| /* GStreamer |
| * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> |
| * 2001 Thomas <thomas@apestaart.org> |
| * 2005,2006 Wim Taymans <wim@fluendo.com> |
| * 2013 Sebastian Dröge <sebastian@centricular.com> |
| * |
| * 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 |
| * |
| * 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: |
| * |
| * <itemizedlist> |
| * <listitem> |
| * "mute": Whether to mute the pad or not (#gboolean) |
| * </listitem> |
| * <listitem> |
| * "volume": The volume of the pad, between 0.0 and 10.0 (#gdouble) |
| * </listitem> |
| * </itemizedlist> |
| * |
| * <refsect2> |
| * <title>Example launch line</title> |
| * |[ |
| * gst-launch audiotestsrc freq=100 ! audiomixer name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix. |
| * ]| This pipeline produces two sine waves mixed together. |
| * </refsect2> |
| * |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include "config.h" |
| #endif |
| |
| #include "gstaudiomixer.h" |
| #include <gst/audio/audio.h> |
| #include <string.h> /* 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; |
| gboolean ret; |
| |
| 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; |
| } |
| } |
| GST_OBJECT_UNLOCK (audiomixer); |
| |
| ret = gst_audio_aggregator_set_src_caps (aagg, caps); |
| |
| if (ret) |
| 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 ret; |
| |
| /* ERRORS */ |
| invalid_format: |
| { |
| gst_caps_unref (caps); |
| GST_WARNING_OBJECT (audiomixer, "invalid format set as caps"); |
| return FALSE; |
| } |
| } |
| |
| 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_pad_template (gstelement_class, |
| gst_static_pad_template_get (&gst_audiomixer_src_template)); |
| gst_element_class_add_pad_template (gstelement_class, |
| gst_static_pad_template_get (&gst_audiomixer_sink_template)); |
| gst_element_class_set_static_metadata (gstelement_class, "AudioMixer", |
| "Generic/Audio", |
| "Mixes multiple audio streams", |
| "Sebastian Dröge <sebastian@centricular.com>"); |
| |
| 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); |
| |
| 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; |
| } |
| |
| 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, "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) |