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gstreamer/gst/audiomixer/gstaudiomixer.c

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/* 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-1.0 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_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 <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;
}
/* 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)
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