/* GStreamer * Copyright (C) <1999> Erik Walthinsen * * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #define GST_TYPE_AUDIORATE \ (gst_audiorate_get_type()) #define GST_AUDIORATE(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORATE,GstAudiorate)) #define GST_AUDIORATE_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORATE,GstAudiorate)) #define GST_IS_AUDIORATE(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORATE)) #define GST_IS_AUDIORATE_CLASS(obj) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORATE)) typedef struct _GstAudiorate GstAudiorate; typedef struct _GstAudiorateClass GstAudiorateClass; struct _GstAudiorate { GstElement element; GstPad *sinkpad, *srcpad; gint bytes_per_sample; /* audio state */ guint64 next_offset; guint64 in, out, add, drop; gboolean silent; }; struct _GstAudiorateClass { GstElementClass parent_class; }; /* elementfactory information */ static GstElementDetails audiorate_details = GST_ELEMENT_DETAILS ("Audio rate adjuster", "Filter/Effect/Audio", "Drops/duplicates/adjusts timestamps on audio samples to make a perfect stream", "Wim Taymans "); /* GstAudiorate signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; #define DEFAULT_SILENT TRUE enum { ARG_0, ARG_IN, ARG_OUT, ARG_ADD, ARG_DROP, ARG_SILENT, /* FILL ME */ }; static GstStaticPadTemplate gst_audiorate_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS) ); static GstStaticPadTemplate gst_audiorate_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_AUDIO_INT_PAD_TEMPLATE_CAPS) ); static void gst_audiorate_base_init (gpointer g_class); static void gst_audiorate_class_init (GstAudiorateClass * klass); static void gst_audiorate_init (GstAudiorate * audiorate); static GstFlowReturn gst_audiorate_chain (GstPad * pad, GstBuffer * buf); static void gst_audiorate_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audiorate_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstStateChangeReturn gst_audiorate_change_state (GstElement * element, GstStateChange transition); static GstElementClass *parent_class = NULL; /*static guint gst_audiorate_signals[LAST_SIGNAL] = { 0 }; */ static GType gst_audiorate_get_type (void) { static GType audiorate_type = 0; if (!audiorate_type) { static const GTypeInfo audiorate_info = { sizeof (GstAudiorateClass), gst_audiorate_base_init, NULL, (GClassInitFunc) gst_audiorate_class_init, NULL, NULL, sizeof (GstAudiorate), 0, (GInstanceInitFunc) gst_audiorate_init, }; audiorate_type = g_type_register_static (GST_TYPE_ELEMENT, "GstAudiorate", &audiorate_info, 0); } return audiorate_type; } static void gst_audiorate_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); gst_element_class_set_details (element_class, &audiorate_details); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_audiorate_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gst_audiorate_src_template)); } static void gst_audiorate_class_init (GstAudiorateClass * klass) { GObjectClass *object_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); parent_class = g_type_class_peek_parent (klass); object_class->set_property = gst_audiorate_set_property; object_class->get_property = gst_audiorate_get_property; g_object_class_install_property (object_class, ARG_IN, g_param_spec_uint64 ("in", "In", "Number of input samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE)); g_object_class_install_property (object_class, ARG_OUT, g_param_spec_uint64 ("out", "Out", "Number of output samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE)); g_object_class_install_property (object_class, ARG_ADD, g_param_spec_uint64 ("add", "Add", "Number of added samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE)); g_object_class_install_property (object_class, ARG_DROP, g_param_spec_uint64 ("drop", "Drop", "Number of dropped samples", 0, G_MAXUINT64, 0, G_PARAM_READABLE)); g_object_class_install_property (object_class, ARG_SILENT, g_param_spec_boolean ("silent", "silent", "Don't emit notify for dropped and duplicated frames", DEFAULT_SILENT, G_PARAM_READWRITE)); element_class->change_state = gst_audiorate_change_state; } static gboolean gst_audiorate_setcaps (GstPad * pad, GstCaps * caps) { GstAudiorate *audiorate; GstStructure *structure; GstPad *otherpad; gint ret, channels, depth; audiorate = GST_AUDIORATE (gst_pad_get_parent (pad)); otherpad = (pad == audiorate->srcpad) ? audiorate->sinkpad : audiorate->srcpad; if (!gst_pad_set_caps (otherpad, caps)) return FALSE; structure = gst_caps_get_structure (caps, 0); ret = gst_structure_get_int (structure, "channels", &channels); ret &= gst_structure_get_int (structure, "depth", &depth); if (!ret) return FALSE; audiorate->bytes_per_sample = channels * (depth / 8); if (audiorate->bytes_per_sample == 0) audiorate->bytes_per_sample = 1; return TRUE; } static void gst_audiorate_init (GstAudiorate * audiorate) { audiorate->sinkpad = gst_pad_new_from_static_template (&gst_audiorate_sink_template, "sink"); gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->sinkpad); gst_pad_set_chain_function (audiorate->sinkpad, gst_audiorate_chain); gst_pad_set_setcaps_function (audiorate->sinkpad, gst_audiorate_setcaps); gst_pad_set_getcaps_function (audiorate->sinkpad, gst_pad_proxy_getcaps); audiorate->srcpad = gst_pad_new_from_static_template (&gst_audiorate_src_template, "src"); gst_element_add_pad (GST_ELEMENT (audiorate), audiorate->srcpad); gst_pad_set_setcaps_function (audiorate->srcpad, gst_audiorate_setcaps); gst_pad_set_getcaps_function (audiorate->srcpad, gst_pad_proxy_getcaps); audiorate->bytes_per_sample = 1; audiorate->in = 0; audiorate->out = 0; audiorate->drop = 0; audiorate->add = 0; audiorate->silent = DEFAULT_SILENT; } static GstFlowReturn gst_audiorate_chain (GstPad * pad, GstBuffer * buf) { GstAudiorate *audiorate; GstClockTime in_time, in_duration; guint64 in_offset, in_offset_end; gint in_size; audiorate = GST_AUDIORATE (gst_pad_get_parent (pad)); audiorate->in++; in_time = GST_BUFFER_TIMESTAMP (buf); in_duration = GST_BUFFER_DURATION (buf); in_size = GST_BUFFER_SIZE (buf); in_offset = GST_BUFFER_OFFSET (buf); in_offset_end = GST_BUFFER_OFFSET_END (buf); if (in_offset == GST_CLOCK_TIME_NONE || in_offset_end == GST_CLOCK_TIME_NONE) { GST_WARNING_OBJECT (audiorate, "audiorate got buffer without offsets"); } /* do we need to insert samples */ if (in_offset > audiorate->next_offset) { GstBuffer *fill; gint fillsize; guint64 fillsamples; fillsamples = in_offset - audiorate->next_offset; fillsize = fillsamples * audiorate->bytes_per_sample; fill = gst_buffer_new_and_alloc (fillsize); memset (GST_BUFFER_DATA (fill), 0, fillsize); GST_LOG_OBJECT (audiorate, "inserting %lld samples", fillsamples); GST_BUFFER_DURATION (fill) = in_duration * fillsize / in_size; GST_BUFFER_TIMESTAMP (fill) = in_time - GST_BUFFER_DURATION (fill); GST_BUFFER_OFFSET (fill) = audiorate->next_offset; GST_BUFFER_OFFSET_END (fill) = in_offset; gst_pad_push (audiorate->srcpad, fill); audiorate->out++; audiorate->add += fillsamples; if (!audiorate->silent) g_object_notify (G_OBJECT (audiorate), "add"); } else if (in_offset < audiorate->next_offset) { /* need to remove samples */ if (in_offset_end <= audiorate->next_offset) { guint64 drop = in_size / audiorate->bytes_per_sample; audiorate->drop += drop; GST_LOG_OBJECT (audiorate, "dropping %lld samples", drop); /* we can drop the buffer completely */ gst_buffer_unref (buf); if (!audiorate->silent) g_object_notify (G_OBJECT (audiorate), "drop"); return GST_FLOW_OK; } else { guint64 truncsamples, truncsize, leftsize; GstBuffer *trunc; /* truncate buffer */ truncsamples = audiorate->next_offset - in_offset; truncsize = truncsamples * audiorate->bytes_per_sample; leftsize = in_size - truncsize; trunc = gst_buffer_create_sub (buf, truncsize, in_size); GST_BUFFER_DURATION (trunc) = in_duration * leftsize / in_size; GST_BUFFER_TIMESTAMP (trunc) = in_time + in_duration - GST_BUFFER_DURATION (trunc); GST_BUFFER_OFFSET (trunc) = audiorate->next_offset; GST_BUFFER_OFFSET_END (trunc) = in_offset_end; GST_LOG_OBJECT (audiorate, "truncating %lld samples", truncsamples); gst_buffer_unref (buf); buf = trunc; audiorate->drop += truncsamples; } } gst_pad_push (audiorate->srcpad, buf); audiorate->out++; audiorate->next_offset = in_offset_end; return GST_FLOW_OK; } static void gst_audiorate_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudiorate *audiorate = GST_AUDIORATE (object); switch (prop_id) { case ARG_SILENT: audiorate->silent = g_value_get_boolean (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audiorate_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudiorate *audiorate = GST_AUDIORATE (object); switch (prop_id) { case ARG_IN: g_value_set_uint64 (value, audiorate->in); break; case ARG_OUT: g_value_set_uint64 (value, audiorate->out); break; case ARG_ADD: g_value_set_uint64 (value, audiorate->add); break; case ARG_DROP: g_value_set_uint64 (value, audiorate->drop); break; case ARG_SILENT: g_value_set_boolean (value, audiorate->silent); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStateChangeReturn gst_audiorate_change_state (GstElement * element, GstStateChange transition) { GstAudiorate *audiorate = GST_AUDIORATE (element); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: audiorate->next_offset = 0; break; default: break; } if (parent_class->change_state) return parent_class->change_state (element, transition); return GST_STATE_CHANGE_SUCCESS; } static gboolean plugin_init (GstPlugin * plugin) { return gst_element_register (plugin, "audiorate", GST_RANK_NONE, GST_TYPE_AUDIORATE); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "audiorate", "Adjusts audio frames", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)