diff --git a/Makefile.am b/Makefile.am index 994cc3942a..08d3ebe08b 100644 --- a/Makefile.am +++ b/Makefile.am @@ -63,6 +63,7 @@ CRUFT_FILES = \ $(top_builddir)/ext/qt/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/aacparse/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/amrparse/.libs/*.{so,dll,DLL,dylib} \ + $(top_builddir)/gst/audiomixer/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/audioparsers/.libs/*.{so,dll,DLL,dylib} \ $(top_builddir)/gst/camerabin2/.libs/libgstcamerabin2.so \ $(top_builddir)/gst/flacparse/.libs/*.{so,dll,DLL,dylib} \ @@ -105,6 +106,7 @@ CRUFT_DIRS = \ $(top_srcdir)/docs/plugins/tmpl \ $(top_srcdir)/gst/aacparse \ $(top_srcdir)/gst/amrparse \ + $(top_srcdir)/gst/audiomixer \ $(top_srcdir)/gst/camerabin \ $(top_srcdir)/gst/dataurisrc \ $(top_srcdir)/gst/flacparse \ diff --git a/configure.ac b/configure.ac index c6b482a66e..ceb89af003 100644 --- a/configure.ac +++ b/configure.ac @@ -424,7 +424,6 @@ AG_GST_CHECK_PLUGIN(videoframe_audiolevel) AG_GST_CHECK_PLUGIN(asfmux) AG_GST_CHECK_PLUGIN(audiobuffersplit) AG_GST_CHECK_PLUGIN(audiofxbad) -AG_GST_CHECK_PLUGIN(audiomixer) AG_GST_CHECK_PLUGIN(audiomixmatrix) AG_GST_CHECK_PLUGIN(compositor) AG_GST_CHECK_PLUGIN(audiovisualizers) @@ -2486,7 +2485,6 @@ gst/videoframe_audiolevel/Makefile gst/asfmux/Makefile gst/audiobuffersplit/Makefile gst/audiofxbad/Makefile -gst/audiomixer/Makefile gst/audiomixmatrix/Makefile gst/audiovisualizers/Makefile gst/autoconvert/Makefile diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml index db624a1d2f..a590060c17 100644 --- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml @@ -21,8 +21,6 @@ - - @@ -118,7 +116,6 @@ gst-plugins-bad Plugins - diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt index 8e98c9c96c..c1bae1d63e 100644 --- a/docs/plugins/gst-plugins-bad-plugins-sections.txt +++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt @@ -190,38 +190,6 @@ GST_TYPE_AUDIO_CHANNEL_MIX gst_audio_channel_mix_get_type -
-element-audiointerleave -audiointerleave -GstAudioInterleave - -GstAudioInterleaveClass -GST_AUDIO_INTERLEAVE -GST_AUDIO_INTERLEAVE_CAST -GST_IS_AUDIO_INTERLEAVE -GST_AUDIO_INTERLEAVE_CLASS -GST_IS_AUDIO_INTERLEAVE_CLASS -GST_TYPE_AUDIO_INTERLEAVE - -gst_audio_interleave_get_type -
- -
-element-audiomixer -audiomixer -GstAudioMixer - -GstAudioMixerClass -GST_AUDIO_MIXER -GST_AUDIO_MIXER_CAST -GST_IS_AUDIO_MIXER -GST_AUDIO_MIXER_CLASS -GST_IS_AUDIO_MIXER_CLASS -GST_TYPE_AUDIO_MIXER - -gst_audio_mixer_get_type -
-
element-audiomixmatrix audiomixmatrix diff --git a/docs/plugins/inspect/plugin-audiomixer.xml b/docs/plugins/inspect/plugin-audiomixer.xml deleted file mode 100644 index 9d0b593af5..0000000000 --- a/docs/plugins/inspect/plugin-audiomixer.xml +++ /dev/null @@ -1,76 +0,0 @@ - - audiomixer - Mixes multiple audio streams - ../../gst/audiomixer/.libs/libgstaudiomixer.so - libgstaudiomixer.so - 1.13.0.1 - LGPL - gst-plugins-bad - GStreamer Bad Plug-ins git - Unknown package origin - - - audiointerleave - AudioInterleave - Generic/Audio - Mixes multiple audio streams - Olivier Crete <olivier.crete@collabora.com> - - - sink_%u - sink - request -
audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)1, format=(string){ S8, U8, S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE }, layout=(string){ non-interleaved, interleaved }
- - - src - source - always -
audio/x-raw, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], format=(string){ S8, U8, S16LE, S16BE, U16LE, U16BE, S24_32LE, S24_32BE, U24_32LE, U24_32BE, S32LE, S32BE, U32LE, U32BE, S24LE, S24BE, U24LE, U24BE, S20LE, S20BE, U20LE, U20BE, S18LE, S18BE, U18LE, U18BE, F32LE, F32BE, F64LE, F64BE }, layout=(string)interleaved
- - - - - audiomixer - AudioMixer - Generic/Audio - Mixes multiple audio streams - Sebastian Dröge <sebastian@centricular.com> - - - sink_%u - sink - request -
audio/x-raw, format=(string){ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string){ interleaved, non-interleaved }
- - - src - source - always -
audio/x-raw, format=(string){ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string){ interleaved, non-interleaved }
- - - - - liveadder - AudioMixer - Generic/Audio - Mixes multiple audio streams - Sebastian Dröge <sebastian@centricular.com> - - - sink_%u - sink - request -
audio/x-raw, format=(string){ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string){ interleaved, non-interleaved }
- - - src - source - always -
audio/x-raw, format=(string){ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], layout=(string){ interleaved, non-interleaved }
- - - - - \ No newline at end of file diff --git a/gst/audiomixer/Makefile.am b/gst/audiomixer/Makefile.am deleted file mode 100644 index f1a4d73953..0000000000 --- a/gst/audiomixer/Makefile.am +++ /dev/null @@ -1,21 +0,0 @@ -plugin_LTLIBRARIES = libgstaudiomixer.la - -ORC_SOURCE=gstaudiomixerorc -include $(top_srcdir)/common/orc.mak - - -libgstaudiomixer_la_SOURCES = gstaudiomixer.c gstaudiointerleave.c -nodist_libgstaudiomixer_la_SOURCES = $(ORC_NODIST_SOURCES) -libgstaudiomixer_la_CFLAGS = \ - -I$(top_srcdir)/gst-libs \ - -I$(top_builddir)/gst-libs \ - $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \ - $(GST_CFLAGS) $(ORC_CFLAGS) -libgstaudiomixer_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -libgstaudiomixer_la_LIBADD = \ - $(top_builddir)/gst-libs/gst/audio/libgstbadaudio-$(GST_API_VERSION).la \ - $(GST_PLUGINS_BASE_LIBS) -lgstaudio-@GST_API_VERSION@ \ - $(GST_BASE_LIBS) $(GST_LIBS) $(ORC_LIBS) - -noinst_HEADERS = gstaudiomixer.h gstaudiointerleave.h - diff --git a/gst/audiomixer/gstaudiointerleave.c b/gst/audiomixer/gstaudiointerleave.c deleted file mode 100644 index 90ec363ea9..0000000000 --- a/gst/audiomixer/gstaudiointerleave.c +++ /dev/null @@ -1,902 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen - * 2000 Wim Taymans - * 2005 Wim Taymans - * 2007 Andy Wingo - * 2008 Sebastian Dröge - * 2014 Collabora - * Olivier Crete - * - * gstaudiointerleave.c: audiointerleave 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-audiointerleave - * @title: audiointerleave - * - */ - -/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray - * with newer GLib versions (>= 2.31.0) */ -#define GLIB_DISABLE_DEPRECATION_WARNINGS - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "gstaudiointerleave.h" -#include - -#include - -#define GST_CAT_DEFAULT gst_audio_interleave_debug -GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); - -enum -{ - PROP_PAD_0, - PROP_PAD_CHANNEL -}; - -G_DEFINE_TYPE (GstAudioInterleavePad, gst_audio_interleave_pad, - GST_TYPE_AUDIO_AGGREGATOR_PAD); - -static void -gst_audio_interleave_pad_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAudioInterleavePad *pad = GST_AUDIO_INTERLEAVE_PAD (object); - - switch (prop_id) { - case PROP_PAD_CHANNEL: - g_value_set_uint (value, pad->channel); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - - -static void -gst_audio_interleave_pad_class_init (GstAudioInterleavePadClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - - gobject_class->get_property = gst_audio_interleave_pad_get_property; - - g_object_class_install_property (gobject_class, - PROP_PAD_CHANNEL, - g_param_spec_uint ("channel", - "Channel number", - "Number of the channel of this pad in the output", 0, G_MAXUINT, 0, - G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); -} - -static void -gst_audio_interleave_pad_init (GstAudioInterleavePad * pad) -{ -} - -enum -{ - PROP_0, - PROP_CHANNEL_POSITIONS, - PROP_CHANNEL_POSITIONS_FROM_INPUT -}; - -/* 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_audio_interleave_sink_template = -GST_STATIC_PAD_TEMPLATE ("sink_%u", - GST_PAD_SINK, - GST_PAD_REQUEST, - GST_STATIC_CAPS ("audio/x-raw, " - "rate = (int) [ 1, MAX ], " - "channels = (int) 1, " - "format = (string) " GST_AUDIO_FORMATS_ALL ", " - "layout = (string) {non-interleaved, interleaved}") - ); - -static GstStaticPadTemplate gst_audio_interleave_src_template = -GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "format = (string) " GST_AUDIO_FORMATS_ALL ", " - "layout = (string) interleaved") - ); - -static void gst_audio_interleave_child_proxy_init (gpointer g_iface, - gpointer iface_data); - -#define gst_audio_interleave_parent_class parent_class -G_DEFINE_TYPE_WITH_CODE (GstAudioInterleave, gst_audio_interleave, - GST_TYPE_AUDIO_AGGREGATOR, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY, - gst_audio_interleave_child_proxy_init)); - -static void gst_audio_interleave_finalize (GObject * object); -static void gst_audio_interleave_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_audio_interleave_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static gboolean gst_audio_interleave_setcaps (GstAudioInterleave * self, - GstPad * pad, GstCaps * caps); -static GstPad *gst_audio_interleave_request_new_pad (GstElement * element, - GstPadTemplate * temp, const gchar * req_name, const GstCaps * caps); -static void gst_audio_interleave_release_pad (GstElement * element, - GstPad * pad); - -static gboolean gst_audio_interleave_stop (GstAggregator * agg); - -static gboolean -gst_audio_interleave_aggregate_one_buffer (GstAudioAggregator * aagg, - GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset, - GstBuffer * outbuf, guint out_offset, guint num_samples); - - -static void -__remove_channels (GstCaps * caps) -{ - GstStructure *s; - gint i, size; - - size = gst_caps_get_size (caps); - for (i = 0; i < size; i++) { - s = gst_caps_get_structure (caps, i); - gst_structure_remove_field (s, "channel-mask"); - gst_structure_remove_field (s, "channels"); - } -} - -static void -__set_channels (GstCaps * caps, gint channels) -{ - GstStructure *s; - gint i, size; - - size = gst_caps_get_size (caps); - for (i = 0; i < size; i++) { - s = gst_caps_get_structure (caps, i); - if (channels > 0) - gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL); - else - gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); - } -} - -/* 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_audio_interleave_sink_getcaps (GstAggregator * agg, GstPad * pad, - GstCaps * filter) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); - GstCaps *result = NULL, *peercaps, *sinkcaps; - - GST_OBJECT_LOCK (self); - /* If we already have caps on one of the sink pads return them */ - if (self->sinkcaps) - result = gst_caps_copy (self->sinkcaps); - GST_OBJECT_UNLOCK (self); - - if (result == NULL) { - /* get the downstream possible caps */ - peercaps = gst_pad_peer_query_caps (agg->srcpad, NULL); - - /* get the allowed caps on this sinkpad */ - sinkcaps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); - __remove_channels (sinkcaps); - if (peercaps) { - peercaps = gst_caps_make_writable (peercaps); - __remove_channels (peercaps); - /* if the peer has caps, intersect */ - GST_DEBUG_OBJECT (pad, "intersecting peer and template caps"); - result = gst_caps_intersect (peercaps, sinkcaps); - gst_caps_unref (peercaps); - gst_caps_unref (sinkcaps); - } else { - /* the peer has no caps (or there is no peer), just use the allowed caps - * of this sinkpad. */ - GST_DEBUG_OBJECT (pad, "no peer caps, using sinkcaps"); - result = sinkcaps; - } - __set_channels (result, 1); - } - - if (filter != NULL) { - GstCaps *caps = result; - - GST_LOG_OBJECT (pad, "intersecting filter caps %" GST_PTR_FORMAT " with " - "preliminary result %" GST_PTR_FORMAT, filter, caps); - - result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); - gst_caps_unref (caps); - } - - GST_DEBUG_OBJECT (pad, "Returning caps %" GST_PTR_FORMAT, result); - - return result; -} - -static gboolean -gst_audio_interleave_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_audio_interleave_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; -} - -static gint -compare_positions (gconstpointer a, gconstpointer b, gpointer user_data) -{ - const gint i = *(const gint *) a; - const gint j = *(const gint *) b; - const gint *pos = (const gint *) user_data; - - if (pos[i] < pos[j]) - return -1; - else if (pos[i] > pos[j]) - return 1; - else - return 0; -} - -static gboolean -gst_audio_interleave_channel_positions_to_mask (GValueArray * positions, - gint default_ordering_map[64], guint64 * mask) -{ - gint i; - guint channels; - GstAudioChannelPosition *pos; - gboolean ret; - - channels = positions->n_values; - pos = g_new (GstAudioChannelPosition, channels); - - for (i = 0; i < channels; i++) { - GValue *val; - - val = g_value_array_get_nth (positions, i); - pos[i] = g_value_get_enum (val); - } - - /* sort the default ordering map according to the position order */ - for (i = 0; i < channels; i++) { - default_ordering_map[i] = i; - } - g_qsort_with_data (default_ordering_map, channels, - sizeof (*default_ordering_map), compare_positions, pos); - - ret = gst_audio_channel_positions_to_mask (pos, channels, FALSE, mask); - g_free (pos); - - return ret; -} - - -/* Must be called with the object lock held */ - -static guint64 -gst_audio_interleave_get_channel_mask (GstAudioInterleave * self) -{ - guint64 channel_mask = 0; - - if (self->channels <= 64 && - self->channel_positions != NULL && - self->channels == self->channel_positions->n_values) { - if (!gst_audio_interleave_channel_positions_to_mask - (self->channel_positions, self->default_channels_ordering_map, - &channel_mask)) { - GST_WARNING_OBJECT (self, "Invalid channel positions, using NONE"); - channel_mask = 0; - } - } else if (self->channels <= 64) { - GST_WARNING_OBJECT (self, "Using NONE channel positions"); - } - - return channel_mask; -} - - -#define MAKE_FUNC(type) \ -static void interleave_##type (guint##type *out, guint##type *in, \ - guint stride, guint nframes) \ -{ \ - gint i; \ - \ - for (i = 0; i < nframes; i++) { \ - *out = in[i]; \ - out += stride; \ - } \ -} - -MAKE_FUNC (8); -MAKE_FUNC (16); -MAKE_FUNC (32); -MAKE_FUNC (64); - -static void -interleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes) -{ - gint i; - - for (i = 0; i < nframes; i++) { - memcpy (out, in, 3); - out += stride * 3; - in += 3; - } -} - -static void -gst_audio_interleave_set_process_function (GstAudioInterleave * self, - GstAudioInfo * info) -{ - switch (GST_AUDIO_INFO_WIDTH (info)) { - case 8: - self->func = (GstInterleaveFunc) interleave_8; - break; - case 16: - self->func = (GstInterleaveFunc) interleave_16; - break; - case 24: - self->func = (GstInterleaveFunc) interleave_24; - break; - case 32: - self->func = (GstInterleaveFunc) interleave_32; - break; - case 64: - self->func = (GstInterleaveFunc) interleave_64; - break; - default: - g_assert_not_reached (); - break; - } -} - - -/* 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_audio_interleave_setcaps (GstAudioInterleave * self, GstPad * pad, - GstCaps * caps) -{ - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self); - GstAudioInfo info; - GValue *val; - guint channel; - gboolean new = FALSE; - - if (!gst_audio_info_from_caps (&info, caps)) - goto invalid_format; - - GST_OBJECT_LOCK (self); - if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps)) - goto cannot_change_caps; - - if (!self->sinkcaps) { - GstCaps *sinkcaps = gst_caps_copy (caps); - GstStructure *s = gst_caps_get_structure (sinkcaps, 0); - - gst_structure_remove_field (s, "channel-mask"); - - GST_DEBUG_OBJECT (self, "setting sinkcaps %" GST_PTR_FORMAT, sinkcaps); - - gst_caps_replace (&self->sinkcaps, sinkcaps); - gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (aagg)); - - gst_caps_unref (sinkcaps); - new = TRUE; - } - - if (self->channel_positions_from_input - && GST_AUDIO_INFO_CHANNELS (&info) == 1) { - channel = GST_AUDIO_INTERLEAVE_PAD (pad)->channel; - val = g_value_array_get_nth (self->input_channel_positions, channel); - g_value_set_enum (val, GST_AUDIO_INFO_POSITION (&info, 0)); - } - GST_OBJECT_UNLOCK (self); - - gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad), - caps); - - if (!new) - return TRUE; - - GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps); - - return TRUE; - - /* ERRORS */ -invalid_format: - { - GST_WARNING_OBJECT (self, "invalid format set as caps: %" GST_PTR_FORMAT, - caps); - return FALSE; - } -cannot_change_caps: - { - GST_OBJECT_UNLOCK (self); - GST_WARNING_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't " - "change", self->sinkcaps); - return FALSE; - } -} - -static gboolean -gst_audio_interleave_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad, - GstEvent * event) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (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_audio_interleave_setcaps (self, 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 GstFlowReturn -gst_audio_interleave_update_src_caps (GstAggregator * agg, GstCaps * caps, - GstCaps ** ret) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); - GstStructure *s; - - /* This means that either no caps have been set on the sink pad (if - * sinkcaps is NULL) or that there is no sink pad (if channels == 0). - */ - GST_OBJECT_LOCK (self); - if (self->sinkcaps == NULL || self->channels == 0) { - GST_OBJECT_UNLOCK (self); - return GST_FLOW_NOT_NEGOTIATED; - } - - *ret = gst_caps_copy (self->sinkcaps); - s = gst_caps_get_structure (*ret, 0); - - gst_structure_set (s, "channels", G_TYPE_INT, self->channels, "layout", - G_TYPE_STRING, "interleaved", "channel-mask", GST_TYPE_BITMASK, - gst_audio_interleave_get_channel_mask (self), NULL); - - GST_OBJECT_UNLOCK (self); - - return GST_FLOW_OK; -} - -static gboolean -gst_audio_interleave_negotiated_src_caps (GstAggregator * agg, GstCaps * caps) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (self); - - if (!GST_AGGREGATOR_CLASS (parent_class)->negotiated_src_caps (agg, caps)) - return FALSE; - - gst_audio_interleave_set_process_function (self, &aagg->info); - - return TRUE; -} - -static void -gst_audio_interleave_class_init (GstAudioInterleaveClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - GstElementClass *gstelement_class = (GstElementClass *) klass; - GstAggregatorClass *agg_class = (GstAggregatorClass *) klass; - GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass; - - GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "audiointerleave", 0, - "audio interleaving element"); - - gobject_class->set_property = gst_audio_interleave_set_property; - gobject_class->get_property = gst_audio_interleave_get_property; - gobject_class->finalize = gst_audio_interleave_finalize; - - gst_element_class_add_static_pad_template (gstelement_class, - &gst_audio_interleave_src_template); - gst_element_class_add_static_pad_template_with_gtype (gstelement_class, - &gst_audio_interleave_sink_template, GST_TYPE_AUDIO_INTERLEAVE_PAD); - gst_element_class_set_static_metadata (gstelement_class, "AudioInterleave", - "Generic/Audio", "Mixes multiple audio streams", - "Olivier Crete "); - - gstelement_class->request_new_pad = - GST_DEBUG_FUNCPTR (gst_audio_interleave_request_new_pad); - gstelement_class->release_pad = - GST_DEBUG_FUNCPTR (gst_audio_interleave_release_pad); - - agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audio_interleave_sink_query); - agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audio_interleave_sink_event); - agg_class->stop = gst_audio_interleave_stop; - agg_class->update_src_caps = gst_audio_interleave_update_src_caps; - agg_class->negotiated_src_caps = gst_audio_interleave_negotiated_src_caps; - - aagg_class->aggregate_one_buffer = gst_audio_interleave_aggregate_one_buffer; - aagg_class->convert_buffer = NULL; - - /** - * GstInterleave:channel-positions - * - * Channel positions: This property controls the channel positions - * that are used on the src caps. The number of elements should be - * the same as the number of sink pads and the array should contain - * a valid list of channel positions. The n-th element of the array - * is the position of the n-th sink pad. - * - * These channel positions will only be used if they're valid and the - * number of elements is the same as the number of channels. If this - * is not given a NONE layout will be used. - * - */ - g_object_class_install_property (gobject_class, PROP_CHANNEL_POSITIONS, - g_param_spec_value_array ("channel-positions", "Channel positions", - "Channel positions used on the output", - g_param_spec_enum ("channel-position", "Channel position", - "Channel position of the n-th input", - GST_TYPE_AUDIO_CHANNEL_POSITION, - GST_AUDIO_CHANNEL_POSITION_NONE, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS), - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - /** - * GstInterleave:channel-positions-from-input - * - * Channel positions from input: If this property is set to %TRUE the channel - * positions will be taken from the input caps if valid channel positions for - * the output can be constructed from them. If this is set to %TRUE setting the - * channel-positions property overwrites this property again. - * - */ - g_object_class_install_property (gobject_class, - PROP_CHANNEL_POSITIONS_FROM_INPUT, - g_param_spec_boolean ("channel-positions-from-input", - "Channel positions from input", - "Take channel positions from the input", TRUE, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); -} - -static void -gst_audio_interleave_init (GstAudioInterleave * self) -{ - self->input_channel_positions = g_value_array_new (0); - self->channel_positions_from_input = TRUE; - self->channel_positions = self->input_channel_positions; -} - -static void -gst_audio_interleave_finalize (GObject * object) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object); - - if (self->channel_positions - && self->channel_positions != self->input_channel_positions) { - g_value_array_free (self->channel_positions); - self->channel_positions = NULL; - } - - if (self->input_channel_positions) { - g_value_array_free (self->input_channel_positions); - self->input_channel_positions = NULL; - } - - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static void -gst_audio_interleave_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object); - - switch (prop_id) { - case PROP_CHANNEL_POSITIONS: - g_return_if_fail ( - ((GValueArray *) g_value_get_boxed (value))->n_values > 0); - - if (self->channel_positions && - self->channel_positions != self->input_channel_positions) - g_value_array_free (self->channel_positions); - - self->channel_positions = g_value_dup_boxed (value); - self->channel_positions_from_input = FALSE; - break; - case PROP_CHANNEL_POSITIONS_FROM_INPUT: - self->channel_positions_from_input = g_value_get_boolean (value); - - if (self->channel_positions_from_input) { - if (self->channel_positions && - self->channel_positions != self->input_channel_positions) - g_value_array_free (self->channel_positions); - self->channel_positions = self->input_channel_positions; - } - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_audio_interleave_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (object); - - switch (prop_id) { - case PROP_CHANNEL_POSITIONS: - g_value_set_boxed (value, self->channel_positions); - break; - case PROP_CHANNEL_POSITIONS_FROM_INPUT: - g_value_set_boolean (value, self->channel_positions_from_input); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static gboolean -gst_audio_interleave_stop (GstAggregator * agg) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (agg); - - if (!GST_AGGREGATOR_CLASS (parent_class)->stop (agg)) - return FALSE; - - gst_caps_replace (&self->sinkcaps, NULL); - - return TRUE; -} - -static GstPad * -gst_audio_interleave_request_new_pad (GstElement * element, - GstPadTemplate * templ, const gchar * req_name, const GstCaps * caps) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (element); - GstAudioInterleavePad *newpad; - gchar *pad_name; - gint channel, padnumber; - GValue val = { 0, }; - - /* FIXME: We ignore req_name, this is evil! */ - - GST_OBJECT_LOCK (self); - padnumber = g_atomic_int_add (&self->padcounter, 1); - channel = self->channels++; - if (!self->channel_positions_from_input) - channel = padnumber; - GST_OBJECT_UNLOCK (self); - - pad_name = g_strdup_printf ("sink_%u", padnumber); - newpad = (GstAudioInterleavePad *) - GST_ELEMENT_CLASS (parent_class)->request_new_pad (element, - templ, pad_name, caps); - g_free (pad_name); - if (newpad == NULL) - goto could_not_create; - - newpad->channel = channel; - gst_pad_use_fixed_caps (GST_PAD (newpad)); - - gst_child_proxy_child_added (GST_CHILD_PROXY (element), G_OBJECT (newpad), - GST_OBJECT_NAME (newpad)); - - - g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION); - g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_NONE); - self->input_channel_positions = - g_value_array_append (self->input_channel_positions, &val); - g_value_unset (&val); - - /* Update the src caps if we already have them */ - gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (self)); - - return GST_PAD_CAST (newpad); - -could_not_create: - { - GST_DEBUG_OBJECT (element, "could not create/add pad"); - return NULL; - } -} - -static void -gst_audio_interleave_release_pad (GstElement * element, GstPad * pad) -{ - GstAudioInterleave *self; - gint position; - GList *l; - - self = GST_AUDIO_INTERLEAVE (element); - - /* Take lock to make sure we're not changing this when processing buffers */ - GST_OBJECT_LOCK (self); - - self->channels--; - - position = GST_AUDIO_INTERLEAVE_PAD (pad)->channel; - g_value_array_remove (self->input_channel_positions, position); - - /* Update channel numbers */ - /* Taken above, GST_OBJECT_LOCK (self); */ - for (l = GST_ELEMENT_CAST (self)->sinkpads; l != NULL; l = l->next) { - GstAudioInterleavePad *ipad = GST_AUDIO_INTERLEAVE_PAD (l->data); - - if (GST_AUDIO_INTERLEAVE_PAD (pad)->channel < ipad->channel) - ipad->channel--; - } - - gst_pad_mark_reconfigure (GST_AGGREGATOR_SRC_PAD (self)); - GST_OBJECT_UNLOCK (self); - - - GST_DEBUG_OBJECT (self, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad)); - - gst_child_proxy_child_removed (GST_CHILD_PROXY (self), 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_audio_interleave_aggregate_one_buffer (GstAudioAggregator * aagg, - GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset, - GstBuffer * outbuf, guint out_offset, guint num_frames) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (aagg); - GstAudioInterleavePad *pad = GST_AUDIO_INTERLEAVE_PAD (aaggpad); - GstMapInfo inmap; - GstMapInfo outmap; - gint out_width, in_bpf, out_bpf, out_channels, channel; - guint8 *outdata; - - GST_OBJECT_LOCK (aagg); - GST_OBJECT_LOCK (aaggpad); - - out_width = GST_AUDIO_INFO_WIDTH (&aagg->info) / 8; - in_bpf = GST_AUDIO_INFO_BPF (&aaggpad->info); - out_bpf = GST_AUDIO_INFO_BPF (&aagg->info); - out_channels = GST_AUDIO_INFO_CHANNELS (&aagg->info); - - gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE); - gst_buffer_map (inbuf, &inmap, GST_MAP_READ); - GST_LOG_OBJECT (pad, "interleaves %u frames on channel %d/%d at offset %u" - " from offset %u", num_frames, pad->channel, out_channels, - out_offset * out_bpf, in_offset * in_bpf); - - if (self->channels > 64) { - channel = pad->channel; - } else { - channel = self->default_channels_ordering_map[pad->channel]; - } - - outdata = outmap.data + (out_offset * out_bpf) + (out_width * channel); - - - self->func (outdata, inmap.data + (in_offset * in_bpf), out_channels, - num_frames); - - - 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_audio_interleave_child_proxy_get_child_by_index (GstChildProxy * - child_proxy, guint index) -{ - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (child_proxy); - GObject *obj = NULL; - - GST_OBJECT_LOCK (self); - obj = g_list_nth_data (GST_ELEMENT_CAST (self)->sinkpads, index); - if (obj) - gst_object_ref (obj); - GST_OBJECT_UNLOCK (self); - - return obj; -} - -static guint -gst_audio_interleave_child_proxy_get_children_count (GstChildProxy * - child_proxy) -{ - guint count = 0; - GstAudioInterleave *self = GST_AUDIO_INTERLEAVE (child_proxy); - - GST_OBJECT_LOCK (self); - count = GST_ELEMENT_CAST (self)->numsinkpads; - GST_OBJECT_UNLOCK (self); - GST_INFO_OBJECT (self, "Children Count: %d", count); - - return count; -} - -static void -gst_audio_interleave_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_audio_interleave_child_proxy_get_child_by_index; - iface->get_children_count = - gst_audio_interleave_child_proxy_get_children_count; -} diff --git a/gst/audiomixer/gstaudiointerleave.h b/gst/audiomixer/gstaudiointerleave.h deleted file mode 100644 index bf46f4a505..0000000000 --- a/gst/audiomixer/gstaudiointerleave.h +++ /dev/null @@ -1,100 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen - * 2000 Wim Taymans - * Copyright (C) 2013 Sebastian Dröge - * - * gstaudiointerleave.h: Header for audiointerleave element - * - * 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. - */ - -#ifndef __GST_AUDIO_INTERLEAVE_H__ -#define __GST_AUDIO_INTERLEAVE_H__ - -#include -#include - -#include - -G_BEGIN_DECLS - -#define GST_TYPE_AUDIO_INTERLEAVE (gst_audio_interleave_get_type()) -#define GST_AUDIO_INTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_INTERLEAVE,GstAudioInterleave)) -#define GST_IS_AUDIO_INTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_INTERLEAVE)) -#define GST_AUDIO_INTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_INTERLEAVE,GstAudioInterleaveClass)) -#define GST_IS_AUDIO_INTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_INTERLEAVE)) -#define GST_AUDIO_INTERLEAVE_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_INTERLEAVE,GstAudioInterleaveClass)) - -typedef struct _GstAudioInterleave GstAudioInterleave; -typedef struct _GstAudioInterleaveClass GstAudioInterleaveClass; - -typedef struct _GstAudioInterleavePad GstAudioInterleavePad; -typedef struct _GstAudioInterleavePadClass GstAudioInterleavePadClass; - -typedef void (*GstInterleaveFunc) (gpointer out, gpointer in, guint stride, - guint nframes); - -/** - * GstAudioInterleave: - * - * The GstAudioInterleave object structure. - */ -struct _GstAudioInterleave { - GstAudioAggregator parent; - - gint padcounter; - guint channels; /* object lock */ - - GstCaps *sinkcaps; - - GValueArray *channel_positions; - GValueArray *input_channel_positions; - gboolean channel_positions_from_input; - - gint default_channels_ordering_map[64]; - - GstInterleaveFunc func; -}; - -struct _GstAudioInterleaveClass { - GstAudioAggregatorClass parent_class; -}; - -GType gst_audio_interleave_get_type (void); - -#define GST_TYPE_AUDIO_INTERLEAVE_PAD (gst_audio_interleave_pad_get_type()) -#define GST_AUDIO_INTERLEAVE_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_INTERLEAVE_PAD,GstAudioInterleavePad)) -#define GST_IS_AUDIO_INTERLEAVE_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_INTERLEAVE_PAD)) -#define GST_AUDIO_INTERLEAVE_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_INTERLEAVE_PAD,GstAudioInterleavePadClass)) -#define GST_IS_AUDIO_INTERLEAVE_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_INTERLEAVE_PAD)) -#define GST_AUDIO_INTERLEAVE_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_INTERLEAVE_PAD,GstAudioInterleavePadClass)) - -struct _GstAudioInterleavePad { - GstAudioAggregatorPad parent; - - guint channel; -}; - -struct _GstAudioInterleavePadClass { - GstAudioAggregatorPadClass parent_class; -}; - -GType gst_audio_interleave_pad_get_type (void); - -G_END_DECLS - - -#endif /* __GST_AUDIO_INTERLEAVE_H__ */ diff --git a/gst/audiomixer/gstaudiomixer.c b/gst/audiomixer/gstaudiomixer.c deleted file mode 100644 index a0f5690101..0000000000 --- a/gst/audiomixer/gstaudiomixer.c +++ /dev/null @@ -1,577 +0,0 @@ -/* 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 - * 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 -#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_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 "); - - 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) diff --git a/gst/audiomixer/gstaudiomixer.h b/gst/audiomixer/gstaudiomixer.h deleted file mode 100644 index 67ccb27e6d..0000000000 --- a/gst/audiomixer/gstaudiomixer.h +++ /dev/null @@ -1,87 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen - * 2000 Wim Taymans - * Copyright (C) 2013 Sebastian Dröge - * - * gstaudiomixer.h: Header for GstAudioMixer element - * - * 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. - */ - -#ifndef __GST_AUDIO_MIXER_H__ -#define __GST_AUDIO_MIXER_H__ - -#include -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_AUDIO_MIXER (gst_audiomixer_get_type()) -#define GST_AUDIO_MIXER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_MIXER,GstAudioMixer)) -#define GST_IS_AUDIO_MIXER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_MIXER)) -#define GST_AUDIO_MIXER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_MIXER,GstAudioMixerClass)) -#define GST_IS_AUDIO_MIXER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_MIXER)) -#define GST_AUDIO_MIXER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_MIXER,GstAudioMixerClass)) - -typedef struct _GstAudioMixer GstAudioMixer; -typedef struct _GstAudioMixerClass GstAudioMixerClass; - -typedef struct _GstAudioMixerPad GstAudioMixerPad; -typedef struct _GstAudioMixerPadClass GstAudioMixerPadClass; - -/** - * GstAudioMixer: - * - * The audiomixer object structure. - */ -struct _GstAudioMixer { - GstAudioAggregator element; -}; - -struct _GstAudioMixerClass { - GstAudioAggregatorClass parent_class; -}; - -GType gst_audiomixer_get_type (void); - -#define GST_TYPE_AUDIO_MIXER_PAD (gst_audiomixer_pad_get_type()) -#define GST_AUDIO_MIXER_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPad)) -#define GST_IS_AUDIO_MIXER_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_MIXER_PAD)) -#define GST_AUDIO_MIXER_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPadClass)) -#define GST_IS_AUDIO_MIXER_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_MIXER_PAD)) -#define GST_AUDIO_MIXER_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPadClass)) - -struct _GstAudioMixerPad { - GstAudioAggregatorConvertPad parent; - - gdouble volume; - gint volume_i32; - gint volume_i16; - gint volume_i8; - gboolean mute; -}; - -struct _GstAudioMixerPadClass { - GstAudioAggregatorConvertPadClass parent_class; -}; - -GType gst_audiomixer_pad_get_type (void); - -G_END_DECLS - - -#endif /* __GST_AUDIO_MIXER_H__ */ diff --git a/gst/audiomixer/gstaudiomixerorc-dist.c b/gst/audiomixer/gstaudiomixerorc-dist.c deleted file mode 100644 index be377f7054..0000000000 --- a/gst/audiomixer/gstaudiomixerorc-dist.c +++ /dev/null @@ -1,2605 +0,0 @@ - -/* autogenerated from gstaudiomixerorc.orc */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif -#include - -#ifndef _ORC_INTEGER_TYPEDEFS_ -#define _ORC_INTEGER_TYPEDEFS_ -#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L -#include -typedef int8_t orc_int8; -typedef int16_t orc_int16; -typedef int32_t orc_int32; -typedef int64_t orc_int64; -typedef uint8_t orc_uint8; -typedef uint16_t orc_uint16; -typedef uint32_t orc_uint32; -typedef uint64_t orc_uint64; -#define ORC_UINT64_C(x) UINT64_C(x) -#elif defined(_MSC_VER) -typedef signed __int8 orc_int8; -typedef signed __int16 orc_int16; -typedef signed __int32 orc_int32; -typedef signed __int64 orc_int64; -typedef unsigned __int8 orc_uint8; -typedef unsigned __int16 orc_uint16; -typedef unsigned __int32 orc_uint32; -typedef unsigned __int64 orc_uint64; -#define ORC_UINT64_C(x) (x##Ui64) -#define inline __inline -#else -#include -typedef signed char orc_int8; -typedef short orc_int16; -typedef int orc_int32; -typedef unsigned char orc_uint8; -typedef unsigned short orc_uint16; -typedef unsigned int orc_uint32; -#if INT_MAX == LONG_MAX -typedef long long orc_int64; -typedef unsigned long long orc_uint64; -#define ORC_UINT64_C(x) (x##ULL) -#else -typedef long orc_int64; -typedef unsigned long orc_uint64; -#define ORC_UINT64_C(x) (x##UL) -#endif -#endif -typedef union -{ - orc_int16 i; - orc_int8 x2[2]; -} orc_union16; -typedef union -{ - orc_int32 i; - float f; - orc_int16 x2[2]; - orc_int8 x4[4]; -} orc_union32; -typedef union -{ - orc_int64 i; - double f; - orc_int32 x2[2]; - float x2f[2]; - orc_int16 x4[4]; -} orc_union64; -#endif -#ifndef ORC_RESTRICT -#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L -#define ORC_RESTRICT restrict -#elif defined(__GNUC__) && __GNUC__ >= 4 -#define ORC_RESTRICT __restrict__ -#else -#define ORC_RESTRICT -#endif -#endif - -#ifndef ORC_INTERNAL -#if defined(__SUNPRO_C) && (__SUNPRO_C >= 0x590) -#define ORC_INTERNAL __attribute__((visibility("hidden"))) -#elif defined(__SUNPRO_C) && (__SUNPRO_C >= 0x550) -#define ORC_INTERNAL __hidden -#elif defined (__GNUC__) -#define ORC_INTERNAL __attribute__((visibility("hidden"))) -#else -#define ORC_INTERNAL -#endif -#endif - - -#ifndef DISABLE_ORC -#include -#endif -void audiomixer_orc_add_s32 (gint32 * ORC_RESTRICT d1, - const gint32 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_s16 (gint16 * ORC_RESTRICT d1, - const gint16 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_s8 (gint8 * ORC_RESTRICT d1, - const gint8 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_u32 (guint32 * ORC_RESTRICT d1, - const guint32 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_u16 (guint16 * ORC_RESTRICT d1, - const guint16 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_u8 (guint8 * ORC_RESTRICT d1, - const guint8 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_f32 (float *ORC_RESTRICT d1, - const float *ORC_RESTRICT s1, int n); -void audiomixer_orc_add_f64 (double *ORC_RESTRICT d1, - const double *ORC_RESTRICT s1, int n); -void audiomixer_orc_volume_u8 (guint8 * ORC_RESTRICT d1, int p1, int n); -void audiomixer_orc_add_volume_u8 (guint8 * ORC_RESTRICT d1, - const guint8 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_s8 (gint8 * ORC_RESTRICT d1, - const gint8 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_u16 (guint16 * ORC_RESTRICT d1, - const guint16 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_s16 (gint16 * ORC_RESTRICT d1, - const gint16 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_u32 (guint32 * ORC_RESTRICT d1, - const guint32 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_s32 (gint32 * ORC_RESTRICT d1, - const gint32 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_f32 (float *ORC_RESTRICT d1, - const float *ORC_RESTRICT s1, float p1, int n); -void audiomixer_orc_add_volume_f64 (double *ORC_RESTRICT d1, - const double *ORC_RESTRICT s1, double p1, int n); - - -/* begin Orc C target preamble */ -#define ORC_CLAMP(x,a,b) ((x)<(a) ? (a) : ((x)>(b) ? (b) : (x))) -#define ORC_ABS(a) ((a)<0 ? -(a) : (a)) -#define ORC_MIN(a,b) ((a)<(b) ? (a) : (b)) -#define ORC_MAX(a,b) ((a)>(b) ? (a) : (b)) -#define ORC_SB_MAX 127 -#define ORC_SB_MIN (-1-ORC_SB_MAX) -#define ORC_UB_MAX 255 -#define ORC_UB_MIN 0 -#define ORC_SW_MAX 32767 -#define ORC_SW_MIN (-1-ORC_SW_MAX) -#define ORC_UW_MAX 65535 -#define ORC_UW_MIN 0 -#define ORC_SL_MAX 2147483647 -#define ORC_SL_MIN (-1-ORC_SL_MAX) -#define ORC_UL_MAX 4294967295U -#define ORC_UL_MIN 0 -#define ORC_CLAMP_SB(x) ORC_CLAMP(x,ORC_SB_MIN,ORC_SB_MAX) -#define ORC_CLAMP_UB(x) ORC_CLAMP(x,ORC_UB_MIN,ORC_UB_MAX) -#define ORC_CLAMP_SW(x) ORC_CLAMP(x,ORC_SW_MIN,ORC_SW_MAX) -#define ORC_CLAMP_UW(x) ORC_CLAMP(x,ORC_UW_MIN,ORC_UW_MAX) -#define ORC_CLAMP_SL(x) ORC_CLAMP(x,ORC_SL_MIN,ORC_SL_MAX) -#define ORC_CLAMP_UL(x) ORC_CLAMP(x,ORC_UL_MIN,ORC_UL_MAX) -#define ORC_SWAP_W(x) ((((x)&0xffU)<<8) | (((x)&0xff00U)>>8)) -#define ORC_SWAP_L(x) ((((x)&0xffU)<<24) | (((x)&0xff00U)<<8) | (((x)&0xff0000U)>>8) | (((x)&0xff000000U)>>24)) -#define ORC_SWAP_Q(x) ((((x)&ORC_UINT64_C(0xff))<<56) | (((x)&ORC_UINT64_C(0xff00))<<40) | (((x)&ORC_UINT64_C(0xff0000))<<24) | (((x)&ORC_UINT64_C(0xff000000))<<8) | (((x)&ORC_UINT64_C(0xff00000000))>>8) | (((x)&ORC_UINT64_C(0xff0000000000))>>24) | (((x)&ORC_UINT64_C(0xff000000000000))>>40) | (((x)&ORC_UINT64_C(0xff00000000000000))>>56)) -#define ORC_PTR_OFFSET(ptr,offset) ((void *)(((unsigned char *)(ptr)) + (offset))) -#define ORC_DENORMAL(x) ((x) & ((((x)&0x7f800000) == 0) ? 0xff800000 : 0xffffffff)) -#define ORC_ISNAN(x) ((((x)&0x7f800000) == 0x7f800000) && (((x)&0x007fffff) != 0)) -#define ORC_DENORMAL_DOUBLE(x) ((x) & ((((x)&ORC_UINT64_C(0x7ff0000000000000)) == 0) ? ORC_UINT64_C(0xfff0000000000000) : ORC_UINT64_C(0xffffffffffffffff))) -#define ORC_ISNAN_DOUBLE(x) ((((x)&ORC_UINT64_C(0x7ff0000000000000)) == ORC_UINT64_C(0x7ff0000000000000)) && (((x)&ORC_UINT64_C(0x000fffffffffffff)) != 0)) -#ifndef ORC_RESTRICT -#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L -#define ORC_RESTRICT restrict -#elif defined(__GNUC__) && __GNUC__ >= 4 -#define ORC_RESTRICT __restrict__ -#else -#define ORC_RESTRICT -#endif -#endif -/* end Orc C target preamble */ - - - -/* audiomixer_orc_add_s32 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_s32 (gint32 * ORC_RESTRICT d1, - const gint32 * ORC_RESTRICT s1, int n) -{ - int i; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var32; - orc_union32 var33; - orc_union32 var34; - - ptr0 = (orc_union32 *) d1; - ptr4 = (orc_union32 *) s1; - - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var32 = ptr0[i]; - /* 1: loadl */ - var33 = ptr4[i]; - /* 2: addssl */ - var34.i = ORC_CLAMP_SL ((orc_int64) var32.i + (orc_int64) var33.i); - /* 3: storel */ - ptr0[i] = var34; - } - -} - -#else -static void -_backup_audiomixer_orc_add_s32 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var32; - orc_union32 var33; - orc_union32 var34; - - ptr0 = (orc_union32 *) ex->arrays[0]; - ptr4 = (orc_union32 *) ex->arrays[4]; - - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var32 = ptr0[i]; - /* 1: loadl */ - var33 = ptr4[i]; - /* 2: addssl */ - var34.i = ORC_CLAMP_SL ((orc_int64) var32.i + (orc_int64) var33.i); - /* 3: storel */ - ptr0[i] = var34; - } - -} - -void -audiomixer_orc_add_s32 (gint32 * ORC_RESTRICT d1, - const gint32 * ORC_RESTRICT s1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 115, 51, 50, 11, 4, 4, 12, 4, 4, 104, - 0, 0, 4, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s32); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_s32"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s32); - orc_program_add_destination (p, 4, "d1"); - orc_program_add_source (p, 4, "s1"); - - orc_program_append_2 (p, "addssl", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_s16 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_s16 (gint16 * ORC_RESTRICT d1, - const gint16 * ORC_RESTRICT s1, int n) -{ - int i; - orc_union16 *ORC_RESTRICT ptr0; - const orc_union16 *ORC_RESTRICT ptr4; - orc_union16 var32; - orc_union16 var33; - orc_union16 var34; - - ptr0 = (orc_union16 *) d1; - ptr4 = (orc_union16 *) s1; - - - for (i = 0; i < n; i++) { - /* 0: loadw */ - var32 = ptr0[i]; - /* 1: loadw */ - var33 = ptr4[i]; - /* 2: addssw */ - var34.i = ORC_CLAMP_SW (var32.i + var33.i); - /* 3: storew */ - ptr0[i] = var34; - } - -} - -#else -static void -_backup_audiomixer_orc_add_s16 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union16 *ORC_RESTRICT ptr0; - const orc_union16 *ORC_RESTRICT ptr4; - orc_union16 var32; - orc_union16 var33; - orc_union16 var34; - - ptr0 = (orc_union16 *) ex->arrays[0]; - ptr4 = (orc_union16 *) ex->arrays[4]; - - - for (i = 0; i < n; i++) { - /* 0: loadw */ - var32 = ptr0[i]; - /* 1: loadw */ - var33 = ptr4[i]; - /* 2: addssw */ - var34.i = ORC_CLAMP_SW (var32.i + var33.i); - /* 3: storew */ - ptr0[i] = var34; - } - -} - -void -audiomixer_orc_add_s16 (gint16 * ORC_RESTRICT d1, - const gint16 * ORC_RESTRICT s1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 115, 49, 54, 11, 2, 2, 12, 2, 2, 71, - 0, 0, 4, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s16); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_s16"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s16); - orc_program_add_destination (p, 2, "d1"); - orc_program_add_source (p, 2, "s1"); - - orc_program_append_2 (p, "addssw", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_s8 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_s8 (gint8 * ORC_RESTRICT d1, const gint8 * ORC_RESTRICT s1, - int n) -{ - int i; - orc_int8 *ORC_RESTRICT ptr0; - const orc_int8 *ORC_RESTRICT ptr4; - orc_int8 var32; - orc_int8 var33; - orc_int8 var34; - - ptr0 = (orc_int8 *) d1; - ptr4 = (orc_int8 *) s1; - - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var32 = ptr0[i]; - /* 1: loadb */ - var33 = ptr4[i]; - /* 2: addssb */ - var34 = ORC_CLAMP_SB (var32 + var33); - /* 3: storeb */ - ptr0[i] = var34; - } - -} - -#else -static void -_backup_audiomixer_orc_add_s8 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_int8 *ORC_RESTRICT ptr0; - const orc_int8 *ORC_RESTRICT ptr4; - orc_int8 var32; - orc_int8 var33; - orc_int8 var34; - - ptr0 = (orc_int8 *) ex->arrays[0]; - ptr4 = (orc_int8 *) ex->arrays[4]; - - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var32 = ptr0[i]; - /* 1: loadb */ - var33 = ptr4[i]; - /* 2: addssb */ - var34 = ORC_CLAMP_SB (var32 + var33); - /* 3: storeb */ - ptr0[i] = var34; - } - -} - -void -audiomixer_orc_add_s8 (gint8 * ORC_RESTRICT d1, const gint8 * ORC_RESTRICT s1, - int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 21, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 115, 56, 11, 1, 1, 12, 1, 1, 34, 0, - 0, 4, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s8); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_s8"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_s8); - orc_program_add_destination (p, 1, "d1"); - orc_program_add_source (p, 1, "s1"); - - orc_program_append_2 (p, "addssb", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_u32 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_u32 (guint32 * ORC_RESTRICT d1, - const guint32 * ORC_RESTRICT s1, int n) -{ - int i; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var32; - orc_union32 var33; - orc_union32 var34; - - ptr0 = (orc_union32 *) d1; - ptr4 = (orc_union32 *) s1; - - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var32 = ptr0[i]; - /* 1: loadl */ - var33 = ptr4[i]; - /* 2: addusl */ - var34.i = - ORC_CLAMP_UL ((orc_int64) (orc_uint32) var32.i + - (orc_int64) (orc_uint32) var33.i); - /* 3: storel */ - ptr0[i] = var34; - } - -} - -#else -static void -_backup_audiomixer_orc_add_u32 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var32; - orc_union32 var33; - orc_union32 var34; - - ptr0 = (orc_union32 *) ex->arrays[0]; - ptr4 = (orc_union32 *) ex->arrays[4]; - - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var32 = ptr0[i]; - /* 1: loadl */ - var33 = ptr4[i]; - /* 2: addusl */ - var34.i = - ORC_CLAMP_UL ((orc_int64) (orc_uint32) var32.i + - (orc_int64) (orc_uint32) var33.i); - /* 3: storel */ - ptr0[i] = var34; - } - -} - -void -audiomixer_orc_add_u32 (guint32 * ORC_RESTRICT d1, - const guint32 * ORC_RESTRICT s1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 117, 51, 50, 11, 4, 4, 12, 4, 4, 105, - 0, 0, 4, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u32); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_u32"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u32); - orc_program_add_destination (p, 4, "d1"); - orc_program_add_source (p, 4, "s1"); - - orc_program_append_2 (p, "addusl", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_u16 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_u16 (guint16 * ORC_RESTRICT d1, - const guint16 * ORC_RESTRICT s1, int n) -{ - int i; - orc_union16 *ORC_RESTRICT ptr0; - const orc_union16 *ORC_RESTRICT ptr4; - orc_union16 var32; - orc_union16 var33; - orc_union16 var34; - - ptr0 = (orc_union16 *) d1; - ptr4 = (orc_union16 *) s1; - - - for (i = 0; i < n; i++) { - /* 0: loadw */ - var32 = ptr0[i]; - /* 1: loadw */ - var33 = ptr4[i]; - /* 2: addusw */ - var34.i = ORC_CLAMP_UW ((orc_uint16) var32.i + (orc_uint16) var33.i); - /* 3: storew */ - ptr0[i] = var34; - } - -} - -#else -static void -_backup_audiomixer_orc_add_u16 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union16 *ORC_RESTRICT ptr0; - const orc_union16 *ORC_RESTRICT ptr4; - orc_union16 var32; - orc_union16 var33; - orc_union16 var34; - - ptr0 = (orc_union16 *) ex->arrays[0]; - ptr4 = (orc_union16 *) ex->arrays[4]; - - - for (i = 0; i < n; i++) { - /* 0: loadw */ - var32 = ptr0[i]; - /* 1: loadw */ - var33 = ptr4[i]; - /* 2: addusw */ - var34.i = ORC_CLAMP_UW ((orc_uint16) var32.i + (orc_uint16) var33.i); - /* 3: storew */ - ptr0[i] = var34; - } - -} - -void -audiomixer_orc_add_u16 (guint16 * ORC_RESTRICT d1, - const guint16 * ORC_RESTRICT s1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 117, 49, 54, 11, 2, 2, 12, 2, 2, 72, - 0, 0, 4, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u16); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_u16"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u16); - orc_program_add_destination (p, 2, "d1"); - orc_program_add_source (p, 2, "s1"); - - orc_program_append_2 (p, "addusw", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_u8 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, - int n) -{ - int i; - orc_int8 *ORC_RESTRICT ptr0; - const orc_int8 *ORC_RESTRICT ptr4; - orc_int8 var32; - orc_int8 var33; - orc_int8 var34; - - ptr0 = (orc_int8 *) d1; - ptr4 = (orc_int8 *) s1; - - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var32 = ptr0[i]; - /* 1: loadb */ - var33 = ptr4[i]; - /* 2: addusb */ - var34 = ORC_CLAMP_UB ((orc_uint8) var32 + (orc_uint8) var33); - /* 3: storeb */ - ptr0[i] = var34; - } - -} - -#else -static void -_backup_audiomixer_orc_add_u8 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_int8 *ORC_RESTRICT ptr0; - const orc_int8 *ORC_RESTRICT ptr4; - orc_int8 var32; - orc_int8 var33; - orc_int8 var34; - - ptr0 = (orc_int8 *) ex->arrays[0]; - ptr4 = (orc_int8 *) ex->arrays[4]; - - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var32 = ptr0[i]; - /* 1: loadb */ - var33 = ptr4[i]; - /* 2: addusb */ - var34 = ORC_CLAMP_UB ((orc_uint8) var32 + (orc_uint8) var33); - /* 3: storeb */ - ptr0[i] = var34; - } - -} - -void -audiomixer_orc_add_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, - int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 21, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 117, 56, 11, 1, 1, 12, 1, 1, 35, 0, - 0, 4, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u8); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_u8"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_u8); - orc_program_add_destination (p, 1, "d1"); - orc_program_add_source (p, 1, "s1"); - - orc_program_append_2 (p, "addusb", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_f32 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_f32 (float *ORC_RESTRICT d1, const float *ORC_RESTRICT s1, - int n) -{ - int i; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var32; - orc_union32 var33; - orc_union32 var34; - - ptr0 = (orc_union32 *) d1; - ptr4 = (orc_union32 *) s1; - - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var32 = ptr0[i]; - /* 1: loadl */ - var33 = ptr4[i]; - /* 2: addf */ - { - orc_union32 _src1; - orc_union32 _src2; - orc_union32 _dest1; - _src1.i = ORC_DENORMAL (var32.i); - _src2.i = ORC_DENORMAL (var33.i); - _dest1.f = _src1.f + _src2.f; - var34.i = ORC_DENORMAL (_dest1.i); - } - /* 3: storel */ - ptr0[i] = var34; - } - -} - -#else -static void -_backup_audiomixer_orc_add_f32 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var32; - orc_union32 var33; - orc_union32 var34; - - ptr0 = (orc_union32 *) ex->arrays[0]; - ptr4 = (orc_union32 *) ex->arrays[4]; - - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var32 = ptr0[i]; - /* 1: loadl */ - var33 = ptr4[i]; - /* 2: addf */ - { - orc_union32 _src1; - orc_union32 _src2; - orc_union32 _dest1; - _src1.i = ORC_DENORMAL (var32.i); - _src2.i = ORC_DENORMAL (var33.i); - _dest1.f = _src1.f + _src2.f; - var34.i = ORC_DENORMAL (_dest1.i); - } - /* 3: storel */ - ptr0[i] = var34; - } - -} - -void -audiomixer_orc_add_f32 (float *ORC_RESTRICT d1, const float *ORC_RESTRICT s1, - int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 102, 51, 50, 11, 4, 4, 12, 4, 4, 200, - 0, 0, 4, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_f32); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_f32"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_f32); - orc_program_add_destination (p, 4, "d1"); - orc_program_add_source (p, 4, "s1"); - - orc_program_append_2 (p, "addf", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_f64 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_f64 (double *ORC_RESTRICT d1, const double *ORC_RESTRICT s1, - int n) -{ - int i; - orc_union64 *ORC_RESTRICT ptr0; - const orc_union64 *ORC_RESTRICT ptr4; - orc_union64 var32; - orc_union64 var33; - orc_union64 var34; - - ptr0 = (orc_union64 *) d1; - ptr4 = (orc_union64 *) s1; - - - for (i = 0; i < n; i++) { - /* 0: loadq */ - var32 = ptr0[i]; - /* 1: loadq */ - var33 = ptr4[i]; - /* 2: addd */ - { - orc_union64 _src1; - orc_union64 _src2; - orc_union64 _dest1; - _src1.i = ORC_DENORMAL_DOUBLE (var32.i); - _src2.i = ORC_DENORMAL_DOUBLE (var33.i); - _dest1.f = _src1.f + _src2.f; - var34.i = ORC_DENORMAL_DOUBLE (_dest1.i); - } - /* 3: storeq */ - ptr0[i] = var34; - } - -} - -#else -static void -_backup_audiomixer_orc_add_f64 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union64 *ORC_RESTRICT ptr0; - const orc_union64 *ORC_RESTRICT ptr4; - orc_union64 var32; - orc_union64 var33; - orc_union64 var34; - - ptr0 = (orc_union64 *) ex->arrays[0]; - ptr4 = (orc_union64 *) ex->arrays[4]; - - - for (i = 0; i < n; i++) { - /* 0: loadq */ - var32 = ptr0[i]; - /* 1: loadq */ - var33 = ptr4[i]; - /* 2: addd */ - { - orc_union64 _src1; - orc_union64 _src2; - orc_union64 _dest1; - _src1.i = ORC_DENORMAL_DOUBLE (var32.i); - _src2.i = ORC_DENORMAL_DOUBLE (var33.i); - _dest1.f = _src1.f + _src2.f; - var34.i = ORC_DENORMAL_DOUBLE (_dest1.i); - } - /* 3: storeq */ - ptr0[i] = var34; - } - -} - -void -audiomixer_orc_add_f64 (double *ORC_RESTRICT d1, const double *ORC_RESTRICT s1, - int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 22, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 102, 54, 52, 11, 8, 8, 12, 8, 8, 212, - 0, 0, 4, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_f64); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_f64"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_f64); - orc_program_add_destination (p, 8, "d1"); - orc_program_add_source (p, 8, "s1"); - - orc_program_append_2 (p, "addd", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_S1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_volume_u8 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_volume_u8 (guint8 * ORC_RESTRICT d1, int p1, int n) -{ - int i; - orc_int8 *ORC_RESTRICT ptr0; - orc_int8 var34; -#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__) - volatile orc_int8 var35; -#else - orc_int8 var35; -#endif - orc_int8 var36; - orc_int8 var37; - orc_int8 var38; - orc_union16 var39; - orc_union16 var40; - orc_int8 var41; - - ptr0 = (orc_int8 *) d1; - - /* 1: loadpb */ - var35 = (int) 0x00000080; /* 128 or 6.32404e-322f */ - /* 3: loadpb */ - var36 = p1; - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var34 = ptr0[i]; - /* 2: xorb */ - var38 = var34 ^ var35; - /* 4: mulsbw */ - var39.i = var38 * var36; - /* 5: shrsw */ - var40.i = var39.i >> 3; - /* 6: convssswb */ - var41 = ORC_CLAMP_SB (var40.i); - /* 7: xorb */ - var37 = var41 ^ var35; - /* 8: storeb */ - ptr0[i] = var37; - } - -} - -#else -static void -_backup_audiomixer_orc_volume_u8 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_int8 *ORC_RESTRICT ptr0; - orc_int8 var34; -#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__) - volatile orc_int8 var35; -#else - orc_int8 var35; -#endif - orc_int8 var36; - orc_int8 var37; - orc_int8 var38; - orc_union16 var39; - orc_union16 var40; - orc_int8 var41; - - ptr0 = (orc_int8 *) ex->arrays[0]; - - /* 1: loadpb */ - var35 = (int) 0x00000080; /* 128 or 6.32404e-322f */ - /* 3: loadpb */ - var36 = ex->params[24]; - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var34 = ptr0[i]; - /* 2: xorb */ - var38 = var34 ^ var35; - /* 4: mulsbw */ - var39.i = var38 * var36; - /* 5: shrsw */ - var40.i = var39.i >> 3; - /* 6: convssswb */ - var41 = ORC_CLAMP_SB (var40.i); - /* 7: xorb */ - var37 = var41 ^ var35; - /* 8: storeb */ - ptr0[i] = var37; - } - -} - -void -audiomixer_orc_volume_u8 (guint8 * ORC_RESTRICT d1, int p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 24, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 118, 111, 108, 117, 109, 101, 95, 117, 56, 11, 1, 1, 14, 1, - 128, 0, 0, 0, 14, 2, 3, 0, 0, 0, 16, 1, 20, 2, 20, 1, - 68, 33, 0, 16, 174, 32, 33, 24, 94, 32, 32, 17, 159, 33, 32, 68, - 0, 33, 16, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_volume_u8); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_volume_u8"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_volume_u8); - orc_program_add_destination (p, 1, "d1"); - orc_program_add_constant (p, 1, 0x00000080, "c1"); - orc_program_add_constant (p, 2, 0x00000003, "c2"); - orc_program_add_parameter (p, 1, "p1"); - orc_program_add_temporary (p, 2, "t1"); - orc_program_add_temporary (p, 1, "t2"); - - orc_program_append_2 (p, "xorb", 0, ORC_VAR_T2, ORC_VAR_D1, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "mulsbw", 0, ORC_VAR_T1, ORC_VAR_T2, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "shrsw", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C2, - ORC_VAR_D1); - orc_program_append_2 (p, "convssswb", 0, ORC_VAR_T2, ORC_VAR_T1, - ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "xorb", 0, ORC_VAR_D1, ORC_VAR_T2, ORC_VAR_C1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->params[ORC_VAR_P1] = p1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_volume_u8 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_volume_u8 (guint8 * ORC_RESTRICT d1, - const guint8 * ORC_RESTRICT s1, int p1, int n) -{ - int i; - orc_int8 *ORC_RESTRICT ptr0; - const orc_int8 *ORC_RESTRICT ptr4; - orc_int8 var34; -#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__) - volatile orc_int8 var35; -#else - orc_int8 var35; -#endif - orc_int8 var36; - orc_int8 var37; - orc_int8 var38; - orc_int8 var39; - orc_union16 var40; - orc_union16 var41; - orc_int8 var42; - orc_int8 var43; - - ptr0 = (orc_int8 *) d1; - ptr4 = (orc_int8 *) s1; - - /* 1: loadpb */ - var35 = (int) 0x00000080; /* 128 or 6.32404e-322f */ - /* 3: loadpb */ - var36 = p1; - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var34 = ptr4[i]; - /* 2: xorb */ - var39 = var34 ^ var35; - /* 4: mulsbw */ - var40.i = var39 * var36; - /* 5: shrsw */ - var41.i = var40.i >> 3; - /* 6: convssswb */ - var42 = ORC_CLAMP_SB (var41.i); - /* 7: xorb */ - var43 = var42 ^ var35; - /* 8: loadb */ - var37 = ptr0[i]; - /* 9: addusb */ - var38 = ORC_CLAMP_UB ((orc_uint8) var37 + (orc_uint8) var43); - /* 10: storeb */ - ptr0[i] = var38; - } - -} - -#else -static void -_backup_audiomixer_orc_add_volume_u8 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_int8 *ORC_RESTRICT ptr0; - const orc_int8 *ORC_RESTRICT ptr4; - orc_int8 var34; -#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__) - volatile orc_int8 var35; -#else - orc_int8 var35; -#endif - orc_int8 var36; - orc_int8 var37; - orc_int8 var38; - orc_int8 var39; - orc_union16 var40; - orc_union16 var41; - orc_int8 var42; - orc_int8 var43; - - ptr0 = (orc_int8 *) ex->arrays[0]; - ptr4 = (orc_int8 *) ex->arrays[4]; - - /* 1: loadpb */ - var35 = (int) 0x00000080; /* 128 or 6.32404e-322f */ - /* 3: loadpb */ - var36 = ex->params[24]; - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var34 = ptr4[i]; - /* 2: xorb */ - var39 = var34 ^ var35; - /* 4: mulsbw */ - var40.i = var39 * var36; - /* 5: shrsw */ - var41.i = var40.i >> 3; - /* 6: convssswb */ - var42 = ORC_CLAMP_SB (var41.i); - /* 7: xorb */ - var43 = var42 ^ var35; - /* 8: loadb */ - var37 = ptr0[i]; - /* 9: addusb */ - var38 = ORC_CLAMP_UB ((orc_uint8) var37 + (orc_uint8) var43); - /* 10: storeb */ - ptr0[i] = var38; - } - -} - -void -audiomixer_orc_add_volume_u8 (guint8 * ORC_RESTRICT d1, - const guint8 * ORC_RESTRICT s1, int p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 28, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 117, 56, 11, - 1, 1, 12, 1, 1, 14, 1, 128, 0, 0, 0, 14, 2, 3, 0, 0, - 0, 16, 1, 20, 2, 20, 1, 68, 33, 4, 16, 174, 32, 33, 24, 94, - 32, 32, 17, 159, 33, 32, 68, 33, 33, 16, 35, 0, 0, 33, 2, 0, - - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_volume_u8); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_volume_u8"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_volume_u8); - orc_program_add_destination (p, 1, "d1"); - orc_program_add_source (p, 1, "s1"); - orc_program_add_constant (p, 1, 0x00000080, "c1"); - orc_program_add_constant (p, 2, 0x00000003, "c2"); - orc_program_add_parameter (p, 1, "p1"); - orc_program_add_temporary (p, 2, "t1"); - orc_program_add_temporary (p, 1, "t2"); - - orc_program_append_2 (p, "xorb", 0, ORC_VAR_T2, ORC_VAR_S1, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "mulsbw", 0, ORC_VAR_T1, ORC_VAR_T2, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "shrsw", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C2, - ORC_VAR_D1); - orc_program_append_2 (p, "convssswb", 0, ORC_VAR_T2, ORC_VAR_T1, - ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "xorb", 0, ORC_VAR_T2, ORC_VAR_T2, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "addusb", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - ex->params[ORC_VAR_P1] = p1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_volume_s8 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_volume_s8 (gint8 * ORC_RESTRICT d1, - const gint8 * ORC_RESTRICT s1, int p1, int n) -{ - int i; - orc_int8 *ORC_RESTRICT ptr0; - const orc_int8 *ORC_RESTRICT ptr4; - orc_int8 var34; - orc_int8 var35; - orc_int8 var36; - orc_int8 var37; - orc_union16 var38; - orc_union16 var39; - orc_int8 var40; - - ptr0 = (orc_int8 *) d1; - ptr4 = (orc_int8 *) s1; - - /* 1: loadpb */ - var35 = p1; - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var34 = ptr4[i]; - /* 2: mulsbw */ - var38.i = var34 * var35; - /* 3: shrsw */ - var39.i = var38.i >> 3; - /* 4: convssswb */ - var40 = ORC_CLAMP_SB (var39.i); - /* 5: loadb */ - var36 = ptr0[i]; - /* 6: addssb */ - var37 = ORC_CLAMP_SB (var36 + var40); - /* 7: storeb */ - ptr0[i] = var37; - } - -} - -#else -static void -_backup_audiomixer_orc_add_volume_s8 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_int8 *ORC_RESTRICT ptr0; - const orc_int8 *ORC_RESTRICT ptr4; - orc_int8 var34; - orc_int8 var35; - orc_int8 var36; - orc_int8 var37; - orc_union16 var38; - orc_union16 var39; - orc_int8 var40; - - ptr0 = (orc_int8 *) ex->arrays[0]; - ptr4 = (orc_int8 *) ex->arrays[4]; - - /* 1: loadpb */ - var35 = ex->params[24]; - - for (i = 0; i < n; i++) { - /* 0: loadb */ - var34 = ptr4[i]; - /* 2: mulsbw */ - var38.i = var34 * var35; - /* 3: shrsw */ - var39.i = var38.i >> 3; - /* 4: convssswb */ - var40 = ORC_CLAMP_SB (var39.i); - /* 5: loadb */ - var36 = ptr0[i]; - /* 6: addssb */ - var37 = ORC_CLAMP_SB (var36 + var40); - /* 7: storeb */ - ptr0[i] = var37; - } - -} - -void -audiomixer_orc_add_volume_s8 (gint8 * ORC_RESTRICT d1, - const gint8 * ORC_RESTRICT s1, int p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 28, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 115, 56, 11, - 1, 1, 12, 1, 1, 14, 2, 3, 0, 0, 0, 16, 1, 20, 2, 20, - 1, 174, 32, 4, 24, 94, 32, 32, 16, 159, 33, 32, 34, 0, 0, 33, - 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_volume_s8); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_volume_s8"); - orc_program_set_backup_function (p, _backup_audiomixer_orc_add_volume_s8); - orc_program_add_destination (p, 1, "d1"); - orc_program_add_source (p, 1, "s1"); - orc_program_add_constant (p, 2, 0x00000003, "c1"); - orc_program_add_parameter (p, 1, "p1"); - orc_program_add_temporary (p, 2, "t1"); - orc_program_add_temporary (p, 1, "t2"); - - orc_program_append_2 (p, "mulsbw", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "shrsw", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "convssswb", 0, ORC_VAR_T2, ORC_VAR_T1, - ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "addssb", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - ex->params[ORC_VAR_P1] = p1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_volume_u16 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_volume_u16 (guint16 * ORC_RESTRICT d1, - const guint16 * ORC_RESTRICT s1, int p1, int n) -{ - int i; - orc_union16 *ORC_RESTRICT ptr0; - const orc_union16 *ORC_RESTRICT ptr4; - orc_union16 var34; -#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__) - volatile orc_union16 var35; -#else - orc_union16 var35; -#endif - orc_union16 var36; - orc_union16 var37; - orc_union16 var38; - orc_union16 var39; - orc_union32 var40; - orc_union32 var41; - orc_union16 var42; - orc_union16 var43; - - ptr0 = (orc_union16 *) d1; - ptr4 = (orc_union16 *) s1; - - /* 1: loadpw */ - var35.i = (int) 0x00008000; /* 32768 or 1.61895e-319f */ - /* 3: loadpw */ - var36.i = p1; - - for (i = 0; i < n; i++) { - /* 0: loadw */ - var34 = ptr4[i]; - /* 2: xorw */ - var39.i = var34.i ^ var35.i; - /* 4: mulswl */ - var40.i = var39.i * var36.i; - /* 5: shrsl */ - var41.i = var40.i >> 11; - /* 6: convssslw */ - var42.i = ORC_CLAMP_SW (var41.i); - /* 7: xorw */ - var43.i = var42.i ^ var35.i; - /* 8: loadw */ - var37 = ptr0[i]; - /* 9: addusw */ - var38.i = ORC_CLAMP_UW ((orc_uint16) var37.i + (orc_uint16) var43.i); - /* 10: storew */ - ptr0[i] = var38; - } - -} - -#else -static void -_backup_audiomixer_orc_add_volume_u16 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union16 *ORC_RESTRICT ptr0; - const orc_union16 *ORC_RESTRICT ptr4; - orc_union16 var34; -#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__) - volatile orc_union16 var35; -#else - orc_union16 var35; -#endif - orc_union16 var36; - orc_union16 var37; - orc_union16 var38; - orc_union16 var39; - orc_union32 var40; - orc_union32 var41; - orc_union16 var42; - orc_union16 var43; - - ptr0 = (orc_union16 *) ex->arrays[0]; - ptr4 = (orc_union16 *) ex->arrays[4]; - - /* 1: loadpw */ - var35.i = (int) 0x00008000; /* 32768 or 1.61895e-319f */ - /* 3: loadpw */ - var36.i = ex->params[24]; - - for (i = 0; i < n; i++) { - /* 0: loadw */ - var34 = ptr4[i]; - /* 2: xorw */ - var39.i = var34.i ^ var35.i; - /* 4: mulswl */ - var40.i = var39.i * var36.i; - /* 5: shrsl */ - var41.i = var40.i >> 11; - /* 6: convssslw */ - var42.i = ORC_CLAMP_SW (var41.i); - /* 7: xorw */ - var43.i = var42.i ^ var35.i; - /* 8: loadw */ - var37 = ptr0[i]; - /* 9: addusw */ - var38.i = ORC_CLAMP_UW ((orc_uint16) var37.i + (orc_uint16) var43.i); - /* 10: storew */ - ptr0[i] = var38; - } - -} - -void -audiomixer_orc_add_volume_u16 (guint16 * ORC_RESTRICT d1, - const guint16 * ORC_RESTRICT s1, int p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 117, 49, 54, - 11, 2, 2, 12, 2, 2, 14, 2, 0, 128, 0, 0, 14, 4, 11, 0, - 0, 0, 16, 2, 20, 4, 20, 2, 101, 33, 4, 16, 176, 32, 33, 24, - 125, 32, 32, 17, 165, 33, 32, 101, 33, 33, 16, 72, 0, 0, 33, 2, - 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_u16); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_volume_u16"); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_u16); - orc_program_add_destination (p, 2, "d1"); - orc_program_add_source (p, 2, "s1"); - orc_program_add_constant (p, 2, 0x00008000, "c1"); - orc_program_add_constant (p, 4, 0x0000000b, "c2"); - orc_program_add_parameter (p, 2, "p1"); - orc_program_add_temporary (p, 4, "t1"); - orc_program_add_temporary (p, 2, "t2"); - - orc_program_append_2 (p, "xorw", 0, ORC_VAR_T2, ORC_VAR_S1, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "mulswl", 0, ORC_VAR_T1, ORC_VAR_T2, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "shrsl", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C2, - ORC_VAR_D1); - orc_program_append_2 (p, "convssslw", 0, ORC_VAR_T2, ORC_VAR_T1, - ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "xorw", 0, ORC_VAR_T2, ORC_VAR_T2, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "addusw", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - ex->params[ORC_VAR_P1] = p1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_volume_s16 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_volume_s16 (gint16 * ORC_RESTRICT d1, - const gint16 * ORC_RESTRICT s1, int p1, int n) -{ - int i; - orc_union16 *ORC_RESTRICT ptr0; - const orc_union16 *ORC_RESTRICT ptr4; - orc_union16 var34; - orc_union16 var35; - orc_union16 var36; - orc_union16 var37; - orc_union32 var38; - orc_union32 var39; - orc_union16 var40; - - ptr0 = (orc_union16 *) d1; - ptr4 = (orc_union16 *) s1; - - /* 1: loadpw */ - var35.i = p1; - - for (i = 0; i < n; i++) { - /* 0: loadw */ - var34 = ptr4[i]; - /* 2: mulswl */ - var38.i = var34.i * var35.i; - /* 3: shrsl */ - var39.i = var38.i >> 11; - /* 4: convssslw */ - var40.i = ORC_CLAMP_SW (var39.i); - /* 5: loadw */ - var36 = ptr0[i]; - /* 6: addssw */ - var37.i = ORC_CLAMP_SW (var36.i + var40.i); - /* 7: storew */ - ptr0[i] = var37; - } - -} - -#else -static void -_backup_audiomixer_orc_add_volume_s16 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union16 *ORC_RESTRICT ptr0; - const orc_union16 *ORC_RESTRICT ptr4; - orc_union16 var34; - orc_union16 var35; - orc_union16 var36; - orc_union16 var37; - orc_union32 var38; - orc_union32 var39; - orc_union16 var40; - - ptr0 = (orc_union16 *) ex->arrays[0]; - ptr4 = (orc_union16 *) ex->arrays[4]; - - /* 1: loadpw */ - var35.i = ex->params[24]; - - for (i = 0; i < n; i++) { - /* 0: loadw */ - var34 = ptr4[i]; - /* 2: mulswl */ - var38.i = var34.i * var35.i; - /* 3: shrsl */ - var39.i = var38.i >> 11; - /* 4: convssslw */ - var40.i = ORC_CLAMP_SW (var39.i); - /* 5: loadw */ - var36 = ptr0[i]; - /* 6: addssw */ - var37.i = ORC_CLAMP_SW (var36.i + var40.i); - /* 7: storew */ - ptr0[i] = var37; - } - -} - -void -audiomixer_orc_add_volume_s16 (gint16 * ORC_RESTRICT d1, - const gint16 * ORC_RESTRICT s1, int p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 115, 49, 54, - 11, 2, 2, 12, 2, 2, 14, 4, 11, 0, 0, 0, 16, 2, 20, 4, - 20, 2, 176, 32, 4, 24, 125, 32, 32, 16, 165, 33, 32, 71, 0, 0, - 33, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_s16); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_volume_s16"); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_s16); - orc_program_add_destination (p, 2, "d1"); - orc_program_add_source (p, 2, "s1"); - orc_program_add_constant (p, 4, 0x0000000b, "c1"); - orc_program_add_parameter (p, 2, "p1"); - orc_program_add_temporary (p, 4, "t1"); - orc_program_add_temporary (p, 2, "t2"); - - orc_program_append_2 (p, "mulswl", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "shrsl", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "convssslw", 0, ORC_VAR_T2, ORC_VAR_T1, - ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "addssw", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - ex->params[ORC_VAR_P1] = p1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_volume_u32 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_volume_u32 (guint32 * ORC_RESTRICT d1, - const guint32 * ORC_RESTRICT s1, int p1, int n) -{ - int i; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var34; -#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__) - volatile orc_union32 var35; -#else - orc_union32 var35; -#endif - orc_union32 var36; - orc_union32 var37; - orc_union32 var38; - orc_union32 var39; - orc_union64 var40; - orc_union64 var41; - orc_union32 var42; - orc_union32 var43; - - ptr0 = (orc_union32 *) d1; - ptr4 = (orc_union32 *) s1; - - /* 1: loadpl */ - var35.i = (int) 0x80000000; /* -2147483648 or 1.061e-314f */ - /* 3: loadpl */ - var36.i = p1; - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var34 = ptr4[i]; - /* 2: xorl */ - var39.i = var34.i ^ var35.i; - /* 4: mulslq */ - var40.i = ((orc_int64) var39.i) * ((orc_int64) var36.i); - /* 5: shrsq */ - var41.i = var40.i >> 27; - /* 6: convsssql */ - var42.i = ORC_CLAMP_SL (var41.i); - /* 7: xorl */ - var43.i = var42.i ^ var35.i; - /* 8: loadl */ - var37 = ptr0[i]; - /* 9: addusl */ - var38.i = - ORC_CLAMP_UL ((orc_int64) (orc_uint32) var37.i + - (orc_int64) (orc_uint32) var43.i); - /* 10: storel */ - ptr0[i] = var38; - } - -} - -#else -static void -_backup_audiomixer_orc_add_volume_u32 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var34; -#if defined(__APPLE__) && __GNUC__ == 4 && __GNUC_MINOR__ == 2 && defined (__i386__) - volatile orc_union32 var35; -#else - orc_union32 var35; -#endif - orc_union32 var36; - orc_union32 var37; - orc_union32 var38; - orc_union32 var39; - orc_union64 var40; - orc_union64 var41; - orc_union32 var42; - orc_union32 var43; - - ptr0 = (orc_union32 *) ex->arrays[0]; - ptr4 = (orc_union32 *) ex->arrays[4]; - - /* 1: loadpl */ - var35.i = (int) 0x80000000; /* -2147483648 or 1.061e-314f */ - /* 3: loadpl */ - var36.i = ex->params[24]; - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var34 = ptr4[i]; - /* 2: xorl */ - var39.i = var34.i ^ var35.i; - /* 4: mulslq */ - var40.i = ((orc_int64) var39.i) * ((orc_int64) var36.i); - /* 5: shrsq */ - var41.i = var40.i >> 27; - /* 6: convsssql */ - var42.i = ORC_CLAMP_SL (var41.i); - /* 7: xorl */ - var43.i = var42.i ^ var35.i; - /* 8: loadl */ - var37 = ptr0[i]; - /* 9: addusl */ - var38.i = - ORC_CLAMP_UL ((orc_int64) (orc_uint32) var37.i + - (orc_int64) (orc_uint32) var43.i); - /* 10: storel */ - ptr0[i] = var38; - } - -} - -void -audiomixer_orc_add_volume_u32 (guint32 * ORC_RESTRICT d1, - const guint32 * ORC_RESTRICT s1, int p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 117, 51, 50, - 11, 4, 4, 12, 4, 4, 14, 4, 0, 0, 0, 128, 15, 8, 27, 0, - 0, 0, 0, 0, 0, 0, 16, 4, 20, 8, 20, 4, 132, 33, 4, 16, - 178, 32, 33, 24, 147, 32, 32, 17, 170, 33, 32, 132, 33, 33, 16, 105, - 0, 0, 33, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_u32); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_volume_u32"); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_u32); - orc_program_add_destination (p, 4, "d1"); - orc_program_add_source (p, 4, "s1"); - orc_program_add_constant (p, 4, 0x80000000, "c1"); - orc_program_add_constant_int64 (p, 8, 0x000000000000001bULL, "c2"); - orc_program_add_parameter (p, 4, "p1"); - orc_program_add_temporary (p, 8, "t1"); - orc_program_add_temporary (p, 4, "t2"); - - orc_program_append_2 (p, "xorl", 0, ORC_VAR_T2, ORC_VAR_S1, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "mulslq", 0, ORC_VAR_T1, ORC_VAR_T2, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "shrsq", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C2, - ORC_VAR_D1); - orc_program_append_2 (p, "convsssql", 0, ORC_VAR_T2, ORC_VAR_T1, - ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "xorl", 0, ORC_VAR_T2, ORC_VAR_T2, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "addusl", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - ex->params[ORC_VAR_P1] = p1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_volume_s32 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_volume_s32 (gint32 * ORC_RESTRICT d1, - const gint32 * ORC_RESTRICT s1, int p1, int n) -{ - int i; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var34; - orc_union32 var35; - orc_union32 var36; - orc_union32 var37; - orc_union64 var38; - orc_union64 var39; - orc_union32 var40; - - ptr0 = (orc_union32 *) d1; - ptr4 = (orc_union32 *) s1; - - /* 1: loadpl */ - var35.i = p1; - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var34 = ptr4[i]; - /* 2: mulslq */ - var38.i = ((orc_int64) var34.i) * ((orc_int64) var35.i); - /* 3: shrsq */ - var39.i = var38.i >> 27; - /* 4: convsssql */ - var40.i = ORC_CLAMP_SL (var39.i); - /* 5: loadl */ - var36 = ptr0[i]; - /* 6: addssl */ - var37.i = ORC_CLAMP_SL ((orc_int64) var36.i + (orc_int64) var40.i); - /* 7: storel */ - ptr0[i] = var37; - } - -} - -#else -static void -_backup_audiomixer_orc_add_volume_s32 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var34; - orc_union32 var35; - orc_union32 var36; - orc_union32 var37; - orc_union64 var38; - orc_union64 var39; - orc_union32 var40; - - ptr0 = (orc_union32 *) ex->arrays[0]; - ptr4 = (orc_union32 *) ex->arrays[4]; - - /* 1: loadpl */ - var35.i = ex->params[24]; - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var34 = ptr4[i]; - /* 2: mulslq */ - var38.i = ((orc_int64) var34.i) * ((orc_int64) var35.i); - /* 3: shrsq */ - var39.i = var38.i >> 27; - /* 4: convsssql */ - var40.i = ORC_CLAMP_SL (var39.i); - /* 5: loadl */ - var36 = ptr0[i]; - /* 6: addssl */ - var37.i = ORC_CLAMP_SL ((orc_int64) var36.i + (orc_int64) var40.i); - /* 7: storel */ - ptr0[i] = var37; - } - -} - -void -audiomixer_orc_add_volume_s32 (gint32 * ORC_RESTRICT d1, - const gint32 * ORC_RESTRICT s1, int p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 115, 51, 50, - 11, 4, 4, 12, 4, 4, 15, 8, 27, 0, 0, 0, 0, 0, 0, 0, - 16, 4, 20, 8, 20, 4, 178, 32, 4, 24, 147, 32, 32, 16, 170, 33, - 32, 104, 0, 0, 33, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_s32); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_volume_s32"); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_s32); - orc_program_add_destination (p, 4, "d1"); - orc_program_add_source (p, 4, "s1"); - orc_program_add_constant_int64 (p, 8, 0x000000000000001bULL, "c1"); - orc_program_add_parameter (p, 4, "p1"); - orc_program_add_temporary (p, 8, "t1"); - orc_program_add_temporary (p, 4, "t2"); - - orc_program_append_2 (p, "mulslq", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "shrsq", 0, ORC_VAR_T1, ORC_VAR_T1, ORC_VAR_C1, - ORC_VAR_D1); - orc_program_append_2 (p, "convsssql", 0, ORC_VAR_T2, ORC_VAR_T1, - ORC_VAR_D1, ORC_VAR_D1); - orc_program_append_2 (p, "addssl", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T2, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - ex->params[ORC_VAR_P1] = p1; - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_volume_f32 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_volume_f32 (float *ORC_RESTRICT d1, - const float *ORC_RESTRICT s1, float p1, int n) -{ - int i; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var33; - orc_union32 var34; - orc_union32 var35; - orc_union32 var36; - orc_union32 var37; - - ptr0 = (orc_union32 *) d1; - ptr4 = (orc_union32 *) s1; - - /* 1: loadpl */ - var34.f = p1; - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var33 = ptr4[i]; - /* 2: mulf */ - { - orc_union32 _src1; - orc_union32 _src2; - orc_union32 _dest1; - _src1.i = ORC_DENORMAL (var33.i); - _src2.i = ORC_DENORMAL (var34.i); - _dest1.f = _src1.f * _src2.f; - var37.i = ORC_DENORMAL (_dest1.i); - } - /* 3: loadl */ - var35 = ptr0[i]; - /* 4: addf */ - { - orc_union32 _src1; - orc_union32 _src2; - orc_union32 _dest1; - _src1.i = ORC_DENORMAL (var35.i); - _src2.i = ORC_DENORMAL (var37.i); - _dest1.f = _src1.f + _src2.f; - var36.i = ORC_DENORMAL (_dest1.i); - } - /* 5: storel */ - ptr0[i] = var36; - } - -} - -#else -static void -_backup_audiomixer_orc_add_volume_f32 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union32 *ORC_RESTRICT ptr0; - const orc_union32 *ORC_RESTRICT ptr4; - orc_union32 var33; - orc_union32 var34; - orc_union32 var35; - orc_union32 var36; - orc_union32 var37; - - ptr0 = (orc_union32 *) ex->arrays[0]; - ptr4 = (orc_union32 *) ex->arrays[4]; - - /* 1: loadpl */ - var34.i = ex->params[24]; - - for (i = 0; i < n; i++) { - /* 0: loadl */ - var33 = ptr4[i]; - /* 2: mulf */ - { - orc_union32 _src1; - orc_union32 _src2; - orc_union32 _dest1; - _src1.i = ORC_DENORMAL (var33.i); - _src2.i = ORC_DENORMAL (var34.i); - _dest1.f = _src1.f * _src2.f; - var37.i = ORC_DENORMAL (_dest1.i); - } - /* 3: loadl */ - var35 = ptr0[i]; - /* 4: addf */ - { - orc_union32 _src1; - orc_union32 _src2; - orc_union32 _dest1; - _src1.i = ORC_DENORMAL (var35.i); - _src2.i = ORC_DENORMAL (var37.i); - _dest1.f = _src1.f + _src2.f; - var36.i = ORC_DENORMAL (_dest1.i); - } - /* 5: storel */ - ptr0[i] = var36; - } - -} - -void -audiomixer_orc_add_volume_f32 (float *ORC_RESTRICT d1, - const float *ORC_RESTRICT s1, float p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 102, 51, 50, - 11, 4, 4, 12, 4, 4, 17, 4, 20, 4, 202, 32, 4, 24, 200, 0, - 0, 32, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_f32); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_volume_f32"); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_f32); - orc_program_add_destination (p, 4, "d1"); - orc_program_add_source (p, 4, "s1"); - orc_program_add_parameter_float (p, 4, "p1"); - orc_program_add_temporary (p, 4, "t1"); - - orc_program_append_2 (p, "mulf", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "addf", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - { - orc_union32 tmp; - tmp.f = p1; - ex->params[ORC_VAR_P1] = tmp.i; - } - - func = c->exec; - func (ex); -} -#endif - - -/* audiomixer_orc_add_volume_f64 */ -#ifdef DISABLE_ORC -void -audiomixer_orc_add_volume_f64 (double *ORC_RESTRICT d1, - const double *ORC_RESTRICT s1, double p1, int n) -{ - int i; - orc_union64 *ORC_RESTRICT ptr0; - const orc_union64 *ORC_RESTRICT ptr4; - orc_union64 var33; - orc_union64 var34; - orc_union64 var35; - orc_union64 var36; - orc_union64 var37; - - ptr0 = (orc_union64 *) d1; - ptr4 = (orc_union64 *) s1; - - /* 1: loadpq */ - var34.f = p1; - - for (i = 0; i < n; i++) { - /* 0: loadq */ - var33 = ptr4[i]; - /* 2: muld */ - { - orc_union64 _src1; - orc_union64 _src2; - orc_union64 _dest1; - _src1.i = ORC_DENORMAL_DOUBLE (var33.i); - _src2.i = ORC_DENORMAL_DOUBLE (var34.i); - _dest1.f = _src1.f * _src2.f; - var37.i = ORC_DENORMAL_DOUBLE (_dest1.i); - } - /* 3: loadq */ - var35 = ptr0[i]; - /* 4: addd */ - { - orc_union64 _src1; - orc_union64 _src2; - orc_union64 _dest1; - _src1.i = ORC_DENORMAL_DOUBLE (var35.i); - _src2.i = ORC_DENORMAL_DOUBLE (var37.i); - _dest1.f = _src1.f + _src2.f; - var36.i = ORC_DENORMAL_DOUBLE (_dest1.i); - } - /* 5: storeq */ - ptr0[i] = var36; - } - -} - -#else -static void -_backup_audiomixer_orc_add_volume_f64 (OrcExecutor * ORC_RESTRICT ex) -{ - int i; - int n = ex->n; - orc_union64 *ORC_RESTRICT ptr0; - const orc_union64 *ORC_RESTRICT ptr4; - orc_union64 var33; - orc_union64 var34; - orc_union64 var35; - orc_union64 var36; - orc_union64 var37; - - ptr0 = (orc_union64 *) ex->arrays[0]; - ptr4 = (orc_union64 *) ex->arrays[4]; - - /* 1: loadpq */ - var34.i = - (ex->params[24] & 0xffffffff) | ((orc_uint64) (ex->params[24 + - (ORC_VAR_T1 - ORC_VAR_P1)]) << 32); - - for (i = 0; i < n; i++) { - /* 0: loadq */ - var33 = ptr4[i]; - /* 2: muld */ - { - orc_union64 _src1; - orc_union64 _src2; - orc_union64 _dest1; - _src1.i = ORC_DENORMAL_DOUBLE (var33.i); - _src2.i = ORC_DENORMAL_DOUBLE (var34.i); - _dest1.f = _src1.f * _src2.f; - var37.i = ORC_DENORMAL_DOUBLE (_dest1.i); - } - /* 3: loadq */ - var35 = ptr0[i]; - /* 4: addd */ - { - orc_union64 _src1; - orc_union64 _src2; - orc_union64 _dest1; - _src1.i = ORC_DENORMAL_DOUBLE (var35.i); - _src2.i = ORC_DENORMAL_DOUBLE (var37.i); - _dest1.f = _src1.f + _src2.f; - var36.i = ORC_DENORMAL_DOUBLE (_dest1.i); - } - /* 5: storeq */ - ptr0[i] = var36; - } - -} - -void -audiomixer_orc_add_volume_f64 (double *ORC_RESTRICT d1, - const double *ORC_RESTRICT s1, double p1, int n) -{ - OrcExecutor _ex, *ex = &_ex; - static volatile int p_inited = 0; - static OrcCode *c = 0; - void (*func) (OrcExecutor *); - - if (!p_inited) { - orc_once_mutex_lock (); - if (!p_inited) { - OrcProgram *p; - -#if 1 - static const orc_uint8 bc[] = { - 1, 9, 29, 97, 117, 100, 105, 111, 109, 105, 120, 101, 114, 95, 111, 114, - 99, 95, 97, 100, 100, 95, 118, 111, 108, 117, 109, 101, 95, 102, 54, 52, - 11, 8, 8, 12, 8, 8, 18, 8, 20, 8, 214, 32, 4, 24, 212, 0, - 0, 32, 2, 0, - }; - p = orc_program_new_from_static_bytecode (bc); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_f64); -#else - p = orc_program_new (); - orc_program_set_name (p, "audiomixer_orc_add_volume_f64"); - orc_program_set_backup_function (p, - _backup_audiomixer_orc_add_volume_f64); - orc_program_add_destination (p, 8, "d1"); - orc_program_add_source (p, 8, "s1"); - orc_program_add_parameter_double (p, 8, "p1"); - orc_program_add_temporary (p, 8, "t1"); - - orc_program_append_2 (p, "muld", 0, ORC_VAR_T1, ORC_VAR_S1, ORC_VAR_P1, - ORC_VAR_D1); - orc_program_append_2 (p, "addd", 0, ORC_VAR_D1, ORC_VAR_D1, ORC_VAR_T1, - ORC_VAR_D1); -#endif - - orc_program_compile (p); - c = orc_program_take_code (p); - orc_program_free (p); - } - p_inited = TRUE; - orc_once_mutex_unlock (); - } - ex->arrays[ORC_VAR_A2] = c; - ex->program = 0; - - ex->n = n; - ex->arrays[ORC_VAR_D1] = d1; - ex->arrays[ORC_VAR_S1] = (void *) s1; - { - orc_union64 tmp; - tmp.f = p1; - ex->params[ORC_VAR_P1] = ((orc_uint64) tmp.i) & 0xffffffff; - ex->params[ORC_VAR_T1] = ((orc_uint64) tmp.i) >> 32; - } - - func = c->exec; - func (ex); -} -#endif diff --git a/gst/audiomixer/gstaudiomixerorc-dist.h b/gst/audiomixer/gstaudiomixerorc-dist.h deleted file mode 100644 index af0de01394..0000000000 --- a/gst/audiomixer/gstaudiomixerorc-dist.h +++ /dev/null @@ -1,106 +0,0 @@ - -/* autogenerated from gstaudiomixerorc.orc */ - -#ifndef _GSTAUDIOMIXERORC_H_ -#define _GSTAUDIOMIXERORC_H_ - -#include - -#ifdef __cplusplus -extern "C" { -#endif - - - -#ifndef _ORC_INTEGER_TYPEDEFS_ -#define _ORC_INTEGER_TYPEDEFS_ -#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L -#include -typedef int8_t orc_int8; -typedef int16_t orc_int16; -typedef int32_t orc_int32; -typedef int64_t orc_int64; -typedef uint8_t orc_uint8; -typedef uint16_t orc_uint16; -typedef uint32_t orc_uint32; -typedef uint64_t orc_uint64; -#define ORC_UINT64_C(x) UINT64_C(x) -#elif defined(_MSC_VER) -typedef signed __int8 orc_int8; -typedef signed __int16 orc_int16; -typedef signed __int32 orc_int32; -typedef signed __int64 orc_int64; -typedef unsigned __int8 orc_uint8; -typedef unsigned __int16 orc_uint16; -typedef unsigned __int32 orc_uint32; -typedef unsigned __int64 orc_uint64; -#define ORC_UINT64_C(x) (x##Ui64) -#define inline __inline -#else -#include -typedef signed char orc_int8; -typedef short orc_int16; -typedef int orc_int32; -typedef unsigned char orc_uint8; -typedef unsigned short orc_uint16; -typedef unsigned int orc_uint32; -#if INT_MAX == LONG_MAX -typedef long long orc_int64; -typedef unsigned long long orc_uint64; -#define ORC_UINT64_C(x) (x##ULL) -#else -typedef long orc_int64; -typedef unsigned long orc_uint64; -#define ORC_UINT64_C(x) (x##UL) -#endif -#endif -typedef union { orc_int16 i; orc_int8 x2[2]; } orc_union16; -typedef union { orc_int32 i; float f; orc_int16 x2[2]; orc_int8 x4[4]; } orc_union32; -typedef union { orc_int64 i; double f; orc_int32 x2[2]; float x2f[2]; orc_int16 x4[4]; } orc_union64; -#endif -#ifndef ORC_RESTRICT -#if defined(__STDC_VERSION__) && __STDC_VERSION__ >= 199901L -#define ORC_RESTRICT restrict -#elif defined(__GNUC__) && __GNUC__ >= 4 -#define ORC_RESTRICT __restrict__ -#else -#define ORC_RESTRICT -#endif -#endif - -#ifndef ORC_INTERNAL -#if defined(__SUNPRO_C) && (__SUNPRO_C >= 0x590) -#define ORC_INTERNAL __attribute__((visibility("hidden"))) -#elif defined(__SUNPRO_C) && (__SUNPRO_C >= 0x550) -#define ORC_INTERNAL __hidden -#elif defined (__GNUC__) -#define ORC_INTERNAL __attribute__((visibility("hidden"))) -#else -#define ORC_INTERNAL -#endif -#endif - -void audiomixer_orc_add_s32 (gint32 * ORC_RESTRICT d1, const gint32 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_s16 (gint16 * ORC_RESTRICT d1, const gint16 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_s8 (gint8 * ORC_RESTRICT d1, const gint8 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_u32 (guint32 * ORC_RESTRICT d1, const guint32 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_u16 (guint16 * ORC_RESTRICT d1, const guint16 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_f32 (float * ORC_RESTRICT d1, const float * ORC_RESTRICT s1, int n); -void audiomixer_orc_add_f64 (double * ORC_RESTRICT d1, const double * ORC_RESTRICT s1, int n); -void audiomixer_orc_volume_u8 (guint8 * ORC_RESTRICT d1, int p1, int n); -void audiomixer_orc_add_volume_u8 (guint8 * ORC_RESTRICT d1, const guint8 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_s8 (gint8 * ORC_RESTRICT d1, const gint8 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_u16 (guint16 * ORC_RESTRICT d1, const guint16 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_s16 (gint16 * ORC_RESTRICT d1, const gint16 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_u32 (guint32 * ORC_RESTRICT d1, const guint32 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_s32 (gint32 * ORC_RESTRICT d1, const gint32 * ORC_RESTRICT s1, int p1, int n); -void audiomixer_orc_add_volume_f32 (float * ORC_RESTRICT d1, const float * ORC_RESTRICT s1, float p1, int n); -void audiomixer_orc_add_volume_f64 (double * ORC_RESTRICT d1, const double * ORC_RESTRICT s1, double p1, int n); - -#ifdef __cplusplus -} -#endif - -#endif - diff --git a/gst/audiomixer/gstaudiomixerorc.orc b/gst/audiomixer/gstaudiomixerorc.orc deleted file mode 100644 index 5eaff2395b..0000000000 --- a/gst/audiomixer/gstaudiomixerorc.orc +++ /dev/null @@ -1,176 +0,0 @@ -.function audiomixer_orc_add_s32 -.dest 4 d1 gint32 -.source 4 s1 gint32 - -addssl d1, d1, s1 - - -.function audiomixer_orc_add_s16 -.dest 2 d1 gint16 -.source 2 s1 gint16 - -addssw d1, d1, s1 - - -.function audiomixer_orc_add_s8 -.dest 1 d1 gint8 -.source 1 s1 gint8 - -addssb d1, d1, s1 - - -.function audiomixer_orc_add_u32 -.dest 4 d1 guint32 -.source 4 s1 guint32 - -addusl d1, d1, s1 - - -.function audiomixer_orc_add_u16 -.dest 2 d1 guint16 -.source 2 s1 guint16 - -addusw d1, d1, s1 - - -.function audiomixer_orc_add_u8 -.dest 1 d1 guint8 -.source 1 s1 guint8 - -addusb d1, d1, s1 - - -.function audiomixer_orc_add_f32 -.dest 4 d1 float -.source 4 s1 float - -addf d1, d1, s1 - -.function audiomixer_orc_add_f64 -.dest 8 d1 double -.source 8 s1 double - -addd d1, d1, s1 - - -.function audiomixer_orc_volume_u8 -.dest 1 d1 guint8 -.param 1 p1 -.const 1 c1 0x80 -.temp 2 t1 -.temp 1 t2 - -xorb t2, d1, c1 -mulsbw t1, t2, p1 -shrsw t1, t1, 3 -convssswb t2, t1 -xorb d1, t2, c1 - - -.function audiomixer_orc_add_volume_u8 -.dest 1 d1 guint8 -.source 1 s1 guint8 -.param 1 p1 -.const 1 c1 0x80 -.temp 2 t1 -.temp 1 t2 - -xorb t2, s1, c1 -mulsbw t1, t2, p1 -shrsw t1, t1, 3 -convssswb t2, t1 -xorb t2, t2, c1 -addusb d1, d1, t2 - - -.function audiomixer_orc_add_volume_s8 -.dest 1 d1 gint8 -.source 1 s1 gint8 -.param 1 p1 -.temp 2 t1 -.temp 1 t2 - -mulsbw t1, s1, p1 -shrsw t1, t1, 3 -convssswb t2, t1 -addssb d1, d1, t2 - - -.function audiomixer_orc_add_volume_u16 -.dest 2 d1 guint16 -.source 2 s1 guint16 -.param 2 p1 -.const 2 c1 0x8000 -.temp 4 t1 -.temp 2 t2 - -xorw t2, s1, c1 -mulswl t1, t2, p1 -shrsl t1, t1, 11 -convssslw t2, t1 -xorw t2, t2, c1 -addusw d1, d1, t2 - - -.function audiomixer_orc_add_volume_s16 -.dest 2 d1 gint16 -.source 2 s1 gint16 -.param 2 p1 -.temp 4 t1 -.temp 2 t2 - -mulswl t1, s1, p1 -shrsl t1, t1, 11 -convssslw t2, t1 -addssw d1, d1, t2 - - -.function audiomixer_orc_add_volume_u32 -.dest 4 d1 guint32 -.source 4 s1 guint32 -.param 4 p1 -.const 4 c1 0x80000000 -.temp 8 t1 -.temp 4 t2 - -xorl t2, s1, c1 -mulslq t1, t2, p1 -shrsq t1, t1, 27 -convsssql t2, t1 -xorl t2, t2, c1 -addusl d1, d1, t2 - - -.function audiomixer_orc_add_volume_s32 -.dest 4 d1 gint32 -.source 4 s1 gint32 -.param 4 p1 -.temp 8 t1 -.temp 4 t2 - -mulslq t1, s1, p1 -shrsq t1, t1, 27 -convsssql t2, t1 -addssl d1, d1, t2 - - -.function audiomixer_orc_add_volume_f32 -.dest 4 d1 float -.source 4 s1 float -.floatparam 4 p1 -.temp 4 t1 - -mulf t1, s1, p1 -addf d1, d1, t1 - - -.function audiomixer_orc_add_volume_f64 -.dest 8 d1 double -.source 8 s1 double -.doubleparam 8 p1 -.temp 8 t1 - -muld t1, s1, p1 -addd d1, d1, t1 - - diff --git a/gst/audiomixer/meson.build b/gst/audiomixer/meson.build deleted file mode 100644 index ccfe1b9d37..0000000000 --- a/gst/audiomixer/meson.build +++ /dev/null @@ -1,32 +0,0 @@ -audiomixer_sources = [ - 'gstaudiomixer.c', - 'gstaudiointerleave.c', -] - -orcsrc = 'gstaudiomixerorc' -if have_orcc - orc_h = custom_target(orcsrc + '.h', - input : orcsrc + '.orc', - output : orcsrc + '.h', - command : orcc_args + ['--header', '-o', '@OUTPUT@', '@INPUT@']) - orc_c = custom_target(orcsrc + '.c', - input : orcsrc + '.orc', - output : orcsrc + '.c', - command : orcc_args + ['--implementation', '-o', '@OUTPUT@', '@INPUT@']) -else - orc_h = configure_file(input : orcsrc + '-dist.h', - output : orcsrc + '.h', - configuration : configuration_data()) - orc_c = configure_file(input : orcsrc + '-dist.c', - output : orcsrc + '.c', - configuration : configuration_data()) -endif - -gstaudiomixer = library('gstaudiomixer', - audiomixer_sources, orc_c, orc_h, - c_args : gst_plugins_bad_args + [ '-DGST_USE_UNSTABLE_API' ], - include_directories : [configinc], - dependencies : [gstbadaudio_dep, gstaudio_dep, gstbase_dep, orc_dep], - install : true, - install_dir : plugins_install_dir, -) diff --git a/gst/meson.build b/gst/meson.build index b56b68d263..3eff27e552 100644 --- a/gst/meson.build +++ b/gst/meson.build @@ -5,7 +5,6 @@ subdir('aiff') subdir('asfmux') subdir('audiobuffersplit') subdir('audiofxbad') -subdir('audiomixer') subdir('audiomixmatrix') subdir('audiovisualizers') subdir('autoconvert') diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 0343da7eb3..5a7c497b28 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -127,7 +127,7 @@ check_kate= endif if HAVE_ORC -check_orc = orc/bayer orc/audiomixer orc/compositor +check_orc = orc/bayer orc/compositor else check_orc = endif @@ -257,8 +257,6 @@ check_PROGRAMS = \ elements/videoframe-audiolevel \ elements/autoconvert \ elements/autovideoconvert \ - elements/audiointerleave \ - elements/audiomixer \ elements/asfmux \ elements/camerabin \ elements/gdppay \ @@ -313,12 +311,6 @@ LDADD = $(GST_CHECK_LIBS) generic_states_CFLAGS = $(AM_CFLAGS) $(GLIB_CFLAGS) generic_states_LDADD = $(LDADD) $(GLIB_LIBS) -elements_audiomixer_LDADD = $(GST_BASE_LIBS) $(GST_CONTROLLER_LIBS) -lgstbase-@GST_API_VERSION@ $(LDADD) -elements_audiomixer_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CONTROLLER_CFLAGS) $(AM_CFLAGS) - -elements_audiointerleave_LDADD = $(GST_BASE_LIBS) -lgstbase-@GST_API_VERSION@ $(GST_AUDIO_LIBS) $(LDADD) -elements_audiointerleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(AM_CFLAGS) - elements_pnm_CFLAGS = \ $(GST_PLUGINS_BASE_CFLAGS) \ $(GST_BASE_CFLAGS) $(GST_CFLAGS) $(AM_CFLAGS) @@ -542,14 +534,6 @@ orc/bayer.c: $(top_srcdir)/gst/bayer/gstbayerorc.orc $(MKDIR_P) orc $(ORCC) --test -o $@ $< -orc_audiomixer_CFLAGS = $(ORC_CFLAGS) -orc_audiomixer_LDADD = $(ORC_LIBS) -lorc-test-0.4 -nodist_orc_audiomixer_SOURCES = orc/audiomixer.c - -orc/audiomixer.c: $(top_srcdir)/gst/audiomixer/gstaudiomixerorc.orc - $(MKDIR_P) orc - $(ORCC) --test -o $@ $< - elements_compositor_LDADD = \ $(GST_PLUGINS_BASE_LIBS) $(GST_VIDEO_LIBS) $(GST_BASE_LIBS) $(LDADD) elements_compositor_CFLAGS = \ diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore index 741772b5a0..d264dae579 100644 --- a/tests/check/elements/.gitignore +++ b/tests/check/elements/.gitignore @@ -2,8 +2,6 @@ aiffparse asfmux assrender -audiointerleave -audiomixer autoconvert autovideoconvert baseaudiovisualizer diff --git a/tests/check/elements/audiointerleave.c b/tests/check/elements/audiointerleave.c deleted file mode 100644 index 71348f4592..0000000000 --- a/tests/check/elements/audiointerleave.c +++ /dev/null @@ -1,1128 +0,0 @@ -/* GStreamer unit tests for the audiointerleave element - * Copyright (C) 2007 Tim-Philipp Müller - * Copyright (C) 2008 Sebastian Dröge - * - * 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. - */ - -/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray - * with newer GLib versions (>= 2.31.0) */ -#define GLIB_DISABLE_DEPRECATION_WARNINGS - -#ifdef HAVE_CONFIG_H -# include "config.h" -#endif - -#ifdef HAVE_VALGRIND -# include -#endif - -#include -#include -#include - -#include - -static void -gst_check_setup_events_audiointerleave (GstPad * srcpad, GstElement * element, - GstCaps * caps, GstFormat format, const gchar * stream_id) -{ - GstSegment segment; - - gst_segment_init (&segment, format); - - fail_unless (gst_pad_push_event (srcpad, - gst_event_new_stream_start (stream_id))); - if (caps) - fail_unless (gst_pad_push_event (srcpad, gst_event_new_caps (caps))); - fail_unless (gst_pad_push_event (srcpad, gst_event_new_segment (&segment))); -} - -GST_START_TEST (test_create_and_unref) -{ - GstElement *interleave; - - interleave = gst_element_factory_make ("audiointerleave", NULL); - fail_unless (interleave != NULL); - - gst_element_set_state (interleave, GST_STATE_NULL); - gst_object_unref (interleave); -} - -GST_END_TEST; - -GST_START_TEST (test_request_pads) -{ - GstElement *interleave; - GstPad *pad1, *pad2; - - interleave = gst_element_factory_make ("audiointerleave", NULL); - fail_unless (interleave != NULL); - - pad1 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (pad1 != NULL); - fail_unless_equals_string (GST_OBJECT_NAME (pad1), "sink_0"); - - pad2 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (pad2 != NULL); - fail_unless_equals_string (GST_OBJECT_NAME (pad2), "sink_1"); - - gst_element_release_request_pad (interleave, pad2); - gst_object_unref (pad2); - gst_element_release_request_pad (interleave, pad1); - gst_object_unref (pad1); - - gst_element_set_state (interleave, GST_STATE_NULL); - gst_object_unref (interleave); -} - -GST_END_TEST; - -static GstPad **mysrcpads, *mysinkpad; -static GstBus *bus; -static GstElement *interleave; -static GMutex data_mutex; -static GCond data_cond; -static gint have_data; -static gfloat input[2]; - -static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw, " - "format = (string) " GST_AUDIO_NE (F32) ", " - "channels = (int) 2, layout = (string) {interleaved, non-interleaved}, rate = (int) 48000")); - -static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw, " - "format = (string) " GST_AUDIO_NE (F32) ", " - "channels = (int) 1, layout = (string) interleaved, rate = (int) 48000")); - -#define CAPS_48khz \ - "audio/x-raw, " \ - "format = (string) " GST_AUDIO_NE (F32) ", " \ - "channels = (int) 1, layout = (string) non-interleaved," \ - "rate = (int) 48000" - -static GstFlowReturn -interleave_chain_func (GstPad * pad, GstObject * parent, GstBuffer * buffer) -{ - GstMapInfo map; - gfloat *outdata; - gint i; - - fail_unless (GST_IS_BUFFER (buffer)); - fail_unless (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)); - gst_buffer_map (buffer, &map, GST_MAP_READ); - outdata = (gfloat *) map.data; - fail_unless (outdata != NULL); - -#ifdef HAVE_VALGRIND - if (!(RUNNING_ON_VALGRIND)) -#endif - for (i = 0; i < map.size / sizeof (float); i += 2) { - fail_unless_equals_float (outdata[i], input[0]); - fail_unless_equals_float (outdata[i + 1], input[1]); - } - - g_mutex_lock (&data_mutex); - have_data += map.size; - g_cond_signal (&data_cond); - g_mutex_unlock (&data_mutex); - - gst_buffer_unmap (buffer, &map); - gst_buffer_unref (buffer); - - - return GST_FLOW_OK; -} - -GST_START_TEST (test_audiointerleave_2ch) -{ - GstElement *queue; - GstPad *sink0, *sink1, *src, *tmp; - GstCaps *caps; - gint i; - GstBuffer *inbuf; - gfloat *indata; - GstMapInfo map; - - mysrcpads = g_new0 (GstPad *, 2); - - have_data = 0; - - interleave = gst_element_factory_make ("audiointerleave", NULL); - fail_unless (interleave != NULL); - - g_object_set (interleave, "latency", GST_SECOND / 4, NULL); - - queue = gst_element_factory_make ("queue", "queue"); - fail_unless (queue != NULL); - - sink0 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sink0 != NULL); - fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0"); - - sink1 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sink1 != NULL); - fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1"); - - mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0"); - fail_unless (mysrcpads[0] != NULL); - - caps = gst_caps_from_string (CAPS_48khz); - gst_pad_set_active (mysrcpads[0], TRUE); - gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps, - GST_FORMAT_TIME, "0"); - gst_pad_use_fixed_caps (mysrcpads[0]); - - mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1"); - fail_unless (mysrcpads[1] != NULL); - - gst_pad_set_active (mysrcpads[1], TRUE); - gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps, - GST_FORMAT_TIME, "1"); - gst_pad_use_fixed_caps (mysrcpads[1]); - - tmp = gst_element_get_static_pad (queue, "sink"); - fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - tmp = gst_element_get_static_pad (queue, "src"); - fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK); - - mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); - fail_unless (mysinkpad != NULL); - gst_pad_set_chain_function (mysinkpad, interleave_chain_func); - gst_pad_set_active (mysinkpad, TRUE); - - src = gst_element_get_static_pad (interleave, "src"); - fail_unless (src != NULL); - fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK); - gst_object_unref (src); - - bus = gst_bus_new (); - gst_element_set_bus (interleave, bus); - - fail_unless (gst_element_set_state (interleave, - GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); - fail_unless (gst_element_set_state (queue, - GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); - - input[0] = -1.0; - inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); - //GST_BUFFER_PTS (inbuf) = 0; - gst_buffer_map (inbuf, &map, GST_MAP_WRITE); - indata = (gfloat *) map.data; - for (i = 0; i < 48000; i++) - indata[i] = -1.0; - gst_buffer_unmap (inbuf, &map); - fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK); - - input[1] = 1.0; - inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); - //GST_BUFFER_PTS (inbuf) = 0; - gst_buffer_map (inbuf, &map, GST_MAP_WRITE); - indata = (gfloat *) map.data; - for (i = 0; i < 48000; i++) - indata[i] = 1.0; - gst_buffer_unmap (inbuf, &map); - fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK); - - inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); - //GST_BUFFER_PTS (inbuf) = GST_SECOND; - gst_buffer_map (inbuf, &map, GST_MAP_WRITE); - indata = (gfloat *) map.data; - for (i = 0; i < 48000; i++) - indata[i] = -1.0; - gst_buffer_unmap (inbuf, &map); - fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK); - - inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); - //GST_BUFFER_PTS (inbuf) = GST_SECOND; - gst_buffer_map (inbuf, &map, GST_MAP_WRITE); - indata = (gfloat *) map.data; - for (i = 0; i < 48000; i++) - indata[i] = 1.0; - gst_buffer_unmap (inbuf, &map); - fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK); - - g_mutex_lock (&data_mutex); - while (have_data < 48000 * 2 * 2 * sizeof (float)) - g_cond_wait (&data_cond, &data_mutex); - g_mutex_unlock (&data_mutex); - - gst_bus_set_flushing (bus, TRUE); - gst_element_set_state (interleave, GST_STATE_NULL); - gst_element_set_state (queue, GST_STATE_NULL); - - gst_object_unref (mysrcpads[0]); - gst_object_unref (mysrcpads[1]); - gst_object_unref (mysinkpad); - - gst_element_release_request_pad (interleave, sink0); - gst_object_unref (sink0); - gst_element_release_request_pad (interleave, sink1); - gst_object_unref (sink1); - - gst_object_unref (interleave); - gst_object_unref (queue); - gst_object_unref (bus); - gst_caps_unref (caps); - - g_free (mysrcpads); -} - -GST_END_TEST; - -GST_START_TEST (test_audiointerleave_2ch_1eos) -{ - GstElement *queue; - GstPad *sink0, *sink1, *src, *tmp; - GstCaps *caps; - gint i; - GstBuffer *inbuf; - gfloat *indata; - GstMapInfo map; - - mysrcpads = g_new0 (GstPad *, 2); - - have_data = 0; - - interleave = gst_element_factory_make ("audiointerleave", NULL); - fail_unless (interleave != NULL); - - g_object_set (interleave, "latency", GST_SECOND / 4, NULL); - - queue = gst_element_factory_make ("queue", "queue"); - fail_unless (queue != NULL); - - sink0 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sink0 != NULL); - fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink_0"); - - sink1 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sink1 != NULL); - fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink_1"); - - mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0"); - fail_unless (mysrcpads[0] != NULL); - - caps = gst_caps_from_string (CAPS_48khz); - gst_pad_set_active (mysrcpads[0], TRUE); - gst_check_setup_events_audiointerleave (mysrcpads[0], interleave, caps, - GST_FORMAT_TIME, "0"); - gst_pad_use_fixed_caps (mysrcpads[0]); - - mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1"); - fail_unless (mysrcpads[1] != NULL); - - gst_pad_set_active (mysrcpads[1], TRUE); - gst_check_setup_events_audiointerleave (mysrcpads[1], interleave, caps, - GST_FORMAT_TIME, "1"); - gst_pad_use_fixed_caps (mysrcpads[1]); - - tmp = gst_element_get_static_pad (queue, "sink"); - fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - tmp = gst_element_get_static_pad (queue, "src"); - fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK); - - mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink"); - fail_unless (mysinkpad != NULL); - gst_pad_set_chain_function (mysinkpad, interleave_chain_func); - gst_pad_set_active (mysinkpad, TRUE); - - src = gst_element_get_static_pad (interleave, "src"); - fail_unless (src != NULL); - fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK); - gst_object_unref (src); - - bus = gst_bus_new (); - gst_element_set_bus (interleave, bus); - - fail_unless (gst_element_set_state (interleave, - GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); - fail_unless (gst_element_set_state (queue, - GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS); - - input[0] = -1.0; - inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); - GST_BUFFER_PTS (inbuf) = 0; - gst_buffer_map (inbuf, &map, GST_MAP_WRITE); - indata = (gfloat *) map.data; - for (i = 0; i < 48000; i++) - indata[i] = -1.0; - gst_buffer_unmap (inbuf, &map); - fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK); - - input[1] = 1.0; - inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); - GST_BUFFER_PTS (inbuf) = 0; - gst_buffer_map (inbuf, &map, GST_MAP_WRITE); - indata = (gfloat *) map.data; - for (i = 0; i < 48000; i++) - indata[i] = 1.0; - gst_buffer_unmap (inbuf, &map); - fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK); - - g_mutex_lock (&data_mutex); - /* 48000 samples per buffer * 2 sources * 2 buffers */ - while (have_data != 48000 * 2 * sizeof (float)) - g_cond_wait (&data_cond, &data_mutex); - g_mutex_unlock (&data_mutex); - - input[0] = 0.0; - gst_pad_push_event (mysrcpads[0], gst_event_new_eos ()); - - input[1] = 1.0; - inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat)); - GST_BUFFER_PTS (inbuf) = GST_SECOND; - gst_buffer_map (inbuf, &map, GST_MAP_WRITE); - indata = (gfloat *) map.data; - for (i = 0; i < 48000; i++) - indata[i] = 1.0; - gst_buffer_unmap (inbuf, &map); - fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK); - - g_mutex_lock (&data_mutex); - /* 48000 samples per buffer * 2 sources * 2 buffers */ - while (have_data != 48000 * 2 * 2 * sizeof (float)) - g_cond_wait (&data_cond, &data_mutex); - g_mutex_unlock (&data_mutex); - - gst_bus_set_flushing (bus, TRUE); - gst_element_set_state (interleave, GST_STATE_NULL); - gst_element_set_state (queue, GST_STATE_NULL); - - gst_object_unref (mysrcpads[0]); - gst_object_unref (mysrcpads[1]); - gst_object_unref (mysinkpad); - - gst_element_release_request_pad (interleave, sink0); - gst_object_unref (sink0); - gst_element_release_request_pad (interleave, sink1); - gst_object_unref (sink1); - - gst_object_unref (interleave); - gst_object_unref (queue); - gst_object_unref (bus); - gst_caps_unref (caps); - - g_free (mysrcpads); -} - -GST_END_TEST; - -static void -src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad, - gboolean interleaved, gpointer user_data) -{ - gint n = GPOINTER_TO_INT (user_data); - gfloat *data; - gint i, num_samples; - GstCaps *caps; - guint64 mask; - GstAudioChannelPosition pos; - GstMapInfo map; - - fail_unless (gst_buffer_is_writable (buffer)); - - switch (n) { - case 0: - case 1: - case 2: - pos = GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT; - break; - case 3: - pos = GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT; - break; - default: - pos = GST_AUDIO_CHANNEL_POSITION_INVALID; - break; - } - - mask = G_GUINT64_CONSTANT (1) << pos; - - caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, GST_AUDIO_NE (F32), - "channels", G_TYPE_INT, 1, - "layout", G_TYPE_STRING, interleaved ? "interleaved" : "non-interleaved", - "channel-mask", GST_TYPE_BITMASK, mask, "rate", G_TYPE_INT, 48000, NULL); - - gst_pad_set_caps (pad, caps); - gst_caps_unref (caps); - - fail_unless (gst_buffer_map (buffer, &map, GST_MAP_WRITE)); - fail_unless (map.size % sizeof (gfloat) == 0); - - fail_unless (map.size > 480); - - num_samples = map.size / sizeof (gfloat); - data = (gfloat *) map.data; - - for (i = 0; i < num_samples; i++) - data[i] = (n % 2 == 0) ? -1.0 : 1.0; - - gst_buffer_unmap (buffer, &map); -} - -static void -src_handoff_float32_audiointerleaved (GstElement * element, GstBuffer * buffer, - GstPad * pad, gpointer user_data) -{ - src_handoff_float32 (element, buffer, pad, TRUE, user_data); -} - -static void -src_handoff_float32_non_audiointerleaved (GstElement * element, - GstBuffer * buffer, GstPad * pad, gpointer user_data) -{ - src_handoff_float32 (element, buffer, pad, FALSE, user_data); -} - -static void -sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad, - gpointer user_data) -{ - gint i; - GstMapInfo map; - gfloat *data; - GstCaps *caps, *ccaps; - gint n = GPOINTER_TO_INT (user_data); - guint64 mask; - - fail_unless (GST_IS_BUFFER (buffer)); - gst_buffer_map (buffer, &map, GST_MAP_READ); - data = (gfloat *) map.data; - - /* Give a little leeway for rounding errors */ - fail_unless (gst_util_uint64_scale (map.size, GST_SECOND, - 48000 * 2 * sizeof (gfloat)) <= GST_BUFFER_DURATION (buffer) + 1 || - gst_util_uint64_scale (map.size, GST_SECOND, - 48000 * 2 * sizeof (gfloat)) >= GST_BUFFER_DURATION (buffer) - 1); - - if (n == 0 || n == 3) { - GstAudioChannelPosition pos[2] = - { GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE }; - gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask); - } else if (n == 1) { - GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT, - GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT - }; - gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask); - } else if (n == 2) { - GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER, - GST_AUDIO_CHANNEL_POSITION_REAR_CENTER - }; - gst_audio_channel_positions_to_mask (pos, 2, FALSE, &mask); - } else { - g_assert_not_reached (); - } - - if (pad) { - caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, GST_AUDIO_NE (F32), - "channels", G_TYPE_INT, 2, "rate", G_TYPE_INT, 48000, - "layout", G_TYPE_STRING, "interleaved", - "channel-mask", GST_TYPE_BITMASK, mask, NULL); - - ccaps = gst_pad_get_current_caps (pad); - fail_unless (gst_caps_is_equal (caps, ccaps)); - gst_caps_unref (ccaps); - gst_caps_unref (caps); - } -#ifdef HAVE_VALGRIND - if (!(RUNNING_ON_VALGRIND)) -#endif - for (i = 0; i < map.size / sizeof (float); i += 2) { - fail_unless_equals_float (data[i], -1.0); - if (n != 3) - fail_unless_equals_float (data[i + 1], 1.0); - } - have_data += map.size; - - gst_buffer_unmap (buffer, &map); - -} - -static void -test_audiointerleave_2ch_pipeline (gboolean interleaved) -{ - GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink; - GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2; - GstMessage *msg; - void *src_handoff_float32 = - interleaved ? &src_handoff_float32_audiointerleaved : - &src_handoff_float32_non_audiointerleaved; - - have_data = 0; - - pipeline = (GstElement *) gst_pipeline_new ("pipeline"); - fail_unless (pipeline != NULL); - - src1 = gst_element_factory_make ("fakesrc", "src1"); - fail_unless (src1 != NULL); - g_object_set (src1, "num-buffers", 4, NULL); - g_object_set (src1, "sizetype", 2, - "sizemax", (int) 48000 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_object_set (src1, "signal-handoffs", TRUE, NULL); - g_object_set (src1, "format", GST_FORMAT_TIME, NULL); - g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32), - GINT_TO_POINTER (0)); - gst_bin_add (GST_BIN (pipeline), src1); - - src2 = gst_element_factory_make ("fakesrc", "src2"); - fail_unless (src2 != NULL); - g_object_set (src2, "num-buffers", 4, NULL); - g_object_set (src2, "sizetype", 2, - "sizemax", (int) 48000 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_object_set (src2, "signal-handoffs", TRUE, NULL); - g_object_set (src2, "format", GST_FORMAT_TIME, NULL); - g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32), - GINT_TO_POINTER (1)); - gst_bin_add (GST_BIN (pipeline), src2); - - queue = gst_element_factory_make ("queue", "queue"); - fail_unless (queue != NULL); - gst_bin_add (GST_BIN (pipeline), queue); - - interleave = gst_element_factory_make ("audiointerleave", "audiointerleave"); - fail_unless (interleave != NULL); - gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave)); - - sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sinkpad0 != NULL); - tmp = gst_element_get_static_pad (src1, "src"); - fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sinkpad1 != NULL); - tmp = gst_element_get_static_pad (src2, "src"); - tmp2 = gst_element_get_static_pad (queue, "sink"); - fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - gst_object_unref (tmp2); - tmp = gst_element_get_static_pad (queue, "src"); - fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - sink = gst_element_factory_make ("fakesink", "sink"); - fail_unless (sink != NULL); - g_object_set (sink, "signal-handoffs", TRUE, NULL); - g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32), - GINT_TO_POINTER (0)); - gst_bin_add (GST_BIN (pipeline), sink); - tmp = gst_element_get_static_pad (interleave, "src"); - tmp2 = gst_element_get_static_pad (sink, "sink"); - fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - gst_object_unref (tmp2); - - gst_element_set_state (pipeline, GST_STATE_PLAYING); - - msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); - gst_message_unref (msg); - - /* 48000 samples per buffer * 2 sources * 4 buffers */ - fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat)); - - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_element_release_request_pad (interleave, sinkpad0); - gst_object_unref (sinkpad0); - gst_element_release_request_pad (interleave, sinkpad1); - gst_object_unref (sinkpad1); - gst_object_unref (interleave); - gst_object_unref (pipeline); -} - -GST_START_TEST (test_audiointerleave_2ch_pipeline_audiointerleaved) -{ - test_audiointerleave_2ch_pipeline (TRUE); -} - -GST_END_TEST; - -GST_START_TEST (test_audiointerleave_2ch_pipeline_non_audiointerleaved) -{ - test_audiointerleave_2ch_pipeline (FALSE); -} - -GST_END_TEST; - -GST_START_TEST (test_audiointerleave_2ch_pipeline_input_chanpos) -{ - GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink; - GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2; - GstMessage *msg; - - have_data = 0; - - pipeline = (GstElement *) gst_pipeline_new ("pipeline"); - fail_unless (pipeline != NULL); - - src1 = gst_element_factory_make ("fakesrc", "src1"); - fail_unless (src1 != NULL); - g_object_set (src1, "num-buffers", 4, NULL); - g_object_set (src1, "sizetype", 2, - "sizemax", (int) 48000 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_object_set (src1, "signal-handoffs", TRUE, NULL); - g_object_set (src1, "format", GST_FORMAT_TIME, NULL); - g_signal_connect (src1, "handoff", - G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2)); - gst_bin_add (GST_BIN (pipeline), src1); - - src2 = gst_element_factory_make ("fakesrc", "src2"); - fail_unless (src2 != NULL); - g_object_set (src2, "num-buffers", 4, NULL); - g_object_set (src2, "sizetype", 2, - "sizemax", (int) 48000 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_object_set (src2, "signal-handoffs", TRUE, NULL); - g_object_set (src2, "format", GST_FORMAT_TIME, NULL); - g_signal_connect (src2, "handoff", - G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (3)); - gst_bin_add (GST_BIN (pipeline), src2); - - queue = gst_element_factory_make ("queue", "queue"); - fail_unless (queue != NULL); - gst_bin_add (GST_BIN (pipeline), queue); - - interleave = gst_element_factory_make ("audiointerleave", "audiointerleave"); - fail_unless (interleave != NULL); - g_object_set (interleave, "channel-positions-from-input", TRUE, NULL); - gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave)); - - sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sinkpad0 != NULL); - tmp = gst_element_get_static_pad (src1, "src"); - fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sinkpad1 != NULL); - tmp = gst_element_get_static_pad (src2, "src"); - tmp2 = gst_element_get_static_pad (queue, "sink"); - fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - gst_object_unref (tmp2); - tmp = gst_element_get_static_pad (queue, "src"); - fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - sink = gst_element_factory_make ("fakesink", "sink"); - fail_unless (sink != NULL); - g_object_set (sink, "signal-handoffs", TRUE, NULL); - g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32), - GINT_TO_POINTER (1)); - gst_bin_add (GST_BIN (pipeline), sink); - tmp = gst_element_get_static_pad (interleave, "src"); - tmp2 = gst_element_get_static_pad (sink, "sink"); - fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - gst_object_unref (tmp2); - - gst_element_set_state (pipeline, GST_STATE_PLAYING); - - msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); - gst_message_unref (msg); - - /* 48000 samples per buffer * 2 sources * 4 buffers */ - fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat)); - - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_element_release_request_pad (interleave, sinkpad0); - gst_object_unref (sinkpad0); - gst_element_release_request_pad (interleave, sinkpad1); - gst_object_unref (sinkpad1); - gst_object_unref (interleave); - gst_object_unref (pipeline); -} - -GST_END_TEST; - -GST_START_TEST (test_audiointerleave_2ch_pipeline_custom_chanpos) -{ - GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink; - GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2; - GstMessage *msg; - GValueArray *arr; - GValue val = { 0, }; - - have_data = 0; - - pipeline = (GstElement *) gst_pipeline_new ("pipeline"); - fail_unless (pipeline != NULL); - - src1 = gst_element_factory_make ("fakesrc", "src1"); - fail_unless (src1 != NULL); - g_object_set (src1, "num-buffers", 4, NULL); - g_object_set (src1, "signal-handoffs", TRUE, NULL); - g_object_set (src1, "sizetype", 2, - "sizemax", (int) 48000 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_object_set (src1, "format", GST_FORMAT_TIME, NULL); - g_signal_connect (src1, "handoff", - G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0)); - gst_bin_add (GST_BIN (pipeline), src1); - - src2 = gst_element_factory_make ("fakesrc", "src2"); - fail_unless (src2 != NULL); - g_object_set (src2, "num-buffers", 4, NULL); - g_object_set (src2, "signal-handoffs", TRUE, NULL); - g_object_set (src2, "sizetype", 2, - "sizemax", (int) 48000 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_object_set (src2, "format", GST_FORMAT_TIME, NULL); - g_signal_connect (src2, "handoff", - G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1)); - gst_bin_add (GST_BIN (pipeline), src2); - - queue = gst_element_factory_make ("queue", "queue"); - fail_unless (queue != NULL); - gst_bin_add (GST_BIN (pipeline), queue); - - interleave = gst_element_factory_make ("audiointerleave", "audiointerleave"); - fail_unless (interleave != NULL); - g_object_set (interleave, "channel-positions-from-input", FALSE, NULL); - arr = g_value_array_new (2); - - g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION); - g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER); - g_value_array_append (arr, &val); - g_value_reset (&val); - g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER); - g_value_array_append (arr, &val); - g_value_unset (&val); - - g_object_set (interleave, "channel-positions", arr, NULL); - g_value_array_free (arr); - gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave)); - - sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sinkpad0 != NULL); - tmp = gst_element_get_static_pad (src1, "src"); - fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sinkpad1 != NULL); - tmp = gst_element_get_static_pad (src2, "src"); - tmp2 = gst_element_get_static_pad (queue, "sink"); - fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - gst_object_unref (tmp2); - tmp = gst_element_get_static_pad (queue, "src"); - fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - sink = gst_element_factory_make ("fakesink", "sink"); - fail_unless (sink != NULL); - g_object_set (sink, "signal-handoffs", TRUE, NULL); - g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32), - GINT_TO_POINTER (2)); - gst_bin_add (GST_BIN (pipeline), sink); - tmp = gst_element_get_static_pad (interleave, "src"); - tmp2 = gst_element_get_static_pad (sink, "sink"); - fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - gst_object_unref (tmp2); - - gst_element_set_state (pipeline, GST_STATE_PLAYING); - - msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); - gst_message_unref (msg); - - /* 48000 samples per buffer * 2 sources * 4 buffers */ - fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat)); - - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_element_release_request_pad (interleave, sinkpad0); - gst_object_unref (sinkpad0); - gst_element_release_request_pad (interleave, sinkpad1); - gst_object_unref (sinkpad1); - gst_object_unref (interleave); - gst_object_unref (pipeline); -} - -GST_END_TEST; - -GST_START_TEST (test_audiointerleave_2ch_pipeline_no_chanpos) -{ - GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink; - GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2; - GstMessage *msg; - - have_data = 0; - - pipeline = (GstElement *) gst_pipeline_new ("pipeline"); - fail_unless (pipeline != NULL); - - src1 = gst_element_factory_make ("fakesrc", "src1"); - fail_unless (src1 != NULL); - g_object_set (src1, "num-buffers", 4, NULL); - g_object_set (src1, "signal-handoffs", TRUE, NULL); - g_object_set (src1, "sizetype", 2, - "sizemax", (int) 48000 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_object_set (src1, "format", GST_FORMAT_TIME, NULL); - g_signal_connect (src1, "handoff", - G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (0)); - gst_bin_add (GST_BIN (pipeline), src1); - - src2 = gst_element_factory_make ("fakesrc", "src2"); - fail_unless (src2 != NULL); - g_object_set (src2, "num-buffers", 4, NULL); - g_object_set (src2, "signal-handoffs", TRUE, NULL); - g_object_set (src2, "sizetype", 2, - "sizemax", (int) 48000 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_object_set (src2, "format", GST_FORMAT_TIME, NULL); - g_signal_connect (src2, "handoff", - G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (1)); - gst_bin_add (GST_BIN (pipeline), src2); - - queue = gst_element_factory_make ("queue", "queue"); - fail_unless (queue != NULL); - gst_bin_add (GST_BIN (pipeline), queue); - - interleave = gst_element_factory_make ("audiointerleave", "audiointerleave"); - fail_unless (interleave != NULL); - g_object_set (interleave, "channel-positions-from-input", FALSE, NULL); - gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave)); - - sinkpad0 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sinkpad0 != NULL); - tmp = gst_element_get_static_pad (src1, "src"); - fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - sinkpad1 = gst_element_get_request_pad (interleave, "sink_%u"); - fail_unless (sinkpad1 != NULL); - tmp = gst_element_get_static_pad (src2, "src"); - tmp2 = gst_element_get_static_pad (queue, "sink"); - fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - gst_object_unref (tmp2); - tmp = gst_element_get_static_pad (queue, "src"); - fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - - sink = gst_element_factory_make ("fakesink", "sink"); - fail_unless (sink != NULL); - g_object_set (sink, "signal-handoffs", TRUE, NULL); - g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32), - GINT_TO_POINTER (0)); - gst_bin_add (GST_BIN (pipeline), sink); - tmp = gst_element_get_static_pad (interleave, "src"); - tmp2 = gst_element_get_static_pad (sink, "sink"); - fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK); - gst_object_unref (tmp); - gst_object_unref (tmp2); - - gst_element_set_state (pipeline, GST_STATE_PLAYING); - - msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1); - gst_message_unref (msg); - - /* 48000 samples per buffer * 2 sources * 4 buffers */ - fail_unless (have_data == 48000 * 2 * 4 * sizeof (gfloat)); - - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_element_release_request_pad (interleave, sinkpad0); - gst_object_unref (sinkpad0); - gst_element_release_request_pad (interleave, sinkpad1); - gst_object_unref (sinkpad1); - gst_object_unref (interleave); - gst_object_unref (pipeline); -} - -GST_END_TEST; - -static void -forward_check_event (GstHarness * h, GstHarness * hsrc, GstEventType type) -{ - GstEvent *e; - - e = gst_harness_pull_event (hsrc); - fail_unless (GST_EVENT_TYPE (e) == type); - gst_harness_push_event (h, e); -} - -GST_START_TEST (test_audiointerleave_2ch_smallbuf) -{ - GstElement *audiointerleave; - GstHarness *hsrc; - GstHarness *h; - GstHarness *h2; - GstBuffer *buffer; - gint i; - GstEvent *ev; - GstCaps *ecaps, *caps; - - audiointerleave = gst_element_factory_make ("audiointerleave", NULL); - - g_object_set (audiointerleave, "latency", GST_SECOND / 2, - "output-buffer-duration", GST_SECOND / 4, NULL); - - h = gst_harness_new_with_element (audiointerleave, "sink_0", "src"); - gst_harness_use_testclock (h); - - h2 = gst_harness_new_with_element (audiointerleave, "sink_1", NULL); - gst_harness_set_src_caps_str (h2, "audio/x-raw, " - "format=" GST_AUDIO_NE (F32) ", channels=(int)1," - " layout=interleaved, rate=48000, channel-mask=(bitmask)8"); - - hsrc = gst_harness_new ("fakesrc"); - gst_harness_use_testclock (hsrc); - g_object_set (hsrc->element, - "is-live", TRUE, - "sync", TRUE, - "signal-handoffs", TRUE, - "format", GST_FORMAT_TIME, - "sizetype", 2, - "sizemax", (int) 480 * sizeof (gfloat), - "datarate", (int) 48000 * sizeof (gfloat), NULL); - g_signal_connect (hsrc->element, "handoff", - G_CALLBACK (src_handoff_float32_audiointerleaved), GINT_TO_POINTER (2)); - gst_harness_play (hsrc); - - gst_harness_crank_single_clock_wait (hsrc); - forward_check_event (h, hsrc, GST_EVENT_STREAM_START); - forward_check_event (h, hsrc, GST_EVENT_CAPS); - forward_check_event (h, hsrc, GST_EVENT_SEGMENT); - gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */ - - for (i = 0; i < 24; i++) { - gst_harness_crank_single_clock_wait (hsrc); - forward_check_event (h, hsrc, GST_EVENT_CAPS); - gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */ - } - - gst_harness_crank_single_clock_wait (h); - - - gst_event_unref (gst_harness_pull_event (h)); /* stream-start */ - ev = gst_harness_pull_event (h); /* caps */ - fail_unless_equals_int (GST_EVENT_CAPS, GST_EVENT_TYPE (ev)); - - caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, GST_AUDIO_NE (F32), - "channels", G_TYPE_INT, 2, - "layout", G_TYPE_STRING, "interleaved", - "rate", G_TYPE_INT, 48000, "channel-mask", GST_TYPE_BITMASK, - (guint64) 0x9, NULL); - - gst_event_parse_caps (ev, &ecaps); - gst_check_caps_equal (ecaps, caps); - gst_caps_unref (caps); - gst_event_unref (ev); - - /* eat the caps processing */ - gst_harness_crank_single_clock_wait (h); - for (i = 0; i < 23; i++) - gst_harness_crank_single_clock_wait (h); - fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK - (h->element)), 750 * GST_MSECOND); - - buffer = gst_harness_pull (h); - sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); - gst_buffer_unref (buffer); - fail_unless_equals_int (gst_harness_buffers_received (h), 1); - - for (i = 0; i < 50; i++) { - gst_harness_crank_single_clock_wait (hsrc); - forward_check_event (h, hsrc, GST_EVENT_CAPS); - gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */ - } - for (i = 0; i < 25; i++) - gst_harness_crank_single_clock_wait (h); - fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK - (h->element)), 1000 * GST_MSECOND); - buffer = gst_harness_pull (h); - sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); - gst_buffer_unref (buffer); - fail_unless_equals_int (gst_harness_buffers_received (h), 2); - - for (i = 0; i < 25; i++) { - gst_harness_crank_single_clock_wait (hsrc); - forward_check_event (h, hsrc, GST_EVENT_CAPS); - gst_harness_push (h, gst_harness_pull (hsrc)); /* buffer */ - } - for (i = 0; i < 25; i++) - gst_harness_crank_single_clock_wait (h); - fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK - (h->element)), 1250 * GST_MSECOND); - buffer = gst_harness_pull (h); - sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); - gst_buffer_unref (buffer); - fail_unless_equals_int (gst_harness_buffers_received (h), 3); - - gst_harness_push_event (h, gst_event_new_eos ()); - - for (i = 0; i < 25; i++) - gst_harness_crank_single_clock_wait (h); - fail_unless_equals_uint64 (gst_clock_get_time (GST_ELEMENT_CLOCK - (h->element)), 1500 * GST_MSECOND); - buffer = gst_harness_pull (h); - sink_handoff_float32 (NULL, buffer, NULL, GUINT_TO_POINTER (3)); - gst_buffer_unref (buffer); - - fail_unless_equals_int (gst_harness_buffers_received (h), 4); - - gst_harness_teardown (h2); - gst_harness_teardown (h); - gst_harness_teardown (hsrc); - gst_object_unref (audiointerleave); -} - -GST_END_TEST; - -static Suite * -audiointerleave_suite (void) -{ - Suite *s = suite_create ("audiointerleave"); - TCase *tc_chain = tcase_create ("general"); - - suite_add_tcase (s, tc_chain); - tcase_set_timeout (tc_chain, 180); - tcase_add_test (tc_chain, test_create_and_unref); - tcase_add_test (tc_chain, test_request_pads); - tcase_add_test (tc_chain, test_audiointerleave_2ch); - tcase_add_test (tc_chain, test_audiointerleave_2ch_1eos); - tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_audiointerleaved); - tcase_add_test (tc_chain, - test_audiointerleave_2ch_pipeline_non_audiointerleaved); - tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_input_chanpos); - tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_custom_chanpos); - tcase_add_test (tc_chain, test_audiointerleave_2ch_pipeline_no_chanpos); - tcase_add_test (tc_chain, test_audiointerleave_2ch_smallbuf); - - return s; -} - -GST_CHECK_MAIN (audiointerleave); diff --git a/tests/check/elements/audiomixer.c b/tests/check/elements/audiomixer.c deleted file mode 100644 index 4a8a8233be..0000000000 --- a/tests/check/elements/audiomixer.c +++ /dev/null @@ -1,1894 +0,0 @@ -/* GStreamer - * - * unit test for audiomixer - * - * Copyright (C) 2005 Thomas Vander Stichele - * Copyright (C) 2013 Sebastian Dröge - * - * 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. - */ - -#ifdef HAVE_CONFIG_H -# include -#endif - -#ifdef HAVE_VALGRIND -# include -#endif - -#include - -#include -#include -#include -#include -#include -#include - -static GMainLoop *main_loop; - -/* fixtures */ - -static void -test_setup (void) -{ - main_loop = g_main_loop_new (NULL, FALSE); -} - -static void -test_teardown (void) -{ - g_main_loop_unref (main_loop); - main_loop = NULL; -} - - -/* some test helpers */ - -static GstElement * -setup_pipeline (GstElement * audiomixer, gint num_srcs, GstElement * capsfilter) -{ - GstElement *pipeline, *src, *sink; - gint i; - - pipeline = gst_pipeline_new ("pipeline"); - if (!audiomixer) { - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - } - - sink = gst_element_factory_make ("fakesink", "sink"); - gst_bin_add_many (GST_BIN (pipeline), audiomixer, sink, NULL); - - if (capsfilter) { - gst_bin_add (GST_BIN (pipeline), capsfilter); - gst_element_link_many (audiomixer, capsfilter, sink, NULL); - } else { - gst_element_link (audiomixer, sink); - } - - for (i = 0; i < num_srcs; i++) { - src = gst_element_factory_make ("audiotestsrc", NULL); - g_object_set (src, "wave", 4, NULL); /* silence */ - gst_bin_add (GST_BIN (pipeline), src); - gst_element_link (src, audiomixer); - } - return pipeline; -} - -static GstCaps * -get_element_sink_pad_caps (GstElement * pipeline, const gchar * element_name) -{ - GstElement *sink; - GstCaps *caps; - GstPad *pad; - - sink = gst_bin_get_by_name (GST_BIN (pipeline), "sink"); - pad = gst_element_get_static_pad (sink, "sink"); - caps = gst_pad_get_current_caps (pad); - gst_object_unref (pad); - gst_object_unref (sink); - - return caps; -} - -static void -set_state_and_wait (GstElement * pipeline, GstState state) -{ - GstStateChangeReturn state_res; - - /* prepare paused/playing */ - state_res = gst_element_set_state (pipeline, state); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - /* wait for preroll */ - state_res = gst_element_get_state (pipeline, NULL, NULL, GST_CLOCK_TIME_NONE); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); -} - -static gboolean -set_playing (GstElement * element) -{ - GstStateChangeReturn state_res; - - state_res = gst_element_set_state (element, GST_STATE_PLAYING); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - return FALSE; -} - -static void -play_and_wait (GstElement * pipeline) -{ - GstStateChangeReturn state_res; - - g_idle_add ((GSourceFunc) set_playing, pipeline); - - GST_INFO ("running main loop"); - g_main_loop_run (main_loop); - - state_res = gst_element_set_state (pipeline, GST_STATE_NULL); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); -} - -static void -message_received (GstBus * bus, GstMessage * message, GstPipeline * bin) -{ - GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, - GST_MESSAGE_SRC (message), message); - - switch (message->type) { - case GST_MESSAGE_EOS: - g_main_loop_quit (main_loop); - break; - case GST_MESSAGE_WARNING:{ - GError *gerror; - gchar *debug; - - gst_message_parse_warning (message, &gerror, &debug); - gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug); - g_error_free (gerror); - g_free (debug); - break; - } - case GST_MESSAGE_ERROR:{ - GError *gerror; - gchar *debug; - - gst_message_parse_error (message, &gerror, &debug); - gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug); - g_error_free (gerror); - g_free (debug); - g_main_loop_quit (main_loop); - break; - } - default: - break; - } -} - -static GstBuffer * -new_buffer (gsize num_bytes, gint data, GstClockTime ts, GstClockTime dur, - GstBufferFlags flags) -{ - GstMapInfo map; - GstBuffer *buffer = gst_buffer_new_and_alloc (num_bytes); - - gst_buffer_map (buffer, &map, GST_MAP_WRITE); - memset (map.data, data, map.size); - gst_buffer_unmap (buffer, &map); - GST_BUFFER_TIMESTAMP (buffer) = ts; - GST_BUFFER_DURATION (buffer) = dur; - if (flags) - GST_BUFFER_FLAG_SET (buffer, flags); - GST_DEBUG ("created buffer %p", buffer); - return buffer; -} - -/* make sure downstream gets a CAPS event before buffers are sent */ -GST_START_TEST (test_caps) -{ - GstElement *pipeline; - GstCaps *caps; - - /* build pipeline */ - pipeline = setup_pipeline (NULL, 1, NULL); - - /* prepare playing */ - set_state_and_wait (pipeline, GST_STATE_PAUSED); - - /* check caps on fakesink */ - caps = get_element_sink_pad_caps (pipeline, "sink"); - fail_unless (caps != NULL); - gst_caps_unref (caps); - - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_object_unref (pipeline); -} - -GST_END_TEST; - -/* check that caps set on the property are honoured */ -GST_START_TEST (test_filter_caps) -{ - GstElement *pipeline, *audiomixer, *capsfilter; - GstCaps *filter_caps, *caps; - - filter_caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, GST_AUDIO_NE (F32), - "layout", G_TYPE_STRING, "interleaved", - "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1, - "channel-mask", GST_TYPE_BITMASK, (guint64) 0x04, NULL); - - capsfilter = gst_element_factory_make ("capsfilter", NULL); - - /* build pipeline */ - audiomixer = gst_element_factory_make ("audiomixer", NULL); - g_object_set (capsfilter, "caps", filter_caps, NULL); - pipeline = setup_pipeline (audiomixer, 1, capsfilter); - - /* prepare playing */ - set_state_and_wait (pipeline, GST_STATE_PAUSED); - - /* check caps on fakesink */ - caps = get_element_sink_pad_caps (pipeline, "sink"); - fail_unless (caps != NULL); - GST_INFO_OBJECT (pipeline, "received caps: %" GST_PTR_FORMAT, caps); - fail_unless (gst_caps_is_equal_fixed (caps, filter_caps)); - gst_caps_unref (caps); - - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_object_unref (pipeline); - - gst_caps_unref (filter_caps); -} - -GST_END_TEST; - -static GstFormat format = GST_FORMAT_UNDEFINED; -static gint64 position = -1; - -static void -test_event_message_received (GstBus * bus, GstMessage * message, - GstPipeline * bin) -{ - GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, - GST_MESSAGE_SRC (message), message); - - switch (message->type) { - case GST_MESSAGE_SEGMENT_DONE: - gst_message_parse_segment_done (message, &format, &position); - GST_INFO ("received segment_done : %" G_GINT64_FORMAT, position); - g_main_loop_quit (main_loop); - break; - default: - g_assert_not_reached (); - break; - } -} - - -GST_START_TEST (test_event) -{ - GstElement *bin, *src1, *src2, *audiomixer, *sink; - GstBus *bus; - GstEvent *seek_event; - gboolean res; - GstPad *srcpad, *sinkpad; - GstStreamConsistency *chk_1, *chk_2, *chk_3; - - GST_INFO ("preparing test"); - - /* build pipeline */ - bin = gst_pipeline_new ("pipeline"); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - src1 = gst_element_factory_make ("audiotestsrc", "src1"); - g_object_set (src1, "wave", 4, NULL); /* silence */ - src2 = gst_element_factory_make ("audiotestsrc", "src2"); - g_object_set (src2, "wave", 4, NULL); /* silence */ - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - sink = gst_element_factory_make ("fakesink", "sink"); - gst_bin_add_many (GST_BIN (bin), src1, src2, audiomixer, sink, NULL); - - res = gst_element_link (src1, audiomixer); - fail_unless (res == TRUE, NULL); - res = gst_element_link (src2, audiomixer); - fail_unless (res == TRUE, NULL); - res = gst_element_link (audiomixer, sink); - fail_unless (res == TRUE, NULL); - - srcpad = gst_element_get_static_pad (audiomixer, "src"); - chk_3 = gst_consistency_checker_new (srcpad); - gst_object_unref (srcpad); - - /* create consistency checkers for the pads */ - srcpad = gst_element_get_static_pad (src1, "src"); - chk_1 = gst_consistency_checker_new (srcpad); - sinkpad = gst_pad_get_peer (srcpad); - gst_consistency_checker_add_pad (chk_3, sinkpad); - gst_object_unref (sinkpad); - gst_object_unref (srcpad); - - srcpad = gst_element_get_static_pad (src2, "src"); - chk_2 = gst_consistency_checker_new (srcpad); - sinkpad = gst_pad_get_peer (srcpad); - gst_consistency_checker_add_pad (chk_3, sinkpad); - gst_object_unref (sinkpad); - gst_object_unref (srcpad); - - seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, - GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH, - GST_SEEK_TYPE_SET, (GstClockTime) 0, - GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND); - - format = GST_FORMAT_UNDEFINED; - position = -1; - - g_signal_connect (bus, "message::segment-done", - (GCallback) test_event_message_received, bin); - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - GST_INFO ("starting test"); - - /* prepare playing */ - set_state_and_wait (bin, GST_STATE_PAUSED); - - res = gst_element_send_event (bin, seek_event); - fail_unless (res == TRUE, NULL); - - /* run pipeline */ - play_and_wait (bin); - - ck_assert_int_eq (position, 2 * GST_SECOND); - - /* cleanup */ - gst_consistency_checker_free (chk_1); - gst_consistency_checker_free (chk_2); - gst_consistency_checker_free (chk_3); - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); -} - -GST_END_TEST; - -static guint play_count = 0; -static GstEvent *play_seek_event = NULL; - -static void -test_play_twice_message_received (GstBus * bus, GstMessage * message, - GstElement * bin) -{ - gboolean res; - GstStateChangeReturn state_res; - - GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, - GST_MESSAGE_SRC (message), message); - - switch (message->type) { - case GST_MESSAGE_SEGMENT_DONE: - play_count++; - if (play_count == 1) { - state_res = gst_element_set_state (bin, GST_STATE_READY); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - /* prepare playing again */ - set_state_and_wait (bin, GST_STATE_PAUSED); - - res = gst_element_send_event (bin, gst_event_ref (play_seek_event)); - fail_unless (res == TRUE, NULL); - - state_res = gst_element_set_state (bin, GST_STATE_PLAYING); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - } else { - g_main_loop_quit (main_loop); - } - break; - default: - g_assert_not_reached (); - break; - } -} - - -GST_START_TEST (test_play_twice) -{ - GstElement *bin, *audiomixer; - GstBus *bus; - gboolean res; - GstPad *srcpad; - GstStreamConsistency *consist; - - GST_INFO ("preparing test"); - - /* build pipeline */ - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - bin = setup_pipeline (audiomixer, 2, NULL); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - srcpad = gst_element_get_static_pad (audiomixer, "src"); - consist = gst_consistency_checker_new (srcpad); - gst_object_unref (srcpad); - - play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, - GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH, - GST_SEEK_TYPE_SET, (GstClockTime) 0, - GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND); - - play_count = 0; - - g_signal_connect (bus, "message::segment-done", - (GCallback) test_play_twice_message_received, bin); - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - GST_INFO ("starting test"); - - /* prepare playing */ - set_state_and_wait (bin, GST_STATE_PAUSED); - - res = gst_element_send_event (bin, gst_event_ref (play_seek_event)); - fail_unless (res == TRUE, NULL); - - GST_INFO ("seeked"); - - /* run pipeline */ - play_and_wait (bin); - - ck_assert_int_eq (play_count, 2); - - /* cleanup */ - gst_consistency_checker_free (consist); - gst_event_unref (play_seek_event); - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); -} - -GST_END_TEST; - -GST_START_TEST (test_play_twice_then_add_and_play_again) -{ - GstElement *bin, *src, *audiomixer; - GstBus *bus; - gboolean res; - GstStateChangeReturn state_res; - gint i; - GstPad *srcpad; - GstStreamConsistency *consist; - - GST_INFO ("preparing test"); - - /* build pipeline */ - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - bin = setup_pipeline (audiomixer, 2, NULL); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - srcpad = gst_element_get_static_pad (audiomixer, "src"); - consist = gst_consistency_checker_new (srcpad); - gst_object_unref (srcpad); - - play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, - GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH, - GST_SEEK_TYPE_SET, (GstClockTime) 0, - GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND); - - g_signal_connect (bus, "message::segment-done", - (GCallback) test_play_twice_message_received, bin); - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - /* run it twice */ - for (i = 0; i < 2; i++) { - play_count = 0; - - GST_INFO ("starting test-loop %d", i); - - /* prepare playing */ - set_state_and_wait (bin, GST_STATE_PAUSED); - - res = gst_element_send_event (bin, gst_event_ref (play_seek_event)); - fail_unless (res == TRUE, NULL); - - GST_INFO ("seeked"); - - /* run pipeline */ - play_and_wait (bin); - - ck_assert_int_eq (play_count, 2); - - /* plug another source */ - if (i == 0) { - src = gst_element_factory_make ("audiotestsrc", NULL); - g_object_set (src, "wave", 4, NULL); /* silence */ - gst_bin_add (GST_BIN (bin), src); - - res = gst_element_link (src, audiomixer); - fail_unless (res == TRUE, NULL); - } - - gst_consistency_checker_reset (consist); - } - - state_res = gst_element_set_state (bin, GST_STATE_NULL); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - /* cleanup */ - gst_event_unref (play_seek_event); - gst_consistency_checker_free (consist); - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); -} - -GST_END_TEST; - - -static GstElement * -test_live_seeking_try_audiosrc (const gchar * factory_name) -{ - GstElement *src; - GstStateChangeReturn state_res; - - if (!(src = gst_element_factory_make (factory_name, NULL))) { - GST_INFO ("can't make '%s', skipping", factory_name); - return NULL; - } - - /* Test that the audio source can get to ready, else skip */ - state_res = gst_element_set_state (src, GST_STATE_READY); - gst_element_set_state (src, GST_STATE_NULL); - - if (state_res == GST_STATE_CHANGE_FAILURE) { - GST_INFO_OBJECT (src, "can't go to ready, skipping"); - gst_object_unref (src); - return NULL; - } - - return src; -} - -/* test failing seeks on live-sources */ -GST_START_TEST (test_live_seeking) -{ - GstElement *bin, *src1 = NULL, *cf, *src2, *audiomixer, *sink; - GstCaps *caps; - GstBus *bus; - gboolean res; - GstPad *srcpad; - GstPad *sinkpad; - gint i; - GstStreamConsistency *consist; - /* don't use autoaudiosrc, as then we can't set anything here */ - const gchar *audio_src_factories[] = { - "alsasrc", - "pulseaudiosrc" - }; - - GST_INFO ("preparing test"); - play_seek_event = NULL; - - /* build pipeline */ - bin = gst_pipeline_new ("pipeline"); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - for (i = 0; (i < G_N_ELEMENTS (audio_src_factories) && src1 == NULL); i++) { - src1 = test_live_seeking_try_audiosrc (audio_src_factories[i]); - } - if (!src1) { - /* normal audiosources behave differently than audiotestsrc */ - GST_WARNING ("no real audiosrc found, using audiotestsrc is-live"); - src1 = gst_element_factory_make ("audiotestsrc", "src1"); - g_object_set (src1, "wave", 4, "is-live", TRUE, NULL); /* silence */ - } else { - /* live sources ignore seeks, force eos after 2 sec (4 buffers half second - * each) - */ - g_object_set (src1, "num-buffers", 4, "blocksize", 44100, NULL); - } - - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - cf = gst_element_factory_make ("capsfilter", "capsfilter"); - sink = gst_element_factory_make ("fakesink", "sink"); - - gst_bin_add_many (GST_BIN (bin), src1, cf, audiomixer, sink, NULL); - res = gst_element_link_many (src1, cf, audiomixer, sink, NULL); - fail_unless (res == TRUE, NULL); - - /* get the caps for the livesrc, we'll reuse this for the non-live source */ - set_state_and_wait (bin, GST_STATE_PLAYING); - - sinkpad = gst_element_get_static_pad (sink, "sink"); - fail_unless (sinkpad != NULL); - caps = gst_pad_get_current_caps (sinkpad); - fail_unless (caps != NULL); - gst_object_unref (sinkpad); - - gst_element_set_state (bin, GST_STATE_NULL); - - g_object_set (cf, "caps", caps, NULL); - - src2 = gst_element_factory_make ("audiotestsrc", "src2"); - g_object_set (src2, "wave", 4, NULL); /* silence */ - gst_bin_add (GST_BIN (bin), src2); - - res = gst_element_link_filtered (src2, audiomixer, caps); - fail_unless (res == TRUE, NULL); - - gst_caps_unref (caps); - - play_seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, - GST_SEEK_FLAG_FLUSH, - GST_SEEK_TYPE_SET, (GstClockTime) 0, - GST_SEEK_TYPE_SET, (GstClockTime) 2 * GST_SECOND); - - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - srcpad = gst_element_get_static_pad (audiomixer, "src"); - consist = gst_consistency_checker_new (srcpad); - gst_object_unref (srcpad); - - GST_INFO ("starting test"); - - /* run it twice */ - for (i = 0; i < 2; i++) { - - GST_INFO ("starting test-loop %d", i); - - /* prepare playing */ - set_state_and_wait (bin, GST_STATE_PAUSED); - - res = gst_element_send_event (bin, gst_event_ref (play_seek_event)); - fail_unless (res == TRUE, NULL); - - GST_INFO ("seeked"); - - /* run pipeline */ - play_and_wait (bin); - - gst_consistency_checker_reset (consist); - } - - /* cleanup */ - GST_INFO ("cleaning up"); - gst_consistency_checker_free (consist); - if (play_seek_event) - gst_event_unref (play_seek_event); - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); -} - -GST_END_TEST; - -/* check if adding pads work as expected */ -GST_START_TEST (test_add_pad) -{ - GstElement *bin, *src1, *src2, *audiomixer, *sink; - GstBus *bus; - GstPad *srcpad; - gboolean res; - GstStateChangeReturn state_res; - - GST_INFO ("preparing test"); - - /* build pipeline */ - bin = gst_pipeline_new ("pipeline"); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - src1 = gst_element_factory_make ("audiotestsrc", "src1"); - g_object_set (src1, "num-buffers", 4, "wave", /* silence */ 4, NULL); - src2 = gst_element_factory_make ("audiotestsrc", "src2"); - /* one buffer less, we connect with 1 buffer of delay */ - g_object_set (src2, "num-buffers", 3, "wave", /* silence */ 4, NULL); - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - sink = gst_element_factory_make ("fakesink", "sink"); - gst_bin_add_many (GST_BIN (bin), src1, audiomixer, sink, NULL); - - res = gst_element_link (src1, audiomixer); - fail_unless (res == TRUE, NULL); - res = gst_element_link (audiomixer, sink); - fail_unless (res == TRUE, NULL); - - srcpad = gst_element_get_static_pad (audiomixer, "src"); - gst_object_unref (srcpad); - - g_signal_connect (bus, "message::segment-done", (GCallback) message_received, - bin); - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - GST_INFO ("starting test"); - - /* prepare playing */ - set_state_and_wait (bin, GST_STATE_PAUSED); - - /* add other element */ - gst_bin_add_many (GST_BIN (bin), src2, NULL); - - /* now link the second element */ - res = gst_element_link (src2, audiomixer); - fail_unless (res == TRUE, NULL); - - /* set to PAUSED as well */ - state_res = gst_element_set_state (src2, GST_STATE_PAUSED); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - /* now play all */ - play_and_wait (bin); - - /* cleanup */ - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); -} - -GST_END_TEST; - -/* check if removing pads work as expected */ -GST_START_TEST (test_remove_pad) -{ - GstElement *bin, *src, *audiomixer, *sink; - GstBus *bus; - GstPad *pad, *srcpad; - gboolean res; - GstStateChangeReturn state_res; - - GST_INFO ("preparing test"); - - /* build pipeline */ - bin = gst_pipeline_new ("pipeline"); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - src = gst_element_factory_make ("audiotestsrc", "src"); - g_object_set (src, "num-buffers", 4, "wave", 4, NULL); - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - sink = gst_element_factory_make ("fakesink", "sink"); - gst_bin_add_many (GST_BIN (bin), src, audiomixer, sink, NULL); - - res = gst_element_link (src, audiomixer); - fail_unless (res == TRUE, NULL); - res = gst_element_link (audiomixer, sink); - fail_unless (res == TRUE, NULL); - - /* create an unconnected sinkpad in audiomixer */ - pad = gst_element_get_request_pad (audiomixer, "sink_%u"); - fail_if (pad == NULL, NULL); - - srcpad = gst_element_get_static_pad (audiomixer, "src"); - gst_object_unref (srcpad); - - g_signal_connect (bus, "message::segment-done", (GCallback) message_received, - bin); - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - GST_INFO ("starting test"); - - /* prepare playing, this will not preroll as audiomixer is waiting - * on the unconnected sinkpad. */ - state_res = gst_element_set_state (bin, GST_STATE_PAUSED); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - /* wait for completion for one second, will return ASYNC */ - state_res = gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, GST_SECOND); - ck_assert_int_eq (state_res, GST_STATE_CHANGE_ASYNC); - - /* get rid of the pad now, audiomixer should stop waiting on it and - * continue the preroll */ - gst_element_release_request_pad (audiomixer, pad); - gst_object_unref (pad); - - /* wait for completion, should work now */ - state_res = - gst_element_get_state (GST_ELEMENT (bin), NULL, NULL, - GST_CLOCK_TIME_NONE); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - /* now play all */ - play_and_wait (bin); - - /* cleanup */ - gst_bus_remove_signal_watch (bus); - gst_object_unref (G_OBJECT (bus)); - gst_object_unref (G_OBJECT (bin)); -} - -GST_END_TEST; - - -static GstBuffer *handoff_buffer = NULL; - -static void -handoff_buffer_cb (GstElement * fakesink, GstBuffer * buffer, GstPad * pad, - gpointer user_data) -{ - GST_DEBUG ("got buffer -- SIZE: %" G_GSIZE_FORMAT - " -- %p PTS is %" GST_TIME_FORMAT " END is %" GST_TIME_FORMAT, - gst_buffer_get_size (buffer), buffer, - GST_TIME_ARGS (GST_BUFFER_PTS (buffer)), - GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer))); - - gst_buffer_replace (&handoff_buffer, buffer); -} - -/* check if clipping works as expected */ -GST_START_TEST (test_clip) -{ - GstSegment segment; - GstElement *bin, *audiomixer, *sink; - GstBus *bus; - GstPad *sinkpad; - gboolean res; - GstStateChangeReturn state_res; - GstFlowReturn ret; - GstEvent *event; - GstBuffer *buffer; - GstCaps *caps; - GstQuery *drain = gst_query_new_drain (); - - GST_INFO ("preparing test"); - - /* build pipeline */ - bin = gst_pipeline_new ("pipeline"); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - /* just an audiomixer and a fakesink */ - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - g_object_set (audiomixer, "output-buffer-duration", 50 * GST_MSECOND, NULL); - sink = gst_element_factory_make ("fakesink", "sink"); - g_object_set (sink, "signal-handoffs", TRUE, NULL); - g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL); - gst_bin_add_many (GST_BIN (bin), audiomixer, sink, NULL); - - res = gst_element_link (audiomixer, sink); - fail_unless (res == TRUE, NULL); - - /* set to playing */ - state_res = gst_element_set_state (bin, GST_STATE_PLAYING); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - /* create an unconnected sinkpad in audiomixer, should also automatically activate - * the pad */ - sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u"); - fail_if (sinkpad == NULL, NULL); - - gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test")); - - caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, GST_AUDIO_NE (S16), - "layout", G_TYPE_STRING, "interleaved", - "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 2, NULL); - - gst_pad_set_caps (sinkpad, caps); - gst_caps_unref (caps); - - /* send segment to audiomixer */ - gst_segment_init (&segment, GST_FORMAT_TIME); - segment.start = GST_SECOND; - segment.stop = 2 * GST_SECOND; - segment.time = 0; - event = gst_event_new_segment (&segment); - gst_pad_send_event (sinkpad, event); - - /* should be clipped and ok */ - buffer = new_buffer (44100, 0, 0, 250 * GST_MSECOND, 0); - GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT, - buffer, - GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer))); - ret = gst_pad_chain (sinkpad, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - /* The aggregation is done in a dedicated thread, so we can't - * know when it is actually going to happen, so we use a DRAIN query - * to wait for it to complete. - */ - gst_pad_query (sinkpad, drain); - fail_unless (handoff_buffer == NULL); - - /* should be partially clipped */ - buffer = new_buffer (44100, 0, 900 * GST_MSECOND, 250 * GST_MSECOND, - GST_BUFFER_FLAG_DISCONT); - GST_DEBUG ("pushing buffer %p START %" GST_TIME_FORMAT " -- DURATION is %" - GST_TIME_FORMAT, buffer, GST_TIME_ARGS (GST_BUFFER_PTS (buffer)), - GST_TIME_ARGS (GST_BUFFER_DURATION (buffer))); - ret = gst_pad_chain (sinkpad, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - gst_pad_query (sinkpad, drain); - - fail_unless (handoff_buffer != NULL); - ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) + - GST_BUFFER_DURATION (handoff_buffer), 150 * GST_MSECOND); - gst_buffer_replace (&handoff_buffer, NULL); - - /* should not be clipped */ - buffer = new_buffer (44100, 0, 1150 * GST_MSECOND, 250 * GST_MSECOND, 0); - GST_DEBUG ("pushing buffer %p END is %" GST_TIME_FORMAT, - buffer, - GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer))); - ret = gst_pad_chain (sinkpad, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - gst_pad_query (sinkpad, drain); - fail_unless (handoff_buffer != NULL); - ck_assert_int_eq (GST_BUFFER_PTS (handoff_buffer) + - GST_BUFFER_DURATION (handoff_buffer), 400 * GST_MSECOND); - gst_buffer_replace (&handoff_buffer, NULL); - fail_unless (handoff_buffer == NULL); - - /* should be clipped and ok */ - buffer = new_buffer (44100, 0, 2 * GST_SECOND, 250 * GST_MSECOND, - GST_BUFFER_FLAG_DISCONT); - GST_DEBUG ("pushing buffer %p PTS is %" GST_TIME_FORMAT - " END is %" GST_TIME_FORMAT, - buffer, - GST_TIME_ARGS (GST_BUFFER_PTS (buffer)), - GST_TIME_ARGS (GST_BUFFER_PTS (buffer) + GST_BUFFER_DURATION (buffer))); - ret = gst_pad_chain (sinkpad, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - gst_pad_query (sinkpad, drain); - fail_unless (handoff_buffer == NULL); - - gst_element_release_request_pad (audiomixer, sinkpad); - gst_object_unref (sinkpad); - gst_element_set_state (bin, GST_STATE_NULL); - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); - gst_query_unref (drain); -} - -GST_END_TEST; - -GST_START_TEST (test_duration_is_max) -{ - GstElement *bin, *src[3], *audiomixer, *sink; - GstStateChangeReturn state_res; - GstFormat format = GST_FORMAT_TIME; - gboolean res; - gint64 duration; - - GST_INFO ("preparing test"); - - /* build pipeline */ - bin = gst_pipeline_new ("pipeline"); - - /* 3 sources, an audiomixer and a fakesink */ - src[0] = gst_element_factory_make ("audiotestsrc", NULL); - src[1] = gst_element_factory_make ("audiotestsrc", NULL); - src[2] = gst_element_factory_make ("audiotestsrc", NULL); - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - sink = gst_element_factory_make ("fakesink", "sink"); - gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink, - NULL); - - gst_element_link (src[0], audiomixer); - gst_element_link (src[1], audiomixer); - gst_element_link (src[2], audiomixer); - gst_element_link (audiomixer, sink); - - /* irks, duration is reset on basesrc */ - state_res = gst_element_set_state (bin, GST_STATE_PAUSED); - fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL); - - /* set durations on src */ - GST_BASE_SRC (src[0])->segment.duration = 1000; - GST_BASE_SRC (src[1])->segment.duration = 3000; - GST_BASE_SRC (src[2])->segment.duration = 2000; - - /* set to playing */ - set_state_and_wait (bin, GST_STATE_PLAYING); - - res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration); - fail_unless (res, NULL); - - ck_assert_int_eq (duration, 3000); - - gst_element_set_state (bin, GST_STATE_NULL); - gst_object_unref (bin); -} - -GST_END_TEST; - -GST_START_TEST (test_duration_unknown_overrides) -{ - GstElement *bin, *src[3], *audiomixer, *sink; - GstStateChangeReturn state_res; - GstFormat format = GST_FORMAT_TIME; - gboolean res; - gint64 duration; - - GST_INFO ("preparing test"); - - /* build pipeline */ - bin = gst_pipeline_new ("pipeline"); - - /* 3 sources, an audiomixer and a fakesink */ - src[0] = gst_element_factory_make ("audiotestsrc", NULL); - src[1] = gst_element_factory_make ("audiotestsrc", NULL); - src[2] = gst_element_factory_make ("audiotestsrc", NULL); - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - sink = gst_element_factory_make ("fakesink", "sink"); - gst_bin_add_many (GST_BIN (bin), src[0], src[1], src[2], audiomixer, sink, - NULL); - - gst_element_link (src[0], audiomixer); - gst_element_link (src[1], audiomixer); - gst_element_link (src[2], audiomixer); - gst_element_link (audiomixer, sink); - - /* irks, duration is reset on basesrc */ - state_res = gst_element_set_state (bin, GST_STATE_PAUSED); - fail_unless (state_res != GST_STATE_CHANGE_FAILURE, NULL); - - /* set durations on src */ - GST_BASE_SRC (src[0])->segment.duration = GST_CLOCK_TIME_NONE; - GST_BASE_SRC (src[1])->segment.duration = 3000; - GST_BASE_SRC (src[2])->segment.duration = 2000; - - /* set to playing */ - set_state_and_wait (bin, GST_STATE_PLAYING); - - res = gst_element_query_duration (GST_ELEMENT (bin), format, &duration); - fail_unless (res, NULL); - - ck_assert_int_eq (duration, GST_CLOCK_TIME_NONE); - - gst_element_set_state (bin, GST_STATE_NULL); - gst_object_unref (bin); -} - -GST_END_TEST; - - -static gboolean looped = FALSE; - -static void -loop_segment_done (GstBus * bus, GstMessage * message, GstElement * bin) -{ - GST_INFO ("bus message from \"%" GST_PTR_FORMAT "\": %" GST_PTR_FORMAT, - GST_MESSAGE_SRC (message), message); - - if (looped) { - g_main_loop_quit (main_loop); - } else { - GstEvent *seek_event; - gboolean res; - - seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, - GST_SEEK_FLAG_SEGMENT, - GST_SEEK_TYPE_SET, (GstClockTime) 0, - GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND); - - res = gst_element_send_event (bin, seek_event); - fail_unless (res == TRUE, NULL); - looped = TRUE; - } -} - -GST_START_TEST (test_loop) -{ - GstElement *bin; - GstBus *bus; - GstEvent *seek_event; - gboolean res; - - GST_INFO ("preparing test"); - - /* build pipeline */ - bin = setup_pipeline (NULL, 2, NULL); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - seek_event = gst_event_new_seek (1.0, GST_FORMAT_TIME, - GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH, - GST_SEEK_TYPE_SET, (GstClockTime) 0, - GST_SEEK_TYPE_SET, (GstClockTime) 1 * GST_SECOND); - - g_signal_connect (bus, "message::segment-done", - (GCallback) loop_segment_done, bin); - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - GST_INFO ("starting test"); - - /* prepare playing */ - set_state_and_wait (bin, GST_STATE_PAUSED); - - res = gst_element_send_event (bin, seek_event); - fail_unless (res == TRUE, NULL); - - /* run pipeline */ - play_and_wait (bin); - - fail_unless (looped); - - /* cleanup */ - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); -} - -GST_END_TEST; - -GST_START_TEST (test_flush_start_flush_stop) -{ - GstPadTemplate *sink_template; - GstPad *tmppad, *srcpad1, *sinkpad1, *sinkpad2, *audiomixer_src; - GstElement *pipeline, *src1, *src2, *audiomixer, *sink; - - GST_INFO ("preparing test"); - - /* build pipeline */ - pipeline = gst_pipeline_new ("pipeline"); - src1 = gst_element_factory_make ("audiotestsrc", "src1"); - g_object_set (src1, "wave", 4, NULL); /* silence */ - src2 = gst_element_factory_make ("audiotestsrc", "src2"); - g_object_set (src2, "wave", 4, NULL); /* silence */ - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - sink = gst_element_factory_make ("fakesink", "sink"); - gst_bin_add_many (GST_BIN (pipeline), src1, src2, audiomixer, sink, NULL); - - sink_template = - gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (audiomixer), - "sink_%u"); - fail_unless (GST_IS_PAD_TEMPLATE (sink_template)); - sinkpad1 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL); - srcpad1 = gst_element_get_static_pad (src1, "src"); - gst_pad_link (srcpad1, sinkpad1); - - sinkpad2 = gst_element_request_pad (audiomixer, sink_template, NULL, NULL); - tmppad = gst_element_get_static_pad (src2, "src"); - gst_pad_link (tmppad, sinkpad2); - gst_object_unref (tmppad); - - gst_element_link (audiomixer, sink); - - /* prepare playing */ - set_state_and_wait (pipeline, GST_STATE_PLAYING); - - audiomixer_src = gst_element_get_static_pad (audiomixer, "src"); - fail_if (GST_PAD_IS_FLUSHING (audiomixer_src)); - gst_pad_send_event (sinkpad1, gst_event_new_flush_start ()); - fail_if (GST_PAD_IS_FLUSHING (audiomixer_src)); - fail_unless (GST_PAD_IS_FLUSHING (sinkpad1)); - /* Hold the streamlock to make sure the flush stop is not between - the attempted push of a segment event and of the following buffer. */ - GST_PAD_STREAM_LOCK (srcpad1); - gst_pad_send_event (sinkpad1, gst_event_new_flush_stop (TRUE)); - GST_PAD_STREAM_UNLOCK (srcpad1); - fail_if (GST_PAD_IS_FLUSHING (audiomixer_src)); - fail_if (GST_PAD_IS_FLUSHING (sinkpad1)); - gst_object_unref (audiomixer_src); - - gst_element_release_request_pad (audiomixer, sinkpad1); - gst_object_unref (sinkpad1); - gst_element_release_request_pad (audiomixer, sinkpad2); - gst_object_unref (sinkpad2); - gst_object_unref (srcpad1); - - /* cleanup */ - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_object_unref (pipeline); -} - -GST_END_TEST; - -static void -handoff_buffer_collect_cb (GstElement * fakesink, GstBuffer * buffer, - GstPad * pad, gpointer user_data) -{ - GList **received_buffers = user_data; - - GST_DEBUG ("got buffer %p", buffer); - *received_buffers = - g_list_append (*received_buffers, gst_buffer_ref (buffer)); -} - -typedef void (*SendBuffersFunction) (GstPad * pad1, GstPad * pad2); -typedef void (*CheckBuffersFunction) (GList * buffers); - -static void -run_sync_test (SendBuffersFunction send_buffers, - CheckBuffersFunction check_buffers) -{ - GstSegment segment; - GstElement *bin, *audiomixer, *queue1, *queue2, *sink; - GstBus *bus; - GstPad *sinkpad1, *sinkpad2; - GstPad *queue1_sinkpad, *queue2_sinkpad; - GstPad *pad; - gboolean res; - GstStateChangeReturn state_res; - GstEvent *event; - GstCaps *caps; - GList *received_buffers = NULL; - - GST_INFO ("preparing test"); - - /* build pipeline */ - bin = gst_pipeline_new ("pipeline"); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - /* just an audiomixer and a fakesink */ - queue1 = gst_element_factory_make ("queue", "queue1"); - queue2 = gst_element_factory_make ("queue", "queue2"); - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - g_object_set (audiomixer, "output-buffer-duration", 500 * GST_MSECOND, NULL); - sink = gst_element_factory_make ("fakesink", "sink"); - g_object_set (sink, "signal-handoffs", TRUE, NULL); - g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_collect_cb, - &received_buffers); - gst_bin_add_many (GST_BIN (bin), queue1, queue2, audiomixer, sink, NULL); - - res = gst_element_link (audiomixer, sink); - fail_unless (res == TRUE, NULL); - - /* set to paused */ - state_res = gst_element_set_state (bin, GST_STATE_PAUSED); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - /* create an unconnected sinkpad in audiomixer, should also automatically activate - * the pad */ - sinkpad1 = gst_element_get_request_pad (audiomixer, "sink_%u"); - fail_if (sinkpad1 == NULL, NULL); - - queue1_sinkpad = gst_element_get_static_pad (queue1, "sink"); - pad = gst_element_get_static_pad (queue1, "src"); - fail_unless (gst_pad_link (pad, sinkpad1) == GST_PAD_LINK_OK); - gst_object_unref (pad); - - sinkpad2 = gst_element_get_request_pad (audiomixer, "sink_%u"); - fail_if (sinkpad2 == NULL, NULL); - - queue2_sinkpad = gst_element_get_static_pad (queue2, "sink"); - pad = gst_element_get_static_pad (queue2, "src"); - fail_unless (gst_pad_link (pad, sinkpad2) == GST_PAD_LINK_OK); - gst_object_unref (pad); - - gst_pad_send_event (queue1_sinkpad, gst_event_new_stream_start ("test")); - gst_pad_send_event (queue2_sinkpad, gst_event_new_stream_start ("test")); - - caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, GST_AUDIO_NE (S16), - "layout", G_TYPE_STRING, "interleaved", - "rate", G_TYPE_INT, 1000, "channels", G_TYPE_INT, 1, NULL); - - gst_pad_set_caps (queue1_sinkpad, caps); - gst_pad_set_caps (queue2_sinkpad, caps); - gst_caps_unref (caps); - - /* send segment to audiomixer */ - gst_segment_init (&segment, GST_FORMAT_TIME); - event = gst_event_new_segment (&segment); - gst_pad_send_event (queue1_sinkpad, gst_event_ref (event)); - gst_pad_send_event (queue2_sinkpad, event); - - /* Push buffers */ - send_buffers (queue1_sinkpad, queue2_sinkpad); - - /* Set PLAYING */ - g_idle_add ((GSourceFunc) set_playing, bin); - - /* Collect buffers and messages */ - g_main_loop_run (main_loop); - - /* Here we get once we got EOS, for errors we failed */ - - check_buffers (received_buffers); - - g_list_free_full (received_buffers, (GDestroyNotify) gst_buffer_unref); - - gst_element_release_request_pad (audiomixer, sinkpad1); - gst_object_unref (sinkpad1); - gst_object_unref (queue1_sinkpad); - gst_element_release_request_pad (audiomixer, sinkpad2); - gst_object_unref (sinkpad2); - gst_object_unref (queue2_sinkpad); - gst_element_set_state (bin, GST_STATE_NULL); - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); -} - -static void -send_buffers_sync (GstPad * pad1, GstPad * pad2) -{ - GstBuffer *buffer; - GstFlowReturn ret; - - buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad1, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - buffer = new_buffer (2000, 1, 2 * GST_SECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad1, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - gst_pad_send_event (pad1, gst_event_new_eos ()); - - buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad2, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad2, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - gst_pad_send_event (pad2, gst_event_new_eos ()); -} - -static void -check_buffers_sync (GList * received_buffers) -{ - GstBuffer *buffer; - GList *l; - gint i; - GstMapInfo map; - - /* Should have 8 * 0.5s buffers */ - fail_unless_equals_int (g_list_length (received_buffers), 8); - for (i = 0, l = received_buffers; l; l = l->next, i++) { - buffer = l->data; - - gst_buffer_map (buffer, &map, GST_MAP_READ); - - if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) { - fail_unless (map.data[0] == 0); - fail_unless (map.data[map.size - 1] == 0); - } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) { - fail_unless (map.data[0] == 0); - fail_unless (map.data[map.size - 1] == 0); - } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) { - fail_unless (map.data[0] == 1); - fail_unless (map.data[map.size - 1] == 1); - } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) { - fail_unless (map.data[0] == 1); - fail_unless (map.data[map.size - 1] == 1); - } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) { - fail_unless (map.data[0] == 3); - fail_unless (map.data[map.size - 1] == 3); - } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) { - fail_unless (map.data[0] == 3); - fail_unless (map.data[map.size - 1] == 3); - } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) { - fail_unless (map.data[0] == 2); - fail_unless (map.data[map.size - 1] == 2); - } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) { - fail_unless (map.data[0] == 2); - fail_unless (map.data[map.size - 1] == 2); - } else { - g_assert_not_reached (); - } - - gst_buffer_unmap (buffer, &map); - - } -} - -GST_START_TEST (test_sync) -{ - run_sync_test (send_buffers_sync, check_buffers_sync); -} - -GST_END_TEST; - -static void -send_buffers_sync_discont (GstPad * pad1, GstPad * pad2) -{ - GstBuffer *buffer; - GstFlowReturn ret; - - buffer = new_buffer (2000, 1, 1 * GST_SECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad1, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - buffer = new_buffer (2000, 1, 3 * GST_SECOND, 1 * GST_SECOND, - GST_BUFFER_FLAG_DISCONT); - ret = gst_pad_chain (pad1, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - gst_pad_send_event (pad1, gst_event_new_eos ()); - - buffer = new_buffer (2000, 2, 2 * GST_SECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad2, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - buffer = new_buffer (2000, 2, 3 * GST_SECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad2, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - gst_pad_send_event (pad2, gst_event_new_eos ()); -} - -static void -check_buffers_sync_discont (GList * received_buffers) -{ - GstBuffer *buffer; - GList *l; - gint i; - GstMapInfo map; - - /* Should have 8 * 0.5s buffers */ - fail_unless_equals_int (g_list_length (received_buffers), 8); - for (i = 0, l = received_buffers; l; l = l->next, i++) { - buffer = l->data; - - gst_buffer_map (buffer, &map, GST_MAP_READ); - - if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) { - fail_unless (map.data[0] == 0); - fail_unless (map.data[map.size - 1] == 0); - } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) { - fail_unless (map.data[0] == 0); - fail_unless (map.data[map.size - 1] == 0); - } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) { - fail_unless (map.data[0] == 1); - fail_unless (map.data[map.size - 1] == 1); - } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) { - fail_unless (map.data[0] == 1); - fail_unless (map.data[map.size - 1] == 1); - } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) { - fail_unless (map.data[0] == 2); - fail_unless (map.data[map.size - 1] == 2); - } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) { - fail_unless (map.data[0] == 2); - fail_unless (map.data[map.size - 1] == 2); - } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) { - fail_unless (map.data[0] == 3); - fail_unless (map.data[map.size - 1] == 3); - } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) { - fail_unless (map.data[0] == 3); - fail_unless (map.data[map.size - 1] == 3); - } else { - g_assert_not_reached (); - } - - gst_buffer_unmap (buffer, &map); - - } -} - -GST_START_TEST (test_sync_discont) -{ - run_sync_test (send_buffers_sync_discont, check_buffers_sync_discont); -} - -GST_END_TEST; - -static void -send_buffers_sync_unaligned (GstPad * pad1, GstPad * pad2) -{ - GstBuffer *buffer; - GstFlowReturn ret; - - buffer = new_buffer (2000, 1, 750 * GST_MSECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad1, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - buffer = new_buffer (2000, 1, 1750 * GST_MSECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad1, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - gst_pad_send_event (pad1, gst_event_new_eos ()); - - buffer = new_buffer (2000, 2, 1750 * GST_MSECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad2, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - buffer = new_buffer (2000, 2, 2750 * GST_MSECOND, 1 * GST_SECOND, 0); - ret = gst_pad_chain (pad2, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - - gst_pad_send_event (pad2, gst_event_new_eos ()); -} - -static void -check_buffers_sync_unaligned (GList * received_buffers) -{ - GstBuffer *buffer; - GList *l; - gint i; - GstMapInfo map; - - /* Should have 8 * 0.5s buffers */ - fail_unless_equals_int (g_list_length (received_buffers), 8); - for (i = 0, l = received_buffers; l; l = l->next, i++) { - buffer = l->data; - - gst_buffer_map (buffer, &map, GST_MAP_READ); - - if (i == 0 && GST_BUFFER_TIMESTAMP (buffer) == 0) { - fail_unless (map.data[0] == 0); - fail_unless (map.data[map.size - 1] == 0); - } else if (i == 1 && GST_BUFFER_TIMESTAMP (buffer) == 500 * GST_MSECOND) { - fail_unless (map.data[0] == 0); - fail_unless (map.data[499] == 0); - fail_unless (map.data[500] == 1); - fail_unless (map.data[map.size - 1] == 1); - } else if (i == 2 && GST_BUFFER_TIMESTAMP (buffer) == 1000 * GST_MSECOND) { - fail_unless (map.data[0] == 1); - fail_unless (map.data[map.size - 1] == 1); - } else if (i == 3 && GST_BUFFER_TIMESTAMP (buffer) == 1500 * GST_MSECOND) { - fail_unless (map.data[0] == 1); - fail_unless (map.data[499] == 1); - fail_unless (map.data[500] == 3); - fail_unless (map.data[map.size - 1] == 3); - } else if (i == 4 && GST_BUFFER_TIMESTAMP (buffer) == 2000 * GST_MSECOND) { - fail_unless (map.data[0] == 3); - fail_unless (map.data[499] == 3); - fail_unless (map.data[500] == 3); - fail_unless (map.data[map.size - 1] == 3); - } else if (i == 5 && GST_BUFFER_TIMESTAMP (buffer) == 2500 * GST_MSECOND) { - fail_unless (map.data[0] == 3); - fail_unless (map.data[499] == 3); - fail_unless (map.data[500] == 2); - fail_unless (map.data[map.size - 1] == 2); - } else if (i == 6 && GST_BUFFER_TIMESTAMP (buffer) == 3000 * GST_MSECOND) { - fail_unless (map.data[0] == 2); - fail_unless (map.data[499] == 2); - fail_unless (map.data[500] == 2); - fail_unless (map.data[map.size - 1] == 2); - } else if (i == 7 && GST_BUFFER_TIMESTAMP (buffer) == 3500 * GST_MSECOND) { - fail_unless (map.size == 500); - fail_unless (GST_BUFFER_DURATION (buffer) == 250 * GST_MSECOND); - fail_unless (map.data[0] == 2); - fail_unless (map.data[499] == 2); - } else { - g_assert_not_reached (); - } - - gst_buffer_unmap (buffer, &map); - - } -} - -GST_START_TEST (test_sync_unaligned) -{ - run_sync_test (send_buffers_sync_unaligned, check_buffers_sync_unaligned); -} - -GST_END_TEST; - -GST_START_TEST (test_segment_base_handling) -{ - GstElement *pipeline, *sink, *mix, *src1, *src2; - GstPad *srcpad, *sinkpad; - GstClockTime end_time; - GstSample *last_sample = NULL; - GstSample *sample; - GstBuffer *buf; - GstCaps *caps; - - caps = gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, 44100, - "channels", G_TYPE_INT, 2, NULL); - - pipeline = gst_pipeline_new ("pipeline"); - mix = gst_element_factory_make ("audiomixer", "audiomixer"); - sink = gst_element_factory_make ("appsink", "sink"); - g_object_set (sink, "caps", caps, "sync", FALSE, NULL); - gst_caps_unref (caps); - /* 50 buffers of 1/10 sec = 5 sec */ - src1 = gst_element_factory_make ("audiotestsrc", "src1"); - g_object_set (src1, "samplesperbuffer", 4410, "num-buffers", 50, NULL); - src2 = gst_element_factory_make ("audiotestsrc", "src2"); - g_object_set (src2, "samplesperbuffer", 4410, "num-buffers", 50, NULL); - gst_bin_add_many (GST_BIN (pipeline), src1, src2, mix, sink, NULL); - fail_unless (gst_element_link (mix, sink)); - - srcpad = gst_element_get_static_pad (src1, "src"); - sinkpad = gst_element_get_request_pad (mix, "sink_1"); - fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK); - gst_object_unref (sinkpad); - gst_object_unref (srcpad); - - srcpad = gst_element_get_static_pad (src2, "src"); - sinkpad = gst_element_get_request_pad (mix, "sink_2"); - fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK); - /* set a pad offset of another 5 seconds */ - gst_pad_set_offset (sinkpad, 5 * GST_SECOND); - gst_object_unref (sinkpad); - gst_object_unref (srcpad); - - gst_element_set_state (pipeline, GST_STATE_PLAYING); - - do { - g_signal_emit_by_name (sink, "pull-sample", &sample); - if (sample == NULL) - break; - if (last_sample) - gst_sample_unref (last_sample); - last_sample = sample; - } while (TRUE); - - buf = gst_sample_get_buffer (last_sample); - end_time = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf); - fail_unless_equals_int64 (end_time, 10 * GST_SECOND); - gst_sample_unref (last_sample); - - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_object_unref (pipeline); -} - -GST_END_TEST; - -static void -set_pad_volume_fade (GstPad * pad, GstClockTime start, gdouble start_value, - GstClockTime end, gdouble end_value) -{ - GstControlSource *cs; - GstTimedValueControlSource *tvcs; - - cs = gst_interpolation_control_source_new (); - fail_unless (gst_object_add_control_binding (GST_OBJECT_CAST (pad), - gst_direct_control_binding_new_absolute (GST_OBJECT_CAST (pad), - "volume", cs))); - - /* set volume interpolation mode */ - g_object_set (cs, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL); - - tvcs = (GstTimedValueControlSource *) cs; - fail_unless (gst_timed_value_control_source_set (tvcs, start, start_value)); - fail_unless (gst_timed_value_control_source_set (tvcs, end, end_value)); - gst_object_unref (cs); -} - -GST_START_TEST (test_sinkpad_property_controller) -{ - GstBus *bus; - GstMessage *msg; - GstElement *pipeline, *sink, *mix, *src1; - GstPad *srcpad, *sinkpad; - GError *error = NULL; - gchar *debug; - - pipeline = gst_pipeline_new ("pipeline"); - mix = gst_element_factory_make ("audiomixer", "audiomixer"); - sink = gst_element_factory_make ("fakesink", "sink"); - src1 = gst_element_factory_make ("audiotestsrc", "src1"); - g_object_set (src1, "num-buffers", 100, NULL); - gst_bin_add_many (GST_BIN (pipeline), src1, mix, sink, NULL); - fail_unless (gst_element_link (mix, sink)); - - srcpad = gst_element_get_static_pad (src1, "src"); - sinkpad = gst_element_get_request_pad (mix, "sink_0"); - fail_unless (gst_pad_link (srcpad, sinkpad) == GST_PAD_LINK_OK); - set_pad_volume_fade (sinkpad, 0, 0, 1.0, 2.0); - gst_object_unref (sinkpad); - gst_object_unref (srcpad); - - gst_element_set_state (pipeline, GST_STATE_PLAYING); - - bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); - msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE, - GST_MESSAGE_EOS | GST_MESSAGE_ERROR); - switch (GST_MESSAGE_TYPE (msg)) { - case GST_MESSAGE_ERROR: - gst_message_parse_error (msg, &error, &debug); - g_printerr ("ERROR from element %s: %s\n", - GST_OBJECT_NAME (msg->src), error->message); - g_printerr ("Debug info: %s\n", debug); - g_error_free (error); - g_free (debug); - break; - case GST_MESSAGE_EOS: - break; - default: - g_assert_not_reached (); - } - gst_message_unref (msg); - g_object_unref (bus); - - gst_element_set_state (pipeline, GST_STATE_NULL); - gst_object_unref (pipeline); -} - -GST_END_TEST; - -static void -change_src_caps (GstElement * fakesink, GstBuffer * buffer, GstPad * pad, - GstElement * capsfilter) -{ - GstCaps *caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, GST_AUDIO_NE (S32), - "layout", G_TYPE_STRING, "interleaved", - "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL); - - g_object_set (capsfilter, "caps", caps, NULL); - g_signal_connect (fakesink, "handoff", (GCallback) handoff_buffer_cb, NULL); - g_signal_handlers_disconnect_by_func (fakesink, change_src_caps, capsfilter); -} - -/* In this test, we create an input buffer with a duration of 2 seconds, - * and require the audiomixer to output 1 second long buffers. - * The input buffer will thus be mixed twice, and the audiomixer will - * output two buffers. - * - * After audiomixer has output a first buffer, we change its output format - * from S8 to S32. As our sample rate stays the same at 10 fps, and we use - * mono, the first buffer should be 10 bytes long, and the second 40. - * - * The input buffer is made up of 15 0-valued bytes, and 5 1-valued bytes. - * We verify that the second buffer contains 5 0-valued integers, and - * 5 1 << 24 valued integers. - */ -GST_START_TEST (test_change_output_caps) -{ - GstSegment segment; - GstElement *bin, *audiomixer, *capsfilter, *sink; - GstBus *bus; - GstPad *sinkpad; - gboolean res; - GstStateChangeReturn state_res; - GstFlowReturn ret; - GstEvent *event; - GstBuffer *buffer; - GstCaps *caps; - GstQuery *drain = gst_query_new_drain (); - GstMapInfo inmap; - GstMapInfo outmap; - gsize i; - - bin = gst_pipeline_new ("pipeline"); - bus = gst_element_get_bus (bin); - gst_bus_add_signal_watch_full (bus, G_PRIORITY_HIGH); - - g_signal_connect (bus, "message::error", (GCallback) message_received, bin); - g_signal_connect (bus, "message::warning", (GCallback) message_received, bin); - g_signal_connect (bus, "message::eos", (GCallback) message_received, bin); - - audiomixer = gst_element_factory_make ("audiomixer", "audiomixer"); - g_object_set (audiomixer, "output-buffer-duration", GST_SECOND, NULL); - capsfilter = gst_element_factory_make ("capsfilter", NULL); - sink = gst_element_factory_make ("fakesink", "sink"); - g_object_set (sink, "signal-handoffs", TRUE, NULL); - g_signal_connect (sink, "handoff", (GCallback) change_src_caps, capsfilter); - gst_bin_add_many (GST_BIN (bin), audiomixer, capsfilter, sink, NULL); - - res = gst_element_link_many (audiomixer, capsfilter, sink, NULL); - fail_unless (res == TRUE, NULL); - - state_res = gst_element_set_state (bin, GST_STATE_PLAYING); - ck_assert_int_ne (state_res, GST_STATE_CHANGE_FAILURE); - - sinkpad = gst_element_get_request_pad (audiomixer, "sink_%u"); - fail_if (sinkpad == NULL, NULL); - - gst_pad_send_event (sinkpad, gst_event_new_stream_start ("test")); - - caps = gst_caps_new_simple ("audio/x-raw", - "format", G_TYPE_STRING, "S8", - "layout", G_TYPE_STRING, "interleaved", - "rate", G_TYPE_INT, 10, "channels", G_TYPE_INT, 1, NULL); - - gst_pad_set_caps (sinkpad, caps); - g_object_set (capsfilter, "caps", caps, NULL); - gst_caps_unref (caps); - - gst_segment_init (&segment, GST_FORMAT_TIME); - segment.start = 0; - segment.stop = 2 * GST_SECOND; - segment.time = 0; - event = gst_event_new_segment (&segment); - gst_pad_send_event (sinkpad, event); - - gst_buffer_replace (&handoff_buffer, NULL); - - buffer = new_buffer (20, 0, 0, 2 * GST_SECOND, 0); - gst_buffer_map (buffer, &inmap, GST_MAP_WRITE); - memset (inmap.data + 15, 1, 5); - gst_buffer_unmap (buffer, &inmap); - ret = gst_pad_chain (sinkpad, buffer); - ck_assert_int_eq (ret, GST_FLOW_OK); - gst_pad_query (sinkpad, drain); - fail_unless (handoff_buffer != NULL); - fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 40); - - gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ); - for (i = 0; i < 10; i++) { - guint32 sample; - -#if G_BYTE_ORDER == G_LITTLE_ENDIAN - sample = GUINT32_FROM_LE (((guint32 *) outmap.data)[i]); -#else - sample = GUINT32_FROM_BE (((guint32 *) outmap.data)[i]); -#endif - - if (i < 5) { - fail_unless_equals_int (sample, 0); - } else { - fail_unless_equals_int (sample, 1 << 24); - } - } - gst_buffer_unmap (handoff_buffer, &outmap); - - gst_element_release_request_pad (audiomixer, sinkpad); - gst_object_unref (sinkpad); - gst_element_set_state (bin, GST_STATE_NULL); - gst_bus_remove_signal_watch (bus); - gst_object_unref (bus); - gst_object_unref (bin); - gst_query_unref (drain); -} - -GST_END_TEST; - -static Suite * -audiomixer_suite (void) -{ - Suite *s = suite_create ("audiomixer"); - TCase *tc_chain = tcase_create ("general"); - - suite_add_tcase (s, tc_chain); - tcase_add_test (tc_chain, test_caps); - tcase_add_test (tc_chain, test_filter_caps); - tcase_add_test (tc_chain, test_event); - tcase_add_test (tc_chain, test_play_twice); - tcase_add_test (tc_chain, test_play_twice_then_add_and_play_again); - tcase_add_test (tc_chain, test_live_seeking); - tcase_add_test (tc_chain, test_add_pad); - tcase_add_test (tc_chain, test_remove_pad); - tcase_add_test (tc_chain, test_clip); - tcase_add_test (tc_chain, test_duration_is_max); - tcase_add_test (tc_chain, test_duration_unknown_overrides); - tcase_add_test (tc_chain, test_loop); - tcase_add_test (tc_chain, test_flush_start_flush_stop); - tcase_add_test (tc_chain, test_sync); - tcase_add_test (tc_chain, test_sync_discont); - tcase_add_test (tc_chain, test_sync_unaligned); - tcase_add_test (tc_chain, test_segment_base_handling); - tcase_add_test (tc_chain, test_sinkpad_property_controller); - tcase_add_checked_fixture (tc_chain, test_setup, test_teardown); - tcase_add_test (tc_chain, test_change_output_caps); - - /* Use a longer timeout */ -#ifdef HAVE_VALGRIND - if (RUNNING_ON_VALGRIND) { - tcase_set_timeout (tc_chain, 5 * 60); - } else -#endif - { - /* this is shorter than the default 60 seconds?! (tpm) */ - /* tcase_set_timeout (tc_chain, 6); */ - } - - return s; -} - -GST_CHECK_MAIN (audiomixer); diff --git a/tests/check/meson.build b/tests/check/meson.build index 55f1513e86..1cb817164c 100644 --- a/tests/check/meson.build +++ b/tests/check/meson.build @@ -18,8 +18,6 @@ base_tests = [ [['elements/aiffparse.c']], [['elements/asfmux.c']], [['elements/assrender.c'], not ass_dep.found(), [ass_dep]], - [['elements/audiointerleave.c']], - [['elements/audiomixer.c']], [['elements/autoconvert.c']], [['elements/autovideoconvert.c']], [['elements/camerabin.c']],