From 10835e99198b4dde6eb71ab0b9d47f22e0159193 Mon Sep 17 00:00:00 2001 From: Mathieu Duponchelle Date: Wed, 28 Feb 2018 18:13:10 +0100 Subject: [PATCH] audioaggregator: refactor conversion API For the rationale, see: https://bugzilla.gnome.org/show_bug.cgi?id=793917 Also test audiomixer conversion of current output buffer --- gst-libs/gst/audio/gstaudioaggregator.c | 141 +++++++++++++------------------- gst-libs/gst/audio/gstaudioaggregator.h | 20 ++--- gst/audiomixer/gstaudiointerleave.c | 5 +- gst/audiomixer/gstaudiomixer.c | 4 +- tests/check/elements/audiomixer.c | 136 ++++++++++++++++++++++++++++++ 5 files changed, 207 insertions(+), 99 deletions(-) diff --git a/gst-libs/gst/audio/gstaudioaggregator.c b/gst-libs/gst/audio/gstaudioaggregator.c index a6ae491..f329448 100644 --- a/gst-libs/gst/audio/gstaudioaggregator.c +++ b/gst-libs/gst/audio/gstaudioaggregator.c @@ -28,22 +28,24 @@ * @title: GstAudioAggregator * @short_description: Base class that manages a set of audio input pads * with the purpose of aggregating or mixing their raw audio input buffers - * @see_also: #GstAggregator + * @see_also: #GstAggregator, #GstAudioMixer * - * #GstAudioAggregator will perform conversion on the data arriving - * on its sink pads, based on the format expected downstream. + * Subclasses must use (a subclass of) #GstAudioAggregatorPad for both + * their source and sink pads, + * gst_element_class_add_static_pad_template_with_gtype() is a convenient + * helper. * - * Subclasses can opt out of the conversion behaviour by setting - * #GstAudioAggregatorClass.convert_buffer() to %NULL. + * #GstAudioAggregator can perform conversion on the data arriving + * on its sink pads, based on the format expected downstream: in order + * to enable that behaviour, the GType of the sink pads must either be + * a (subclass of) #GstAudioAggregatorConvertPad to use the default + * #GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad + * implementing #GstAudioAggregatorPad.convert_buffer. * - * Subclasses that wish to use the default conversion implementation - * should use a (subclass of) #GstAudioAggregatorConvertPad as their - * #GstAggregatorClass.sinkpads_type, as it will cache the created - * #GstAudioConverter and install a property allowing to configure it, - * #GstAudioAggregatorPad:converter-config. + * To allow for the output caps to change, the mechanism is the same as + * above, with the GType of the source pad. * - * Subclasses that wish to perform custom conversion should override - * #GstAudioAggregatorClass.convert_buffer(). + * See #GstAudioMixer for an example. * * When conversion is enabled, #GstAudioAggregator will accept * any type of raw audio caps and perform conversion @@ -54,10 +56,6 @@ * the first configured sink pad to finish fixating its source pad * caps. * - * Additionally, handling audio conversion directly in the element - * means that this base class supports safely reconfiguring its - * source pad. - * * A notable exception for now is the sample rate, sink pads must * have the same sample rate as either the downstream requirement, * or the first configured pad, or a combination of both (when @@ -223,12 +221,22 @@ gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad aaggcpad->priv->converter_config_changed = FALSE; } +static void +gst_audio_aggregator_pad_update_conversion_info (GstAudioAggregatorPad * + aaggpad) +{ + GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->priv->converter_config_changed = + TRUE; +} + static GstBuffer * -gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorConvertPad * - aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info, +gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorPad * + aaggpad, GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * input_buffer) { GstBuffer *res; + GstAudioAggregatorConvertPad *aaggcpad = + GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad); gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info, out_info); @@ -327,6 +335,8 @@ gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; + GstAudioAggregatorPadClass *aaggpad_class = + (GstAudioAggregatorPadClass *) klass; g_type_class_add_private (klass, sizeof (GstAudioAggregatorConvertPadPrivate)); @@ -339,6 +349,12 @@ gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass * "when converting this pad's audio buffers", GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + aaggpad_class->convert_buffer = + gst_audio_aggregator_convert_pad_convert_buffer; + + aaggpad_class->update_conversion_info = + gst_audio_aggregator_pad_update_conversion_info; + gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize; } @@ -450,63 +466,15 @@ gst_audio_aggregator_get_next_time (GstAggregator * agg) } static GstBuffer * -gst_audio_aggregator_convert_once (GstAudioAggregator * aagg, GstPad * pad, - GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer) -{ - GstAudioConverter *converter = - gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE, - in_info, out_info, NULL); - gint insize = gst_buffer_get_size (buffer); - gsize insamples = insize / in_info->bpf; - gsize outsamples = gst_audio_converter_get_out_frames (converter, - insamples); - gint outsize = outsamples * out_info->bpf; - GstMapInfo inmap, outmap; - GstBuffer *converted = gst_buffer_new_allocate (NULL, outsize, NULL); - - gst_buffer_copy_into (converted, buffer, - GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS | - GST_BUFFER_COPY_META, 0, -1); - - gst_buffer_map (buffer, &inmap, GST_MAP_READ); - gst_buffer_map (converted, &outmap, GST_MAP_WRITE); - - gst_audio_converter_samples (converter, - GST_AUDIO_CONVERTER_FLAG_NONE, - (gpointer *) & inmap.data, insamples, - (gpointer *) & outmap.data, outsamples); - - gst_buffer_unmap (buffer, &inmap); - gst_buffer_unmap (converted, &outmap); - gst_audio_converter_free (converter); - - return converted; -} - -static GstBuffer * -gst_audio_aggregator_default_convert_buffer (GstAudioAggregator * aagg, - GstPad * pad, GstAudioInfo * in_info, GstAudioInfo * out_info, - GstBuffer * buffer) -{ - if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad)) - return - gst_audio_aggregator_convert_pad_convert_buffer - (GST_AUDIO_AGGREGATOR_CONVERT_PAD (pad), - &GST_AUDIO_AGGREGATOR_PAD (pad)->info, out_info, buffer); - else - return gst_audio_aggregator_convert_once (aagg, pad, in_info, out_info, - buffer); -} - -static GstBuffer * gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad, GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer) { - GstAudioAggregatorClass *klass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg); + GstAudioAggregatorPadClass *klass = GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad); + GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (pad); g_assert (klass->convert_buffer); - return klass->convert_buffer (aagg, pad, in_info, out_info, buffer); + return klass->convert_buffer (aaggpad, in_info, out_info, buffer); } static void @@ -542,7 +510,6 @@ gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass) gst_audio_aggregator_negotiated_src_caps; klass->create_output_buffer = gst_audio_aggregator_create_output_buffer; - klass->convert_buffer = gst_audio_aggregator_default_convert_buffer; GST_DEBUG_CATEGORY_INIT (audio_aggregator_debug, "audioaggregator", GST_DEBUG_FG_MAGENTA, "GstAudioAggregator"); @@ -746,11 +713,12 @@ gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad, gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ()); ret = FALSE; } else { + GstAudioAggregatorPadClass *klass = + GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad); GST_OBJECT_LOCK (aaggpad); gst_audio_info_from_caps (&aaggpad->info, caps); - if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)) - GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)-> - priv->converter_config_changed = TRUE; + if (klass->update_conversion_info) + klass->update_conversion_info (aaggpad); GST_OBJECT_UNLOCK (aaggpad); } @@ -792,10 +760,9 @@ gst_audio_aggregator_update_src_caps (GstAggregator * agg, static GstCaps * gst_audio_aggregator_fixate_src_caps (GstAggregator * agg, GstCaps * caps) { - GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg); GstAudioAggregatorPad *first_configured_pad; - if (!aaggclass->convert_buffer) + if (!GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer) return GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->fixate_src_caps (agg, caps); @@ -844,10 +811,11 @@ gst_audio_aggregator_update_converters (GstAudioAggregator * aagg, for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) { GstAudioAggregatorPad *aaggpad = l->data; + GstAudioAggregatorPadClass *klass = + GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad); - if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)) - GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)-> - priv->converter_config_changed = TRUE; + if (klass->update_conversion_info) + klass->update_conversion_info (aaggpad); /* If we currently were mixing a buffer, we need to convert it to the new * format */ @@ -865,7 +833,6 @@ static gboolean gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); - GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (agg); GstAudioInfo info; GST_INFO_OBJECT (agg, "src caps negotiated %" GST_PTR_FORMAT, caps); @@ -878,12 +845,19 @@ gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, GstCaps * caps) GST_AUDIO_AGGREGATOR_LOCK (aagg); GST_OBJECT_LOCK (aagg); - if (aaggclass->convert_buffer) { + if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer) { gst_audio_aggregator_update_converters (aagg, &info); if (aagg->priv->current_buffer && !gst_audio_info_is_equal (&aagg->info, &info)) { - GstBuffer *converted = + GstBuffer *converted; + GstAudioAggregatorPadClass *klass = + GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad); + + if (klass->update_conversion_info) + klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->srcpad)); + + converted = gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &aagg->info, &info, aagg->priv->current_buffer); gst_buffer_unref (aagg->priv->current_buffer); @@ -1324,7 +1298,6 @@ static gboolean gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad) { - GstAudioAggregatorClass *aaggclass = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg); GstClockTime start_time, end_time; gboolean discont = FALSE; guint64 start_offset, end_offset; @@ -1333,7 +1306,7 @@ gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg, GstAggregator *agg = GST_AGGREGATOR (aagg); GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad); - if (aaggclass->convert_buffer) { + if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) { rate = GST_AUDIO_INFO_RATE (&aagg->info); bpf = GST_AUDIO_INFO_BPF (&aagg->info); } else { @@ -1802,7 +1775,7 @@ gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout) /* New buffer? */ if (!pad->priv->buffer) { - if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad)) + if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) pad->priv->buffer = gst_audio_aggregator_convert_buffer (aagg, GST_PAD (pad), &pad->info, &aagg->info, diff --git a/gst-libs/gst/audio/gstaudioaggregator.h b/gst-libs/gst/audio/gstaudioaggregator.h index a608983..5c48fbf 100644 --- a/gst-libs/gst/audio/gstaudioaggregator.h +++ b/gst-libs/gst/audio/gstaudioaggregator.h @@ -79,12 +79,21 @@ struct _GstAudioAggregatorPad /** * GstAudioAggregatorPadClass: - * + * @convert_buffer: Convert a buffer from one format to another. + * @update_conversion_info: Called when either the input or output + * formats have changed. */ struct _GstAudioAggregatorPadClass { GstAggregatorPadClass parent_class; + GstBuffer * (* convert_buffer) (GstAudioAggregatorPad * pad, + GstAudioInfo *in_info, + GstAudioInfo *out_info, + GstBuffer * buffer); + + void (* update_conversion_info) (GstAudioAggregatorPad *pad); + /*< private >*/ gpointer _gst_reserved[GST_PADDING_LARGE]; }; @@ -181,10 +190,6 @@ struct _GstAudioAggregator * buffer. The in_offset and out_offset are in "frames", which is * the size of a sample times the number of channels. Returns TRUE if * any non-silence was added to the buffer - * @convert_buffer: Convert a buffer from one format to another. The pad - * is either a sinkpad, when converting an input buffer, or the source pad, - * when converting the output buffer after a downstream format change is - * requested. */ struct _GstAudioAggregatorClass { GstAggregatorClass parent_class; @@ -194,11 +199,6 @@ struct _GstAudioAggregatorClass { gboolean (* aggregate_one_buffer) (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad, GstBuffer * inbuf, guint in_offset, GstBuffer * outbuf, guint out_offset, guint num_frames); - GstBuffer * (* convert_buffer) (GstAudioAggregator *aagg, - GstPad * pad, - GstAudioInfo *in_info, - GstAudioInfo *out_info, - GstBuffer * buffer); /*< private >*/ gpointer _gst_reserved[GST_PADDING_LARGE]; diff --git a/gst/audiomixer/gstaudiointerleave.c b/gst/audiomixer/gstaudiointerleave.c index 90ec363..31cfbf0 100644 --- a/gst/audiomixer/gstaudiointerleave.c +++ b/gst/audiomixer/gstaudiointerleave.c @@ -560,8 +560,8 @@ gst_audio_interleave_class_init (GstAudioInterleaveClass * klass) 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_src_template, GST_TYPE_AUDIO_AGGREGATOR_PAD); 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", @@ -580,7 +580,6 @@ gst_audio_interleave_class_init (GstAudioInterleaveClass * klass) 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 diff --git a/gst/audiomixer/gstaudiomixer.c b/gst/audiomixer/gstaudiomixer.c index a0f5690..6eb91e3 100644 --- a/gst/audiomixer/gstaudiomixer.c +++ b/gst/audiomixer/gstaudiomixer.c @@ -224,8 +224,8 @@ 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_src_template, GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD); gst_element_class_add_static_pad_template_with_gtype (gstelement_class, &gst_audiomixer_sink_template, GST_TYPE_AUDIO_MIXER_PAD); gst_element_class_set_static_metadata (gstelement_class, "AudioMixer", diff --git a/tests/check/elements/audiomixer.c b/tests/check/elements/audiomixer.c index 4a8a823..7db70a1 100644 --- a/tests/check/elements/audiomixer.c +++ b/tests/check/elements/audiomixer.c @@ -1849,6 +1849,141 @@ GST_START_TEST (test_change_output_caps) GST_END_TEST; +/* In this test, we create two input buffers with a duration of 1 second, + * and require the audiomixer to output 1.5 second long buffers. + * + * After we have input two buffers, we change the output format + * from S8 to S32, then push a last buffer. + * + * This makes audioaggregator convert its "half-mixed" current_buffer, + * we can then ensure that the second output buffer is as expected. + */ +GST_START_TEST (test_change_output_caps_mid_output_buffer) +{ + 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; + GstMapInfo inmap; + GstMapInfo outmap; + guint 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", 1500 * GST_MSECOND, NULL); + capsfilter = gst_element_factory_make ("capsfilter", NULL); + sink = gst_element_factory_make ("fakesink", "sink"); + 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 = 3 * GST_SECOND; + segment.time = 0; + event = gst_event_new_segment (&segment); + gst_pad_send_event (sinkpad, event); + + buffer = new_buffer (10, 0, 0, 1 * GST_SECOND, 0); + ret = gst_pad_chain (sinkpad, buffer); + ck_assert_int_eq (ret, GST_FLOW_OK); + + buffer = new_buffer (10, 0, 1 * GST_SECOND, 1 * GST_SECOND, 0); + gst_buffer_map (buffer, &inmap, GST_MAP_WRITE); + memset (inmap.data, 1, 10); + gst_buffer_unmap (buffer, &inmap); + ret = gst_pad_chain (sinkpad, buffer); + ck_assert_int_eq (ret, GST_FLOW_OK); + + drain = gst_query_new_drain (); + gst_pad_query (sinkpad, drain); + gst_query_unref (drain); + + 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); + gst_caps_unref (caps); + + gst_buffer_replace (&handoff_buffer, NULL); + g_object_set (sink, "signal-handoffs", TRUE, NULL); + g_signal_connect (sink, "handoff", (GCallback) handoff_buffer_cb, NULL); + + buffer = new_buffer (10, 0, 2 * GST_SECOND, 1 * GST_SECOND, 0); + gst_buffer_map (buffer, &inmap, GST_MAP_WRITE); + memset (inmap.data, 0, 10); + gst_buffer_unmap (buffer, &inmap); + ret = gst_pad_chain (sinkpad, buffer); + ck_assert_int_eq (ret, GST_FLOW_OK); + + drain = gst_query_new_drain (); + gst_pad_query (sinkpad, drain); + gst_query_unref (drain); + + fail_unless (handoff_buffer); + fail_unless_equals_int (gst_buffer_get_size (handoff_buffer), 60); + + gst_buffer_map (handoff_buffer, &outmap, GST_MAP_READ); + for (i = 0; i < 15; 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, 1 << 24); + } else { + fail_unless_equals_int (sample, 0); + } + } + + 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_END_TEST; static Suite * audiomixer_suite (void) { @@ -1876,6 +2011,7 @@ audiomixer_suite (void) 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); + tcase_add_test (tc_chain, test_change_output_caps_mid_output_buffer); /* Use a longer timeout */ #ifdef HAVE_VALGRIND -- 2.7.4