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384 lines
11 KiB
C
384 lines
11 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-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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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch -v interaudiosrc ! queue ! audiosink
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* ]|
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*
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* The interaudiosrc element cannot be used effectively with gst-launch,
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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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* </refsect2>
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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 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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};
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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 ("audio/x-raw, format = (string) " GST_AUDIO_NE (S16) ", "
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"rate = (int) 48000, channels = (int) 2")
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);
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/* class initialization */
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G_DEFINE_TYPE (GstInterAudioSrc, gst_inter_audio_src, GST_TYPE_BASE_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_pad_template (element_class,
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gst_static_pad_template_get (&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->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", 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");
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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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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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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 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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const GstStructure *structure;
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GstAudioInfo info;
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gboolean ret;
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int sample_rate;
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GST_DEBUG_OBJECT (interaudiosrc, "set_caps");
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structure = gst_caps_get_structure (caps, 0);
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ret = gst_structure_get_int (structure, "rate", &sample_rate);
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if (ret) {
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interaudiosrc->sample_rate = sample_rate;
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ret = gst_pad_set_caps (src->srcpad, caps);
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}
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if (gst_audio_info_from_caps (&info, caps)) {
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interaudiosrc->finfo = info.finfo;
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}
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return ret;
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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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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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interaudiosrc->finfo = 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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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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GstClockTime timestamp = GST_BUFFER_TIMESTAMP (buffer);
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if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
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/* get duration to calculate end time */
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GstClockTime duration = GST_BUFFER_DURATION (buffer);
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if (GST_CLOCK_TIME_IS_VALID (duration)) {
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*end = timestamp + duration;
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}
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*start = timestamp;
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}
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} else {
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*start = -1;
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*end = -1;
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}
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}
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#define SIZE 1600
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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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GstBuffer *buffer;
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int n;
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GST_DEBUG_OBJECT (interaudiosrc, "create");
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buffer = NULL;
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g_mutex_lock (&interaudiosrc->surface->mutex);
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n = gst_adapter_available (interaudiosrc->surface->audio_adapter) / 4;
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if (n > SIZE * 3) {
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GST_WARNING ("flushing %d samples", SIZE / 2);
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gst_adapter_flush (interaudiosrc->surface->audio_adapter, (SIZE / 2) * 4);
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n -= (SIZE / 2);
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}
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if (n > SIZE)
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n = SIZE;
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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 * 4);
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} else {
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buffer = gst_buffer_new ();
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}
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g_mutex_unlock (&interaudiosrc->surface->mutex);
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if (n < SIZE) {
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GstMapInfo map;
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GstMemory *mem;
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GST_WARNING ("creating %d samples of silence", SIZE - n);
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mem = gst_allocator_alloc (NULL, (SIZE - n) * 4, NULL);
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if (gst_memory_map (mem, &map, GST_MAP_WRITE)) {
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gst_audio_format_fill_silence (interaudiosrc->finfo, map.data, map.size);
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gst_memory_unmap (mem, &map);
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}
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buffer = gst_buffer_make_writable (buffer);
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gst_buffer_prepend_memory (buffer, mem);
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}
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n = SIZE;
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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_TIMESTAMP (buffer) =
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gst_util_uint64_scale_int (interaudiosrc->n_samples, GST_SECOND,
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interaudiosrc->sample_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) =
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gst_util_uint64_scale_int (interaudiosrc->n_samples + n, GST_SECOND,
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interaudiosrc->sample_rate) - GST_BUFFER_TIMESTAMP (buffer);
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GST_BUFFER_OFFSET (buffer) = interaudiosrc->n_samples;
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GST_BUFFER_OFFSET_END (buffer) = -1;
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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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gboolean ret;
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GST_DEBUG_OBJECT (src, "query");
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_LATENCY:{
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GstClockTime min_latency, max_latency;
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min_latency = 30 * gst_util_uint64_scale_int (GST_SECOND, SIZE, 48000);
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max_latency = min_latency;
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GST_ERROR_OBJECT (src,
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"report latency min %" GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
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GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
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gst_query_set_latency (query,
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gst_base_src_is_live (src), min_latency, max_latency);
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ret = TRUE;
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break;
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}
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default:
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ret = GST_BASE_SRC_CLASS (gst_inter_audio_src_parent_class)->query (src,
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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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static GstCaps *
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gst_inter_audio_src_fixate (GstBaseSrc * src, GstCaps * caps)
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{
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GstStructure *structure;
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GST_DEBUG_OBJECT (src, "fixate");
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caps = gst_caps_make_writable (caps);
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structure = gst_caps_get_structure (caps, 0);
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gst_structure_fixate_field_nearest_int (structure, "channels", 2);
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gst_structure_fixate_field_nearest_int (structure, "rate", 48000);
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return caps;
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}
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