;
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}
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g_mutex_unlock (&interaudiosrc->surface->mutex);
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if (caps)
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return caps;
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else
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return GST_BASE_SRC_CLASS (parent_class)->get_caps (src, filter);
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}
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static gboolean
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gst_inter_audio_src_set_caps (GstBaseSrc * src, GstCaps * caps)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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GST_DEBUG_OBJECT (interaudiosrc, "set_caps");
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if (!gst_audio_info_from_caps (&interaudiosrc->info, caps)) {
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GST_ERROR_OBJECT (src, "Failed to parse caps %" GST_PTR_FORMAT, caps);
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return FALSE;
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}
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return TRUE;
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}
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static gboolean
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gst_inter_audio_src_start (GstBaseSrc * src)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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GST_DEBUG_OBJECT (interaudiosrc, "start");
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interaudiosrc->surface = gst_inter_surface_get (interaudiosrc->channel);
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interaudiosrc->timestamp_offset = 0;
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interaudiosrc->n_samples = 0;
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g_mutex_lock (&interaudiosrc->surface->mutex);
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interaudiosrc->surface->audio_buffer_time = interaudiosrc->buffer_time;
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interaudiosrc->surface->audio_latency_time = interaudiosrc->latency_time;
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interaudiosrc->surface->audio_period_time = interaudiosrc->period_time;
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g_mutex_unlock (&interaudiosrc->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_src_stop (GstBaseSrc * src)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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GST_DEBUG_OBJECT (interaudiosrc, "stop");
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gst_inter_surface_unref (interaudiosrc->surface);
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interaudiosrc->surface = NULL;
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return TRUE;
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}
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static void
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gst_inter_audio_src_get_times (GstBaseSrc * src, GstBuffer * buffer,
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GstClockTime * start, GstClockTime * end)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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GST_DEBUG_OBJECT (src, "get_times");
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/* for live sources, sync on the timestamp of the buffer */
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if (gst_base_src_is_live (src)) {
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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 (interaudiosrc->info.rate > 0) {
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*end = *start +
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gst_util_uint64_scale_int (gst_buffer_get_size (buffer),
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GST_SECOND, interaudiosrc->info.rate * interaudiosrc->info.bpf);
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}
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}
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}
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}
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}
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static GstFlowReturn
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gst_inter_audio_src_create (GstBaseSrc * src, guint64 offset, guint size,
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GstBuffer ** buf)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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GstCaps *caps;
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GstBuffer *buffer;
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guint n, bpf;
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guint64 period_time;
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guint64 period_samples;
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GST_DEBUG_OBJECT (interaudiosrc, "create");
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buffer = NULL;
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caps = NULL;
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g_mutex_lock (&interaudiosrc->surface->mutex);
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if (interaudiosrc->surface->audio_info.finfo) {
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if (!gst_audio_info_is_equal (&interaudiosrc->surface->audio_info,
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&interaudiosrc->info)) {
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caps = gst_audio_info_to_caps (&interaudiosrc->surface->audio_info);
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interaudiosrc->timestamp_offset +=
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gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
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interaudiosrc->info.rate);
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interaudiosrc->n_samples = 0;
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}
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}
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bpf = interaudiosrc->surface->audio_info.bpf;
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period_time = interaudiosrc->surface->audio_period_time;
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period_samples =
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gst_util_uint64_scale (period_time, interaudiosrc->info.rate, GST_SECOND);
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if (bpf > 0)
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n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / bpf;
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else
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n = 0;
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if (n > period_samples)
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n = period_samples;
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if (n > 0) {
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buffer = gst_adapter_take_buffer (interaudiosrc->surface->audio_adapter,
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n * bpf);
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} else {
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buffer = gst_buffer_new ();
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GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
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}
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g_mutex_unlock (&interaudiosrc->surface->mutex);
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if (caps) {
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gboolean ret = gst_base_src_set_caps (src, caps);
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gst_caps_unref (caps);
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if (!ret) {
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GST_ERROR_OBJECT (src, "Failed to set caps %" GST_PTR_FORMAT, caps);
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if (buffer)
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gst_buffer_unref (buffer);
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return GST_FLOW_NOT_NEGOTIATED;
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}
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}
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buffer = gst_buffer_make_writable (buffer);
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bpf = interaudiosrc->info.bpf;
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if (n < period_samples) {
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GstMapInfo map;
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GstMemory *mem;
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GST_DEBUG_OBJECT (interaudiosrc,
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"creating %" G_GUINT64_FORMAT " samples of silence",
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period_samples - n);
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mem = gst_allocator_alloc (NULL, (period_samples - n) * bpf, NULL);
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if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
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gst_audio_format_info_fill_silence (interaudiosrc->info.finfo, map.data,
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map.size);
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gst_memory_unmap (mem, &map);
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}
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gst_buffer_prepend_memory (buffer, mem);
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}
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n = period_samples;
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GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
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GST_BUFFER_OFFSET_END (buffer) = interaudiosrc->n_samples + n;
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GST_BUFFER_DTS (buffer) = GST_CLOCK_TIME_NONE;
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GST_BUFFER_PTS (buffer) = interaudiosrc->timestamp_offset +
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gst_util_uint64_scale (interaudiosrc->n_samples, GST_SECOND,
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interaudiosrc->info.rate);
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GST_DEBUG_OBJECT (interaudiosrc, "create ts %" GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)));
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GST_BUFFER_DURATION (buffer) = interaudiosrc->timestamp_offset +
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gst_util_uint64_scale (interaudiosrc->n_samples + n, GST_SECOND,
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interaudiosrc->info.rate) - GST_BUFFER_TIMESTAMP (buffer);
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GST_BUFFER_FLAG_UNSET (buffer, GST_BUFFER_FLAG_DISCONT);
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if (interaudiosrc->n_samples == 0) {
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GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);
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}
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interaudiosrc->n_samples += n;
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*buf = buffer;
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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_src_query (GstBaseSrc * src, GstQuery * query)
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{
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GstInterAudioSrc *interaudiosrc = GST_INTER_AUDIO_SRC (src);
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gboolean ret;
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GST_DEBUG_OBJECT (src, "query");
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:{
|
|
GstClockTime min_latency, max_latency;
|
|
|
|
min_latency = interaudiosrc->latency_time;
|
|
max_latency = interaudiosrc->buffer_time;
|
|
|
|
GST_DEBUG_OBJECT (src,
|
|
"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
|
|
|
|
gst_query_set_latency (query,
|
|
gst_base_src_is_live (src), min_latency, max_latency);
|
|
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
ret = GST_BASE_SRC_CLASS (gst_inter_audio_src_parent_class)->query (src,
|
|
query);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps)
|
|
{
|
|
GstStructure *structure;
|
|
|
|
GST_DEBUG_OBJECT (src, "fixate");
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
caps = gst_caps_truncate (caps);
|
|
|
|
structure = gst_caps_get_structure (caps, 0);
|
|
|
|
gst_structure_fixate_field_string (structure, "format", GST_AUDIO_NE (S16));
|
|
gst_structure_fixate_field_nearest_int (structure, "channels", 2);
|
|
gst_structure_fixate_field_nearest_int (structure, "rate", 48000);
|
|
gst_structure_fixate_field_string (structure, "layout", "interleaved");
|
|
|
|
return caps;
|
|
}
|