/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2001 Thomas * 2005,2006 Wim Taymans * 2013 Sebastian Dröge * * audiomixer.c: AudioMixer element, N in, one out, samples are added * * 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 St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-audiomixer * @title: audiomixer * * The audiomixer allows to mix several streams into one by adding the data. * Mixed data is clamped to the min/max values of the data format. * * Unlike the adder element audiomixer properly synchronises all input streams. * * The input pads are from a GstPad subclass and have additional * properties to mute each pad individually and set the volume: * * * "mute": Whether to mute the pad or not (#gboolean) * * "volume": The volume of the pad, between 0.0 and 10.0 (#gdouble) * * ## Example launch line * |[ * gst-launch-1.0 audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix. * ]| This pipeline produces two sine waves mixed together. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstaudiomixer.h" #include #include /* strcmp */ #include "gstaudiomixerorc.h" #include "gstaudiointerleave.h" #define GST_CAT_DEFAULT gst_audiomixer_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); #define DEFAULT_PAD_VOLUME (1.0) #define DEFAULT_PAD_MUTE (FALSE) /* some defines for audio processing */ /* the volume factor is a range from 0.0 to (arbitrary) VOLUME_MAX_DOUBLE = 10.0 * we map 1.0 to VOLUME_UNITY_INT* */ #define VOLUME_UNITY_INT8 8 /* internal int for unity 2^(8-5) */ #define VOLUME_UNITY_INT8_BIT_SHIFT 3 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT16 2048 /* internal int for unity 2^(16-5) */ #define VOLUME_UNITY_INT16_BIT_SHIFT 11 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT24 524288 /* internal int for unity 2^(24-5) */ #define VOLUME_UNITY_INT24_BIT_SHIFT 19 /* number of bits to shift for unity */ #define VOLUME_UNITY_INT32 134217728 /* internal int for unity 2^(32-5) */ #define VOLUME_UNITY_INT32_BIT_SHIFT 27 enum { PROP_PAD_0, PROP_PAD_VOLUME, PROP_PAD_MUTE }; G_DEFINE_TYPE (GstAudioMixerPad, gst_audiomixer_pad, GST_TYPE_AUDIO_AGGREGATOR_PAD); static void gst_audiomixer_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object); switch (prop_id) { case PROP_PAD_VOLUME: g_value_set_double (value, pad->volume); break; case PROP_PAD_MUTE: g_value_set_boolean (value, pad->mute); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audiomixer_pad_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (object); switch (prop_id) { case PROP_PAD_VOLUME: GST_OBJECT_LOCK (pad); pad->volume = g_value_get_double (value); pad->volume_i8 = pad->volume * VOLUME_UNITY_INT8; pad->volume_i16 = pad->volume * VOLUME_UNITY_INT16; pad->volume_i32 = pad->volume * VOLUME_UNITY_INT32; GST_OBJECT_UNLOCK (pad); break; case PROP_PAD_MUTE: GST_OBJECT_LOCK (pad); pad->mute = g_value_get_boolean (value); GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audiomixer_pad_class_init (GstAudioMixerPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; gobject_class->set_property = gst_audiomixer_pad_set_property; gobject_class->get_property = gst_audiomixer_pad_get_property; g_object_class_install_property (gobject_class, PROP_PAD_VOLUME, g_param_spec_double ("volume", "Volume", "Volume of this pad", 0.0, 10.0, DEFAULT_PAD_VOLUME, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PAD_MUTE, g_param_spec_boolean ("mute", "Mute", "Mute this pad", DEFAULT_PAD_MUTE, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS)); } static void gst_audiomixer_pad_init (GstAudioMixerPad * pad) { pad->volume = DEFAULT_PAD_VOLUME; pad->mute = DEFAULT_PAD_MUTE; } enum { PROP_0, PROP_FILTER_CAPS }; /* elementfactory information */ #if G_BYTE_ORDER == G_LITTLE_ENDIAN #define CAPS \ GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \ ", layout = (string) { interleaved, non-interleaved }" #else #define CAPS \ GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \ ", layout = (string) { interleaved, non-interleaved }" #endif static GstStaticPadTemplate gst_audiomixer_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (CAPS) ); static GstStaticPadTemplate gst_audiomixer_sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS (CAPS) ); static void gst_audiomixer_child_proxy_init (gpointer g_iface, gpointer iface_data); #define gst_audiomixer_parent_class parent_class G_DEFINE_TYPE_WITH_CODE (GstAudioMixer, gst_audiomixer, GST_TYPE_AUDIO_AGGREGATOR, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY, gst_audiomixer_child_proxy_init)); static void gst_audiomixer_dispose (GObject * object); static void gst_audiomixer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audiomixer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad, GstCaps * caps); static GstPad *gst_audiomixer_request_new_pad (GstElement * element, GstPadTemplate * temp, const gchar * req_name, const GstCaps * caps); static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad); static gboolean gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg, GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset, GstBuffer * outbuf, guint out_offset, guint num_samples); /* we can only accept caps that we and downstream can handle. * if we have filtercaps set, use those to constrain the target caps. */ static GstCaps * gst_audiomixer_sink_getcaps (GstAggregator * agg, GstPad * pad, GstCaps * filter) { GstAudioAggregator *aagg; GstAudioMixer *audiomixer; GstCaps *result, *peercaps, *current_caps, *filter_caps; GstStructure *s; gint i, n; audiomixer = GST_AUDIO_MIXER (agg); aagg = GST_AUDIO_AGGREGATOR (agg); GST_OBJECT_LOCK (audiomixer); /* take filter */ if ((filter_caps = audiomixer->filter_caps)) { if (filter) filter_caps = gst_caps_intersect_full (filter, filter_caps, GST_CAPS_INTERSECT_FIRST); else gst_caps_ref (filter_caps); } else { filter_caps = filter ? gst_caps_ref (filter) : NULL; } GST_OBJECT_UNLOCK (audiomixer); if (filter_caps && gst_caps_is_empty (filter_caps)) { GST_WARNING_OBJECT (pad, "Empty filter caps"); return filter_caps; } /* get the downstream possible caps */ peercaps = gst_pad_peer_query_caps (agg->srcpad, filter_caps); /* get the allowed caps on this sinkpad */ GST_OBJECT_LOCK (audiomixer); current_caps = aagg->current_caps ? gst_caps_ref (aagg->current_caps) : NULL; if (current_caps == NULL) { current_caps = gst_pad_get_pad_template_caps (pad); if (!current_caps) current_caps = gst_caps_new_any (); } GST_OBJECT_UNLOCK (audiomixer); if (peercaps) { /* if the peer has caps, intersect */ GST_DEBUG_OBJECT (audiomixer, "intersecting peer and our caps"); result = gst_caps_intersect_full (peercaps, current_caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (peercaps); gst_caps_unref (current_caps); } else { /* the peer has no caps (or there is no peer), just use the allowed caps * of this sinkpad. */ /* restrict with filter-caps if any */ if (filter_caps) { GST_DEBUG_OBJECT (audiomixer, "no peer caps, using filtered caps"); result = gst_caps_intersect_full (filter_caps, current_caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (current_caps); } else { GST_DEBUG_OBJECT (audiomixer, "no peer caps, using our caps"); result = current_caps; } } result = gst_caps_make_writable (result); n = gst_caps_get_size (result); for (i = 0; i < n; i++) { GstStructure *sref; s = gst_caps_get_structure (result, i); sref = gst_structure_copy (s); gst_structure_set (sref, "channels", GST_TYPE_INT_RANGE, 0, 2, NULL); if (gst_structure_is_subset (s, sref)) { /* This field is irrelevant when in mono or stereo */ gst_structure_remove_field (s, "channel-mask"); } gst_structure_free (sref); } if (filter_caps) gst_caps_unref (filter_caps); GST_LOG_OBJECT (audiomixer, "getting caps on pad %p,%s to %" GST_PTR_FORMAT, pad, GST_PAD_NAME (pad), result); return result; } static gboolean gst_audiomixer_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad, GstQuery * query) { gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CAPS: { GstCaps *filter, *caps; gst_query_parse_caps (query, &filter); caps = gst_audiomixer_sink_getcaps (agg, GST_PAD (aggpad), filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } default: res = GST_AGGREGATOR_CLASS (parent_class)->sink_query (agg, aggpad, query); break; } return res; } /* the first caps we receive on any of the sinkpads will define the caps for all * the other sinkpads because we can only mix streams with the same caps. */ static gboolean gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad, GstCaps * orig_caps) { GstAggregator *agg = GST_AGGREGATOR (audiomixer); GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (audiomixer); GstCaps *caps; GstAudioInfo info; GstStructure *s; gint channels = 0; caps = gst_caps_copy (orig_caps); s = gst_caps_get_structure (caps, 0); if (gst_structure_get_int (s, "channels", &channels)) if (channels <= 2) gst_structure_remove_field (s, "channel-mask"); if (!gst_audio_info_from_caps (&info, caps)) goto invalid_format; if (channels == 1) { GstCaps *filter; GstCaps *downstream_caps; if (audiomixer->filter_caps) filter = gst_caps_intersect_full (caps, audiomixer->filter_caps, GST_CAPS_INTERSECT_FIRST); else filter = gst_caps_ref (caps); downstream_caps = gst_pad_peer_query_caps (agg->srcpad, filter); gst_caps_unref (filter); if (downstream_caps) { gst_caps_unref (caps); caps = downstream_caps; if (gst_caps_is_empty (caps)) { gst_caps_unref (caps); return FALSE; } caps = gst_caps_fixate (caps); } } GST_OBJECT_LOCK (audiomixer); /* don't allow reconfiguration for now; there's still a race between the * different upstream threads doing query_caps + accept_caps + sending * (possibly different) CAPS events, but there's not much we can do about * that, upstream needs to deal with it. */ if (aagg->current_caps != NULL) { if (gst_audio_info_is_equal (&info, &aagg->info)) { GST_OBJECT_UNLOCK (audiomixer); gst_caps_unref (caps); gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad), orig_caps); return TRUE; } else { GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but " "current caps are %" GST_PTR_FORMAT, caps, aagg->current_caps); GST_OBJECT_UNLOCK (audiomixer); gst_pad_push_event (pad, gst_event_new_reconfigure ()); gst_caps_unref (caps); return FALSE; } } else { gst_caps_replace (&aagg->current_caps, caps); aagg->info = info; gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (agg)); } GST_OBJECT_UNLOCK (audiomixer); gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad), orig_caps); GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps); gst_caps_unref (caps); return TRUE; /* ERRORS */ invalid_format: { gst_caps_unref (caps); GST_WARNING_OBJECT (audiomixer, "invalid format set as caps"); return FALSE; } } static GstFlowReturn gst_audiomixer_update_src_caps (GstAggregator * agg, GstCaps * caps, GstCaps ** ret) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); if (aagg->current_caps == NULL) return GST_AGGREGATOR_FLOW_NEED_DATA; *ret = gst_caps_ref (aagg->current_caps); return GST_FLOW_OK; } static gboolean gst_audiomixer_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad, GstEvent * event) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (agg); gboolean res = TRUE; GST_DEBUG_OBJECT (aggpad, "Got %s event on sink pad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; gst_event_parse_caps (event, &caps); res = gst_audiomixer_setcaps (audiomixer, GST_PAD_CAST (aggpad), caps); gst_event_unref (event); event = NULL; break; } default: break; } if (event != NULL) return GST_AGGREGATOR_CLASS (parent_class)->sink_event (agg, aggpad, event); return res; } static void gst_audiomixer_class_init (GstAudioMixerClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstElementClass *gstelement_class = (GstElementClass *) klass; GstAggregatorClass *agg_class = (GstAggregatorClass *) klass; GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass; gobject_class->set_property = gst_audiomixer_set_property; gobject_class->get_property = gst_audiomixer_get_property; gobject_class->dispose = gst_audiomixer_dispose; g_object_class_install_property (gobject_class, PROP_FILTER_CAPS, g_param_spec_boxed ("caps", "Target caps", "Set target format for mixing (NULL means ANY). " "Setting this property takes a reference to the supplied GstCaps " "object", GST_TYPE_CAPS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &gst_audiomixer_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_audiomixer_sink_template); gst_element_class_set_static_metadata (gstelement_class, "AudioMixer", "Generic/Audio", "Mixes multiple audio streams", "Sebastian Dröge "); gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_audiomixer_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_audiomixer_release_pad); agg_class->sinkpads_type = GST_TYPE_AUDIO_MIXER_PAD; agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query); agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event); agg_class->update_src_caps = GST_DEBUG_FUNCPTR (gst_audiomixer_update_src_caps); aagg_class->aggregate_one_buffer = gst_audiomixer_aggregate_one_buffer; } static void gst_audiomixer_init (GstAudioMixer * audiomixer) { audiomixer->filter_caps = NULL; } static void gst_audiomixer_dispose (GObject * object) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object); gst_caps_replace (&audiomixer->filter_caps, NULL); G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_audiomixer_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object); switch (prop_id) { case PROP_FILTER_CAPS:{ GstCaps *new_caps = NULL; GstCaps *old_caps; const GstCaps *new_caps_val = gst_value_get_caps (value); if (new_caps_val != NULL) { new_caps = (GstCaps *) new_caps_val; gst_caps_ref (new_caps); } GST_OBJECT_LOCK (audiomixer); old_caps = audiomixer->filter_caps; audiomixer->filter_caps = new_caps; GST_OBJECT_UNLOCK (audiomixer); if (old_caps) gst_caps_unref (old_caps); GST_DEBUG_OBJECT (audiomixer, "set new caps %" GST_PTR_FORMAT, new_caps); break; } default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audiomixer_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object); switch (prop_id) { case PROP_FILTER_CAPS: GST_OBJECT_LOCK (audiomixer); gst_value_set_caps (value, audiomixer->filter_caps); GST_OBJECT_UNLOCK (audiomixer); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstPad * gst_audiomixer_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * req_name, const GstCaps * caps) { GstAudioMixerPad *newpad; newpad = (GstAudioMixerPad *) GST_ELEMENT_CLASS (parent_class)->request_new_pad (element, templ, req_name, caps); if (newpad == NULL) goto could_not_create; gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (newpad), GST_OBJECT_NAME (newpad)); return GST_PAD_CAST (newpad); could_not_create: { GST_DEBUG_OBJECT (element, "could not create/add pad"); return NULL; } } static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad) { GstAudioMixer *audiomixer; audiomixer = GST_AUDIO_MIXER (element); GST_DEBUG_OBJECT (audiomixer, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad)); gst_child_proxy_child_removed (GST_CHILD_PROXY (audiomixer), G_OBJECT (pad), GST_OBJECT_NAME (pad)); GST_ELEMENT_CLASS (parent_class)->release_pad (element, pad); } /* Called with object lock and pad object lock held */ static gboolean gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg, GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset, GstBuffer * outbuf, guint out_offset, guint num_frames) { GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aaggpad); GstMapInfo inmap; GstMapInfo outmap; gint bpf; if (pad->mute || pad->volume < G_MINDOUBLE) { GST_DEBUG_OBJECT (pad, "Skipping muted pad"); return FALSE; } bpf = GST_AUDIO_INFO_BPF (&aagg->info); gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE); gst_buffer_map (inbuf, &inmap, GST_MAP_READ); GST_LOG_OBJECT (pad, "mixing %u bytes at offset %u from offset %u", num_frames * bpf, out_offset * bpf, in_offset * bpf); /* further buffers, need to add them */ if (pad->volume == 1.0) { switch (aagg->info.finfo->format) { case GST_AUDIO_FORMAT_U8: audiomixer_orc_add_u8 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_S8: audiomixer_orc_add_s8 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_U16: audiomixer_orc_add_u16 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_S16: audiomixer_orc_add_s16 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_U32: audiomixer_orc_add_u32 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_S32: audiomixer_orc_add_s32 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_F32: audiomixer_orc_add_f32 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_F64: audiomixer_orc_add_f64 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), num_frames * aagg->info.channels); break; default: g_assert_not_reached (); break; } } else { switch (aagg->info.finfo->format) { case GST_AUDIO_FORMAT_U8: audiomixer_orc_add_volume_u8 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), pad->volume_i8, num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_S8: audiomixer_orc_add_volume_s8 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), pad->volume_i8, num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_U16: audiomixer_orc_add_volume_u16 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), pad->volume_i16, num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_S16: audiomixer_orc_add_volume_s16 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), pad->volume_i16, num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_U32: audiomixer_orc_add_volume_u32 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), pad->volume_i32, num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_S32: audiomixer_orc_add_volume_s32 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), pad->volume_i32, num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_F32: audiomixer_orc_add_volume_f32 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), pad->volume, num_frames * aagg->info.channels); break; case GST_AUDIO_FORMAT_F64: audiomixer_orc_add_volume_f64 ((gpointer) (outmap.data + out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf), pad->volume, num_frames * aagg->info.channels); break; default: g_assert_not_reached (); break; } } gst_buffer_unmap (inbuf, &inmap); gst_buffer_unmap (outbuf, &outmap); return TRUE; } /* GstChildProxy implementation */ static GObject * gst_audiomixer_child_proxy_get_child_by_index (GstChildProxy * child_proxy, guint index) { GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy); GObject *obj = NULL; GST_OBJECT_LOCK (audiomixer); obj = g_list_nth_data (GST_ELEMENT_CAST (audiomixer)->sinkpads, index); if (obj) gst_object_ref (obj); GST_OBJECT_UNLOCK (audiomixer); return obj; } static guint gst_audiomixer_child_proxy_get_children_count (GstChildProxy * child_proxy) { guint count = 0; GstAudioMixer *audiomixer = GST_AUDIO_MIXER (child_proxy); GST_OBJECT_LOCK (audiomixer); count = GST_ELEMENT_CAST (audiomixer)->numsinkpads; GST_OBJECT_UNLOCK (audiomixer); GST_INFO_OBJECT (audiomixer, "Children Count: %d", count); return count; } static void gst_audiomixer_child_proxy_init (gpointer g_iface, gpointer iface_data) { GstChildProxyInterface *iface = g_iface; GST_INFO ("intializing child proxy interface"); iface->get_child_by_index = gst_audiomixer_child_proxy_get_child_by_index; iface->get_children_count = gst_audiomixer_child_proxy_get_children_count; } /* Empty liveadder alias with non-zero latency */ typedef GstAudioMixer GstLiveAdder; typedef GstAudioMixerClass GstLiveAdderClass; static GType gst_live_adder_get_type (void); #define GST_TYPE_LIVE_ADDER gst_live_adder_get_type () G_DEFINE_TYPE (GstLiveAdder, gst_live_adder, GST_TYPE_AUDIO_MIXER); enum { LIVEADDER_PROP_LATENCY = 1 }; static void gst_live_adder_init (GstLiveAdder * self) { } static void gst_live_adder_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { switch (prop_id) { case LIVEADDER_PROP_LATENCY: { GParamSpec *parent_spec = g_object_class_find_property (G_OBJECT_CLASS (gst_live_adder_parent_class), "latency"); GObjectClass *pspec_class = g_type_class_peek (parent_spec->owner_type); GValue v = { 0 }; g_value_init (&v, G_TYPE_INT64); g_value_set_int64 (&v, g_value_get_uint (value) * GST_MSECOND); G_OBJECT_CLASS (pspec_class)->set_property (object, parent_spec->param_id, &v, parent_spec); break; } default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_live_adder_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { switch (prop_id) { case LIVEADDER_PROP_LATENCY: { GParamSpec *parent_spec = g_object_class_find_property (G_OBJECT_CLASS (gst_live_adder_parent_class), "latency"); GObjectClass *pspec_class = g_type_class_peek (parent_spec->owner_type); GValue v = { 0 }; g_value_init (&v, G_TYPE_INT64); G_OBJECT_CLASS (pspec_class)->get_property (object, parent_spec->param_id, &v, parent_spec); g_value_set_uint (value, g_value_get_int64 (&v) / GST_MSECOND); break; } default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_live_adder_class_init (GstLiveAdderClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); gobject_class->set_property = gst_live_adder_set_property; gobject_class->get_property = gst_live_adder_get_property; g_object_class_install_property (gobject_class, LIVEADDER_PROP_LATENCY, g_param_spec_uint ("latency", "Buffer latency", "Additional latency in live mode to allow upstream " "to take longer to produce buffers for the current " "position (in milliseconds)", 0, G_MAXUINT, 30, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT)); } static gboolean plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "audiomixer", 0, "audio mixing element"); if (!gst_element_register (plugin, "audiomixer", GST_RANK_NONE, GST_TYPE_AUDIO_MIXER)) return FALSE; if (!gst_element_register (plugin, "liveadder", GST_RANK_NONE, GST_TYPE_LIVE_ADDER)) return FALSE; if (!gst_element_register (plugin, "audiointerleave", GST_RANK_NONE, GST_TYPE_AUDIO_INTERLEAVE)) return FALSE; return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, audiomixer, "Mixes multiple audio streams", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)