/* 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
* @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
* and also handles live inputs such as capture sources or RTP properly.
*
* The audiomixer element can accept any sort of raw audio data, it will
* be converted to the target format if necessary, with the exception
* of the sample rate, which has to be identical to either what downstream
* expects, or the sample rate of the first configured pad. Use a capsfilter
* after the audiomixer element if you want to precisely control the format
* that comes out of the audiomixer, which supports changing the format of
* its output while playing.
*
* If you want to control the manner in which incoming data gets converted,
* see the #GstAudioAggregatorPad:converter-config property, which will let
* you for example change the way in which channels may get remapped.
*
* 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 <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_CONVERT_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
};
/* These are the formats we can mix natively */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define CAPS \
GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \
", layout = interleaved"
#else
#define CAPS \
GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \
", layout = interleaved"
#endif
static GstStaticPadTemplate gst_audiomixer_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CAPS)
);
#define SINK_CAPS \
GST_STATIC_CAPS (GST_AUDIO_CAPS_MAKE (GST_AUDIO_FORMATS_ALL) \
", layout=interleaved")
static GstStaticPadTemplate gst_audiomixer_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
SINK_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 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);
static void
gst_audiomixer_class_init (GstAudioMixerClass * klass)
{
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass;
gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
&gst_audiomixer_src_template, GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD);
gst_element_class_add_static_pad_template_with_gtype (gstelement_class,
&gst_audiomixer_sink_template, GST_TYPE_AUDIO_MIXER_PAD);
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);
aagg_class->aggregate_one_buffer = gst_audiomixer_aggregate_one_buffer;
}
static void
gst_audiomixer_init (GstAudioMixer * audiomixer)
{
}
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);
}
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;
GstAggregator *agg = GST_AGGREGATOR (aagg);
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
GST_OBJECT_LOCK (aagg);
GST_OBJECT_LOCK (aaggpad);
if (pad->mute || pad->volume < G_MINDOUBLE) {
GST_DEBUG_OBJECT (pad, "Skipping muted pad");
GST_OBJECT_UNLOCK (aaggpad);
GST_OBJECT_UNLOCK (aagg);
return FALSE;
}
bpf = GST_AUDIO_INFO_BPF (&srcpad->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 (srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->info.channels);
break;
default:
g_assert_not_reached ();
break;
}
} else {
switch (srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->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 * srcpad->info.channels);
break;
default:
g_assert_not_reached ();
break;
}
}
gst_buffer_unmap (inbuf, &inmap);
gst_buffer_unmap (outbuf, &outmap);
GST_OBJECT_UNLOCK (aaggpad);
GST_OBJECT_UNLOCK (aagg);
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_UINT64);
g_value_set_uint64 (&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_UINT64);
G_OBJECT_CLASS (pspec_class)->get_property (object,
parent_spec->param_id, &v, parent_spec);
g_value_set_uint (value, g_value_get_uint64 (&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)