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Diffstat (limited to 'gst/audiomixer/gstaudiomixer.c')
| -rw-r--r-- | gst/audiomixer/gstaudiomixer.c | 577 |
1 files changed, 577 insertions, 0 deletions
diff --git a/gst/audiomixer/gstaudiomixer.c b/gst/audiomixer/gstaudiomixer.c new file mode 100644 index 000000000..a0f569010 --- /dev/null +++ b/gst/audiomixer/gstaudiomixer.c @@ -0,0 +1,577 @@ +/* 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 (gstelement_class, + &gst_audiomixer_src_template); + 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; + + 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 (&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); + + 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) |