From 5b1a96884032fe0e9421169caca0b3edae915a75 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 13 Feb 2018 14:11:49 +0000 Subject: audioaggregator: remove, moved to -base https://bugzilla.gnome.org/show_bug.cgi?id=791218 --- docs/libs/gst-plugins-bad-libs.types | 4 - docs/plugins/gst-plugins-bad-plugins.hierarchy | 7 - gst-libs/gst/audio/Makefile.am | 3 +- gst-libs/gst/audio/gstaudioaggregator.c | 1995 ------------------------ gst-libs/gst/audio/gstaudioaggregator.h | 228 --- gst-libs/gst/audio/meson.build | 4 +- 6 files changed, 3 insertions(+), 2238 deletions(-) delete mode 100644 gst-libs/gst/audio/gstaudioaggregator.c delete mode 100644 gst-libs/gst/audio/gstaudioaggregator.h diff --git a/docs/libs/gst-plugins-bad-libs.types b/docs/libs/gst-plugins-bad-libs.types index 7b4851c42..ba43ffdbc 100644 --- a/docs/libs/gst-plugins-bad-libs.types +++ b/docs/libs/gst-plugins-bad-libs.types @@ -1,6 +1,5 @@ #include -#include #include #include #include @@ -9,9 +8,6 @@ #include #include -gst_audio_aggregator_get_type -gst_audio_aggregator_pad_get_type - gst_video_aggregator_get_type gst_video_aggregator_pad_get_type diff --git a/docs/plugins/gst-plugins-bad-plugins.hierarchy b/docs/plugins/gst-plugins-bad-plugins.hierarchy index 63c911ea3..9455b4786 100644 --- a/docs/plugins/gst-plugins-bad-plugins.hierarchy +++ b/docs/plugins/gst-plugins-bad-plugins.hierarchy @@ -17,10 +17,6 @@ GObject GstControlSource GstElement GstAggregator - GstAudioAggregator - GstAudioInterleave - GstAudioMixer - GstLiveAdder GstMXFMux GstVideoAggregator GstCompositor @@ -327,9 +323,6 @@ GObject GstGLContext GstPad GstAggregatorPad - GstAudioAggregatorPad - GstAudioInterleavePad - GstAudioMixerPad GstMXFMuxPad GstVideoAggregatorPad GstCompositorPad diff --git a/gst-libs/gst/audio/Makefile.am b/gst-libs/gst/audio/Makefile.am index ca9e3f7e4..89d8ce743 100644 --- a/gst-libs/gst/audio/Makefile.am +++ b/gst-libs/gst/audio/Makefile.am @@ -4,7 +4,6 @@ lib_LTLIBRARIES = libgstbadaudio-@GST_API_VERSION@.la CLEANFILES = libgstbadaudio_@GST_API_VERSION@_la_SOURCES = \ - gstaudioaggregator.c \ gstnonstreamaudiodecoder.c nodist_libgstbadaudio_@GST_API_VERSION@_la_SOURCES = $(BUILT_SOURCES) @@ -24,4 +23,4 @@ libgstbadaudio_@GST_API_VERSION@_la_LIBADD = \ libgstbadaudio_@GST_API_VERSION@_la_LDFLAGS = $(GST_LIB_LDFLAGS) $(GST_ALL_LDFLAGS) $(GST_LT_LDFLAGS) libgstaudio_@GST_API_VERSION@includedir = $(includedir)/gstreamer-@GST_API_VERSION@/gst/audio -libgstaudio_@GST_API_VERSION@include_HEADERS = gstaudioaggregator.h gstnonstreamaudiodecoder.h +libgstaudio_@GST_API_VERSION@include_HEADERS = gstnonstreamaudiodecoder.h diff --git a/gst-libs/gst/audio/gstaudioaggregator.c b/gst-libs/gst/audio/gstaudioaggregator.c deleted file mode 100644 index fa9911b31..000000000 --- a/gst-libs/gst/audio/gstaudioaggregator.c +++ /dev/null @@ -1,1995 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen - * 2001 Thomas - * 2005,2006 Wim Taymans - * 2013 Sebastian Dröge - * 2014 Collabora - * Olivier Crete - * - * gstaudioaggregator.c: - * - * 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: gstaudioaggregator - * @short_description: manages a set of pads with the purpose of - * aggregating their buffers for raw audio - * @see_also: #GstAggregator - * - * #GstAudioAggregator will perform conversion on the data arriving - * on its sink pads, based on the format expected downstream. - * - * Subclasses can opt out of the conversion behaviour by setting - * #GstAudioAggregator.convert_buffer() to %NULL. - * - * 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, - * #GstAudioAggregatorPadClass:converter-config. - * - * Subclasses that wish to perform custom conversion should override - * #GstAudioAggregator.convert_buffer(). - * - * When conversion is enabled, #GstAudioAggregator will accept - * any type of raw audio caps and perform conversion - * on the data arriving on its sink pads, with whatever downstream - * expects as the target format. - * - * In case downstream caps are not fully fixated, it will use - * 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 - * downstream specifies a range or a set of acceptable rates). - */ - - -#ifdef HAVE_CONFIG_H -# include "config.h" -#endif - -#include "gstaudioaggregator.h" - -#include - -GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug); -#define GST_CAT_DEFAULT audio_aggregator_debug - -struct _GstAudioAggregatorPadPrivate -{ - /* All members are protected by the pad object lock */ - - GstBuffer *buffer; /* current buffer we're mixing, for - comparison with a new input buffer from - aggregator to see if we need to update our - cached values. */ - - guint position, size; /* position in the input buffer and size of the - input buffer in number of samples */ - - GstBuffer *input_buffer; - - guint64 output_offset; /* Sample offset in output segment relative to - pad.segment.start that position refers to - in the current buffer. */ - - guint64 next_offset; /* Next expected sample offset relative to - pad.segment.start */ - - /* Last time we noticed a discont */ - GstClockTime discont_time; - - /* A new unhandled segment event has been received */ - gboolean new_segment; -}; - - -/***************************************** - * GstAudioAggregatorPad implementation * - *****************************************/ -G_DEFINE_TYPE (GstAudioAggregatorPad, gst_audio_aggregator_pad, - GST_TYPE_AGGREGATOR_PAD); - -enum -{ - PROP_PAD_0, - PROP_PAD_CONVERTER_CONFIG, -}; - -static GstFlowReturn -gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad, - GstAggregator * aggregator); - -static void -gst_audio_aggregator_pad_finalize (GObject * object) -{ - GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) object; - - gst_buffer_replace (&pad->priv->buffer, NULL); - gst_buffer_replace (&pad->priv->input_buffer, NULL); - - G_OBJECT_CLASS (gst_audio_aggregator_pad_parent_class)->finalize (object); -} - -static void -gst_audio_aggregator_pad_class_init (GstAudioAggregatorPadClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass; - - g_type_class_add_private (klass, sizeof (GstAudioAggregatorPadPrivate)); - - gobject_class->finalize = gst_audio_aggregator_pad_finalize; - aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audio_aggregator_pad_flush_pad); -} - -static void -gst_audio_aggregator_pad_init (GstAudioAggregatorPad * pad) -{ - pad->priv = - G_TYPE_INSTANCE_GET_PRIVATE (pad, GST_TYPE_AUDIO_AGGREGATOR_PAD, - GstAudioAggregatorPadPrivate); - - gst_audio_info_init (&pad->info); - - pad->priv->buffer = NULL; - pad->priv->input_buffer = NULL; - pad->priv->position = 0; - pad->priv->size = 0; - pad->priv->output_offset = -1; - pad->priv->next_offset = -1; - pad->priv->discont_time = GST_CLOCK_TIME_NONE; -} - - -static GstFlowReturn -gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad, - GstAggregator * aggregator) -{ - GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad); - - GST_OBJECT_LOCK (aggpad); - pad->priv->position = pad->priv->size = 0; - pad->priv->output_offset = pad->priv->next_offset = -1; - pad->priv->discont_time = GST_CLOCK_TIME_NONE; - gst_buffer_replace (&pad->priv->buffer, NULL); - gst_buffer_replace (&pad->priv->input_buffer, NULL); - GST_OBJECT_UNLOCK (aggpad); - - return GST_FLOW_OK; -} - -struct _GstAudioAggregatorConvertPadPrivate -{ - /* All members are protected by the pad object lock */ - GstAudioConverter *converter; - GstStructure *converter_config; - gboolean converter_config_changed; -}; - - -G_DEFINE_TYPE (GstAudioAggregatorConvertPad, gst_audio_aggregator_convert_pad, - GST_TYPE_AUDIO_AGGREGATOR_PAD); - -static void -gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad - * aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info) -{ - if (!aaggcpad->priv->converter_config_changed) - return; - - if (aaggcpad->priv->converter) { - gst_audio_converter_free (aaggcpad->priv->converter); - aaggcpad->priv->converter = NULL; - } - - if (gst_audio_info_is_equal (in_info, out_info) || - in_info->finfo->format == GST_AUDIO_FORMAT_UNKNOWN) { - if (aaggcpad->priv->converter) { - gst_audio_converter_free (aaggcpad->priv->converter); - aaggcpad->priv->converter = NULL; - } - } else { - /* If we haven't received caps yet, this pad should not have - * a buffer to convert anyway */ - aaggcpad->priv->converter = - gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE, - in_info, out_info, - aaggcpad->priv->converter_config ? gst_structure_copy (aaggcpad-> - priv->converter_config) : NULL); - } - - aaggcpad->priv->converter_config_changed = FALSE; -} - -static GstBuffer * -gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorConvertPad * - aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info, - GstBuffer * input_buffer) -{ - GstBuffer *res; - - gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info, - out_info); - - if (aaggcpad->priv->converter) { - gint insize = gst_buffer_get_size (input_buffer); - gsize insamples = insize / in_info->bpf; - gsize outsamples = - gst_audio_converter_get_out_frames (aaggcpad->priv->converter, - insamples); - gint outsize = outsamples * out_info->bpf; - GstMapInfo inmap, outmap; - - res = gst_buffer_new_allocate (NULL, outsize, NULL); - - /* We create a perfectly similar buffer, except obviously for - * its converted contents */ - gst_buffer_copy_into (res, input_buffer, - GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS | - GST_BUFFER_COPY_META, 0, -1); - - gst_buffer_map (input_buffer, &inmap, GST_MAP_READ); - gst_buffer_map (res, &outmap, GST_MAP_WRITE); - - gst_audio_converter_samples (aaggcpad->priv->converter, - GST_AUDIO_CONVERTER_FLAG_NONE, - (gpointer *) & inmap.data, insamples, - (gpointer *) & outmap.data, outsamples); - - gst_buffer_unmap (input_buffer, &inmap); - gst_buffer_unmap (res, &outmap); - } else { - res = gst_buffer_ref (input_buffer); - } - - return res; -} - -static void -gst_audio_aggregator_convert_pad_finalize (GObject * object) -{ - GstAudioAggregatorConvertPad *pad = (GstAudioAggregatorConvertPad *) object; - - if (pad->priv->converter) - gst_audio_converter_free (pad->priv->converter); - - if (pad->priv->converter_config) - gst_structure_free (pad->priv->converter_config); - - G_OBJECT_CLASS (gst_audio_aggregator_convert_pad_parent_class)->finalize - (object); -} - -static void -gst_audio_aggregator_convert_pad_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object); - - switch (prop_id) { - case PROP_PAD_CONVERTER_CONFIG: - GST_OBJECT_LOCK (pad); - if (pad->priv->converter_config) - g_value_set_boxed (value, pad->priv->converter_config); - GST_OBJECT_UNLOCK (pad); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_audio_aggregator_convert_pad_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object); - - switch (prop_id) { - case PROP_PAD_CONVERTER_CONFIG: - GST_OBJECT_LOCK (pad); - if (pad->priv->converter_config) - gst_structure_free (pad->priv->converter_config); - pad->priv->converter_config = g_value_dup_boxed (value); - pad->priv->converter_config_changed = TRUE; - GST_OBJECT_UNLOCK (pad); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass * - klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - g_type_class_add_private (klass, - sizeof (GstAudioAggregatorConvertPadPrivate)); - - gobject_class->set_property = gst_audio_aggregator_convert_pad_set_property; - gobject_class->get_property = gst_audio_aggregator_convert_pad_get_property; - - g_object_class_install_property (gobject_class, PROP_PAD_CONVERTER_CONFIG, - g_param_spec_boxed ("converter-config", "Converter configuration", - "A GstStructure describing the configuration that should be used " - "when converting this pad's audio buffers", - GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize; -} - -static void -gst_audio_aggregator_convert_pad_init (GstAudioAggregatorConvertPad * pad) -{ - pad->priv = - G_TYPE_INSTANCE_GET_PRIVATE (pad, GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD, - GstAudioAggregatorConvertPadPrivate); -} - -/************************************** - * GstAudioAggregator implementation * - **************************************/ - -struct _GstAudioAggregatorPrivate -{ - GMutex mutex; - - /* All three properties are unprotected, can't be modified while streaming */ - /* Size in frames that is output per buffer */ - GstClockTime output_buffer_duration; - GstClockTime alignment_threshold; - GstClockTime discont_wait; - - /* Protected by srcpad stream clock */ - /* Output buffer starting at offset containing blocksize frames (calculated - * from output_buffer_duration) */ - GstBuffer *current_buffer; - - /* counters to keep track of timestamps */ - /* Readable with object lock, writable with both aag lock and object lock */ - - /* Sample offset starting from 0 at aggregator.segment.start */ - gint64 offset; -}; - -#define GST_AUDIO_AGGREGATOR_LOCK(self) g_mutex_lock (&(self)->priv->mutex); -#define GST_AUDIO_AGGREGATOR_UNLOCK(self) g_mutex_unlock (&(self)->priv->mutex); - -static void gst_audio_aggregator_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_audio_aggregator_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static void gst_audio_aggregator_dispose (GObject * object); - -static gboolean gst_audio_aggregator_src_event (GstAggregator * agg, - GstEvent * event); -static gboolean gst_audio_aggregator_sink_event (GstAggregator * agg, - GstAggregatorPad * aggpad, GstEvent * event); -static gboolean gst_audio_aggregator_src_query (GstAggregator * agg, - GstQuery * query); -static gboolean -gst_audio_aggregator_sink_query (GstAggregator * agg, GstAggregatorPad * aggpad, - GstQuery * query); -static gboolean gst_audio_aggregator_start (GstAggregator * agg); -static gboolean gst_audio_aggregator_stop (GstAggregator * agg); -static GstFlowReturn gst_audio_aggregator_flush (GstAggregator * agg); - -static GstBuffer *gst_audio_aggregator_create_output_buffer (GstAudioAggregator - * aagg, guint num_frames); -static GstBuffer *gst_audio_aggregator_do_clip (GstAggregator * agg, - GstAggregatorPad * bpad, GstBuffer * buffer); -static GstFlowReturn gst_audio_aggregator_aggregate (GstAggregator * agg, - gboolean timeout); -static gboolean sync_pad_values (GstElement * aagg, GstPad * pad, gpointer ud); -static gboolean gst_audio_aggregator_negotiated_src_caps (GstAggregator * agg, - GstCaps * caps); -static GstFlowReturn -gst_audio_aggregator_update_src_caps (GstAggregator * agg, - GstCaps * caps, GstCaps ** ret); -static GstCaps *gst_audio_aggregator_fixate_src_caps (GstAggregator * agg, - GstCaps * caps); - -#define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND) -#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND) -#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND) - -enum -{ - PROP_0, - PROP_OUTPUT_BUFFER_DURATION, - PROP_ALIGNMENT_THRESHOLD, - PROP_DISCONT_WAIT, -}; - -G_DEFINE_ABSTRACT_TYPE (GstAudioAggregator, gst_audio_aggregator, - GST_TYPE_AGGREGATOR); - -static GstClockTime -gst_audio_aggregator_get_next_time (GstAggregator * agg) -{ - GstClockTime next_time; - - GST_OBJECT_LOCK (agg); - if (agg->segment.position == -1 || agg->segment.position < agg->segment.start) - next_time = agg->segment.start; - else - next_time = agg->segment.position; - - if (agg->segment.stop != -1 && next_time > agg->segment.stop) - next_time = agg->segment.stop; - - next_time = - gst_segment_to_running_time (&agg->segment, GST_FORMAT_TIME, next_time); - GST_OBJECT_UNLOCK (agg); - - return next_time; -} - -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); - - g_assert (klass->convert_buffer); - - return klass->convert_buffer (aagg, pad, in_info, out_info, buffer); -} - -static void -gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - GstAggregatorClass *gstaggregator_class = (GstAggregatorClass *) klass; - - g_type_class_add_private (klass, sizeof (GstAudioAggregatorPrivate)); - - gobject_class->set_property = gst_audio_aggregator_set_property; - gobject_class->get_property = gst_audio_aggregator_get_property; - gobject_class->dispose = gst_audio_aggregator_dispose; - - gstaggregator_class->src_event = - GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_event); - gstaggregator_class->sink_event = - GST_DEBUG_FUNCPTR (gst_audio_aggregator_sink_event); - gstaggregator_class->src_query = - GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_query); - gstaggregator_class->sink_query = gst_audio_aggregator_sink_query; - gstaggregator_class->start = gst_audio_aggregator_start; - gstaggregator_class->stop = gst_audio_aggregator_stop; - gstaggregator_class->flush = gst_audio_aggregator_flush; - gstaggregator_class->aggregate = - GST_DEBUG_FUNCPTR (gst_audio_aggregator_aggregate); - gstaggregator_class->clip = GST_DEBUG_FUNCPTR (gst_audio_aggregator_do_clip); - gstaggregator_class->get_next_time = gst_audio_aggregator_get_next_time; - gstaggregator_class->update_src_caps = - GST_DEBUG_FUNCPTR (gst_audio_aggregator_update_src_caps); - gstaggregator_class->fixate_src_caps = gst_audio_aggregator_fixate_src_caps; - gstaggregator_class->negotiated_src_caps = - 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"); - - g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION, - g_param_spec_uint64 ("output-buffer-duration", "Output Buffer Duration", - "Output block size in nanoseconds", 1, - G_MAXUINT64, DEFAULT_OUTPUT_BUFFER_DURATION, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD, - g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold", - "Timestamp alignment threshold in nanoseconds", 0, - G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT, - g_param_spec_uint64 ("discont-wait", "Discont Wait", - "Window of time in nanoseconds to wait before " - "creating a discontinuity", 0, - G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); -} - -static void -gst_audio_aggregator_init (GstAudioAggregator * aagg) -{ - aagg->priv = - G_TYPE_INSTANCE_GET_PRIVATE (aagg, GST_TYPE_AUDIO_AGGREGATOR, - GstAudioAggregatorPrivate); - - g_mutex_init (&aagg->priv->mutex); - - aagg->priv->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION; - aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD; - aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT; - - aagg->current_caps = NULL; - gst_audio_info_init (&aagg->info); - - gst_aggregator_set_latency (GST_AGGREGATOR (aagg), - aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration); -} - -static void -gst_audio_aggregator_dispose (GObject * object) -{ - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object); - - gst_caps_replace (&aagg->current_caps, NULL); - - g_mutex_clear (&aagg->priv->mutex); - - G_OBJECT_CLASS (gst_audio_aggregator_parent_class)->dispose (object); -} - -static void -gst_audio_aggregator_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object); - - switch (prop_id) { - case PROP_OUTPUT_BUFFER_DURATION: - aagg->priv->output_buffer_duration = g_value_get_uint64 (value); - gst_aggregator_set_latency (GST_AGGREGATOR (aagg), - aagg->priv->output_buffer_duration, - aagg->priv->output_buffer_duration); - break; - case PROP_ALIGNMENT_THRESHOLD: - aagg->priv->alignment_threshold = g_value_get_uint64 (value); - break; - case PROP_DISCONT_WAIT: - aagg->priv->discont_wait = g_value_get_uint64 (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_audio_aggregator_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object); - - switch (prop_id) { - case PROP_OUTPUT_BUFFER_DURATION: - g_value_set_uint64 (value, aagg->priv->output_buffer_duration); - break; - case PROP_ALIGNMENT_THRESHOLD: - g_value_set_uint64 (value, aagg->priv->alignment_threshold); - break; - case PROP_DISCONT_WAIT: - g_value_set_uint64 (value, aagg->priv->discont_wait); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -/* Caps negotiation */ - -/* Unref after usage */ -static GstAudioAggregatorPad * -gst_audio_aggregator_get_first_configured_pad (GstAggregator * agg) -{ - GstAudioAggregatorPad *res = NULL; - GList *l; - - GST_OBJECT_LOCK (agg); - for (l = GST_ELEMENT (agg)->sinkpads; l; l = l->next) { - GstAudioAggregatorPad *aaggpad = l->data; - - if (GST_AUDIO_INFO_FORMAT (&aaggpad->info) != GST_AUDIO_FORMAT_UNKNOWN) { - res = gst_object_ref (aaggpad); - break; - } - } - GST_OBJECT_UNLOCK (agg); - - return res; -} - -static GstCaps * -gst_audio_aggregator_sink_getcaps (GstPad * pad, GstAggregator * agg, - GstCaps * filter) -{ - GstAudioAggregatorPad *first_configured_pad = - gst_audio_aggregator_get_first_configured_pad (agg); - GstCaps *sink_template_caps = gst_pad_get_pad_template_caps (pad); - GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad); - GstCaps *sink_caps; - GstStructure *s, *s2; - gint downstream_rate; - - sink_template_caps = gst_caps_make_writable (sink_template_caps); - s = gst_caps_get_structure (sink_template_caps, 0); - - if (downstream_caps && !gst_caps_is_empty (downstream_caps)) - s2 = gst_caps_get_structure (downstream_caps, 0); - else - s2 = NULL; - - if (s2 && gst_structure_get_int (s2, "rate", &downstream_rate)) { - gst_structure_fixate_field_nearest_int (s, "rate", downstream_rate); - } else if (first_configured_pad) { - gst_structure_fixate_field_nearest_int (s, "rate", - first_configured_pad->info.rate); - } - - if (first_configured_pad) - gst_object_unref (first_configured_pad); - - sink_caps = filter ? gst_caps_intersect (sink_template_caps, - filter) : gst_caps_ref (sink_template_caps); - - GST_INFO_OBJECT (pad, "Getting caps with filter %" GST_PTR_FORMAT, filter); - GST_DEBUG_OBJECT (pad, "sink template caps : %" GST_PTR_FORMAT, - sink_template_caps); - GST_DEBUG_OBJECT (pad, "downstream caps %" GST_PTR_FORMAT, downstream_caps); - GST_INFO_OBJECT (pad, "returned sink caps : %" GST_PTR_FORMAT, sink_caps); - - gst_caps_unref (sink_template_caps); - - if (downstream_caps) - gst_caps_unref (downstream_caps); - - return sink_caps; -} - -static gboolean -gst_audio_aggregator_sink_setcaps (GstAudioAggregatorPad * aaggpad, - GstAggregator * agg, GstCaps * caps) -{ - GstAudioAggregatorPad *first_configured_pad = - gst_audio_aggregator_get_first_configured_pad (agg); - GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad); - GstAudioInfo info; - gboolean ret = TRUE; - gint downstream_rate; - GstStructure *s; - - if (!downstream_caps || gst_caps_is_empty (downstream_caps)) { - ret = FALSE; - goto done; - } - - gst_audio_info_from_caps (&info, caps); - s = gst_caps_get_structure (downstream_caps, 0); - - /* TODO: handle different rates on sinkpads, a bit complex - * because offsets will have to be updated, and audio resampling - * has a latency to take into account - */ - if ((gst_structure_get_int (s, "rate", &downstream_rate) - && info.rate != downstream_rate) || (first_configured_pad - && info.rate != first_configured_pad->info.rate)) { - gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ()); - ret = FALSE; - } else { - 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; - GST_OBJECT_UNLOCK (aaggpad); - } - -done: - if (first_configured_pad) - gst_object_unref (first_configured_pad); - - if (downstream_caps) - gst_caps_unref (downstream_caps); - - return ret; -} - -static GstFlowReturn -gst_audio_aggregator_update_src_caps (GstAggregator * agg, - GstCaps * caps, GstCaps ** ret) -{ - GstCaps *src_template_caps = gst_pad_get_pad_template_caps (agg->srcpad); - GstCaps *downstream_caps = - gst_pad_peer_query_caps (agg->srcpad, src_template_caps); - - gst_caps_unref (src_template_caps); - - *ret = gst_caps_intersect (caps, downstream_caps); - - GST_INFO ("Updated src caps to %" GST_PTR_FORMAT, *ret); - - if (downstream_caps) - gst_caps_unref (downstream_caps); - - return GST_FLOW_OK; -} - -/* At that point if the caps are not fixed, this means downstream - * didn't have fully specified requirements, we'll just go ahead - * and fixate raw audio fields using our first configured pad, we don't for - * now need a more complicated heuristic - */ -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) - return - GST_AGGREGATOR_CLASS - (gst_audio_aggregator_parent_class)->fixate_src_caps (agg, caps); - - first_configured_pad = gst_audio_aggregator_get_first_configured_pad (agg); - - if (first_configured_pad) { - GstStructure *s, *s2; - GstCaps *first_configured_caps = - gst_audio_info_to_caps (&first_configured_pad->info); - gint first_configured_rate, first_configured_channels; - - caps = gst_caps_make_writable (caps); - s = gst_caps_get_structure (caps, 0); - s2 = gst_caps_get_structure (first_configured_caps, 0); - - gst_structure_get_int (s2, "rate", &first_configured_rate); - gst_structure_get_int (s2, "channels", &first_configured_channels); - - gst_structure_fixate_field_string (s, "format", - gst_structure_get_string (s2, "format")); - gst_structure_fixate_field_string (s, "layout", - gst_structure_get_string (s2, "layout")); - gst_structure_fixate_field_nearest_int (s, "rate", first_configured_rate); - gst_structure_fixate_field_nearest_int (s, "channels", - first_configured_channels); - - gst_caps_unref (first_configured_caps); - gst_object_unref (first_configured_pad); - } - - if (!gst_caps_is_fixed (caps)) - caps = gst_caps_fixate (caps); - - GST_INFO_OBJECT (agg, "Fixated src caps to %" GST_PTR_FORMAT, caps); - - return caps; -} - -/* Must be called with OBJECT_LOCK taken */ -static void -gst_audio_aggregator_update_converters (GstAudioAggregator * aagg, - GstAudioInfo * new_info) -{ - GList *l; - - for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) { - GstAudioAggregatorPad *aaggpad = l->data; - - if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)) - GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)-> - priv->converter_config_changed = TRUE; - - /* If we currently were mixing a buffer, we need to convert it to the new - * format */ - if (aaggpad->priv->buffer) { - GstBuffer *new_converted_buffer = - gst_audio_aggregator_convert_buffer (aagg, GST_PAD (aaggpad), - &aaggpad->info, new_info, aaggpad->priv->input_buffer); - gst_buffer_replace (&aaggpad->priv->buffer, new_converted_buffer); - } - } -} - -/* We now have our final output caps, we can create the required converters */ -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); - - if (!gst_audio_info_from_caps (&info, caps)) { - GST_WARNING_OBJECT (aagg, "Rejecting invalid caps: %" GST_PTR_FORMAT, caps); - return FALSE; - } - - GST_AUDIO_AGGREGATOR_LOCK (aagg); - GST_OBJECT_LOCK (aagg); - - if (aaggclass->convert_buffer) { - gst_audio_aggregator_update_converters (aagg, &info); - - if (aagg->priv->current_buffer - && !gst_audio_info_is_equal (&aagg->info, &info)) { - GstBuffer *converted = - gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &aagg->info, - &info, aagg->priv->current_buffer); - gst_buffer_unref (aagg->priv->current_buffer); - aagg->priv->current_buffer = converted; - } - } - - if (!gst_audio_info_is_equal (&info, &aagg->info)) { - GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps); - gst_caps_replace (&aagg->current_caps, caps); - - memcpy (&aagg->info, &info, sizeof (info)); - } - - GST_OBJECT_UNLOCK (aagg); - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - - return - GST_AGGREGATOR_CLASS - (gst_audio_aggregator_parent_class)->negotiated_src_caps (agg, caps); -} - -/* event handling */ - -static gboolean -gst_audio_aggregator_src_event (GstAggregator * agg, GstEvent * event) -{ - gboolean result; - - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); - GST_DEBUG_OBJECT (agg->srcpad, "Got %s event on src pad", - GST_EVENT_TYPE_NAME (event)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_QOS: - /* QoS might be tricky */ - gst_event_unref (event); - return FALSE; - case GST_EVENT_NAVIGATION: - /* navigation is rather pointless. */ - gst_event_unref (event); - return FALSE; - break; - case GST_EVENT_SEEK: - { - GstSeekFlags flags; - gdouble rate; - GstSeekType start_type, stop_type; - gint64 start, stop; - GstFormat seek_format, dest_format; - - /* parse the seek parameters */ - gst_event_parse_seek (event, &rate, &seek_format, &flags, &start_type, - &start, &stop_type, &stop); - - /* Check the seeking parameters before linking up */ - if ((start_type != GST_SEEK_TYPE_NONE) - && (start_type != GST_SEEK_TYPE_SET)) { - result = FALSE; - GST_DEBUG_OBJECT (aagg, - "seeking failed, unhandled seek type for start: %d", start_type); - goto done; - } - if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) { - result = FALSE; - GST_DEBUG_OBJECT (aagg, - "seeking failed, unhandled seek type for end: %d", stop_type); - goto done; - } - - GST_OBJECT_LOCK (agg); - dest_format = agg->segment.format; - GST_OBJECT_UNLOCK (agg); - if (seek_format != dest_format) { - result = FALSE; - GST_DEBUG_OBJECT (aagg, - "seeking failed, unhandled seek format: %s", - gst_format_get_name (seek_format)); - goto done; - } - } - break; - default: - break; - } - - return - GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_event (agg, - event); - -done: - return result; -} - - -static gboolean -gst_audio_aggregator_sink_event (GstAggregator * agg, - GstAggregatorPad * aggpad, GstEvent * event) -{ - GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (aggpad); - 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_SEGMENT: - { - const GstSegment *segment; - gst_event_parse_segment (event, &segment); - - if (segment->format != GST_FORMAT_TIME) { - GST_ERROR_OBJECT (agg, "Segment of type %s are not supported," - " only TIME segments are supported", - gst_format_get_name (segment->format)); - gst_event_unref (event); - event = NULL; - res = FALSE; - break; - } - - GST_OBJECT_LOCK (agg); - if (segment->rate != agg->segment.rate) { - GST_ERROR_OBJECT (aggpad, - "Got segment event with wrong rate %lf, expected %lf", - segment->rate, agg->segment.rate); - res = FALSE; - gst_event_unref (event); - event = NULL; - } else if (segment->rate < 0.0) { - GST_ERROR_OBJECT (aggpad, "Negative rates not supported yet"); - res = FALSE; - gst_event_unref (event); - event = NULL; - } else { - GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad); - - GST_OBJECT_LOCK (pad); - pad->priv->new_segment = TRUE; - GST_OBJECT_UNLOCK (pad); - } - GST_OBJECT_UNLOCK (agg); - - break; - } - case GST_EVENT_CAPS: - { - GstCaps *caps; - - gst_event_parse_caps (event, &caps); - GST_INFO_OBJECT (aggpad, "Got caps %" GST_PTR_FORMAT, caps); - res = gst_audio_aggregator_sink_setcaps (aaggpad, agg, caps); - gst_event_unref (event); - event = NULL; - break; - } - default: - break; - } - - if (event != NULL) - return - GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_event - (agg, aggpad, event); - - return res; -} - -static gboolean -gst_audio_aggregator_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_aggregator_sink_getcaps (GST_PAD (aggpad), agg, filter); - gst_query_set_caps_result (query, caps); - gst_caps_unref (caps); - res = TRUE; - break; - } - default: - res = - GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_query - (agg, aggpad, query); - break; - } - - return res; -} - - -/* FIXME, the duration query should reflect how long you will produce - * data, that is the amount of stream time until you will emit EOS. - * - * For synchronized mixing this is always the max of all the durations - * of upstream since we emit EOS when all of them finished. - * - * We don't do synchronized mixing so this really depends on where the - * streams where punched in and what their relative offsets are against - * eachother which we can get from the first timestamps we see. - * - * When we add a new stream (or remove a stream) the duration might - * also become invalid again and we need to post a new DURATION - * message to notify this fact to the parent. - * For now we take the max of all the upstream elements so the simple - * cases work at least somewhat. - */ -static gboolean -gst_audio_aggregator_query_duration (GstAudioAggregator * aagg, - GstQuery * query) -{ - gint64 max; - gboolean res; - GstFormat format; - GstIterator *it; - gboolean done; - GValue item = { 0, }; - - /* parse format */ - gst_query_parse_duration (query, &format, NULL); - - max = -1; - res = TRUE; - done = FALSE; - - it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (aagg)); - while (!done) { - GstIteratorResult ires; - - ires = gst_iterator_next (it, &item); - switch (ires) { - case GST_ITERATOR_DONE: - done = TRUE; - break; - case GST_ITERATOR_OK: - { - GstPad *pad = g_value_get_object (&item); - gint64 duration; - - /* ask sink peer for duration */ - res &= gst_pad_peer_query_duration (pad, format, &duration); - /* take max from all valid return values */ - if (res) { - /* valid unknown length, stop searching */ - if (duration == -1) { - max = duration; - done = TRUE; - } - /* else see if bigger than current max */ - else if (duration > max) - max = duration; - } - g_value_reset (&item); - break; - } - case GST_ITERATOR_RESYNC: - max = -1; - res = TRUE; - gst_iterator_resync (it); - break; - default: - res = FALSE; - done = TRUE; - break; - } - } - g_value_unset (&item); - gst_iterator_free (it); - - if (res) { - /* and store the max */ - GST_DEBUG_OBJECT (aagg, "Total duration in format %s: %" - GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max)); - gst_query_set_duration (query, format, max); - } - - return res; -} - - -static gboolean -gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query) -{ - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); - gboolean res = FALSE; - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_DURATION: - res = gst_audio_aggregator_query_duration (aagg, query); - break; - case GST_QUERY_POSITION: - { - GstFormat format; - - gst_query_parse_position (query, &format, NULL); - - GST_OBJECT_LOCK (aagg); - - switch (format) { - case GST_FORMAT_TIME: - gst_query_set_position (query, format, - gst_segment_to_stream_time (&agg->segment, GST_FORMAT_TIME, - agg->segment.position)); - res = TRUE; - break; - case GST_FORMAT_BYTES: - if (GST_AUDIO_INFO_BPF (&aagg->info)) { - gst_query_set_position (query, format, aagg->priv->offset * - GST_AUDIO_INFO_BPF (&aagg->info)); - res = TRUE; - } - break; - case GST_FORMAT_DEFAULT: - gst_query_set_position (query, format, aagg->priv->offset); - res = TRUE; - break; - default: - break; - } - - GST_OBJECT_UNLOCK (aagg); - - break; - } - default: - res = - GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_query - (agg, query); - break; - } - - return res; -} - - -void -gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg, - GstAudioAggregatorPad * pad, GstCaps * caps) -{ -#ifndef G_DISABLE_ASSERT - gboolean valid; - - GST_OBJECT_LOCK (pad); - valid = gst_audio_info_from_caps (&pad->info, caps); - g_assert (valid); - GST_OBJECT_UNLOCK (pad); -#else - GST_OBJECT_LOCK (pad); - (void) gst_audio_info_from_caps (&pad->info, caps); - GST_OBJECT_UNLOCK (pad); -#endif -} - -/* Must hold object lock and aagg lock to call */ - -static void -gst_audio_aggregator_reset (GstAudioAggregator * aagg) -{ - GstAggregator *agg = GST_AGGREGATOR (aagg); - - GST_AUDIO_AGGREGATOR_LOCK (aagg); - GST_OBJECT_LOCK (aagg); - agg->segment.position = -1; - aagg->priv->offset = -1; - gst_audio_info_init (&aagg->info); - gst_caps_replace (&aagg->current_caps, NULL); - gst_buffer_replace (&aagg->priv->current_buffer, NULL); - GST_OBJECT_UNLOCK (aagg); - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); -} - -static gboolean -gst_audio_aggregator_start (GstAggregator * agg) -{ - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); - - gst_audio_aggregator_reset (aagg); - - return TRUE; -} - -static gboolean -gst_audio_aggregator_stop (GstAggregator * agg) -{ - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); - - gst_audio_aggregator_reset (aagg); - - return TRUE; -} - -static GstFlowReturn -gst_audio_aggregator_flush (GstAggregator * agg) -{ - GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); - - GST_AUDIO_AGGREGATOR_LOCK (aagg); - GST_OBJECT_LOCK (aagg); - agg->segment.position = -1; - aagg->priv->offset = -1; - gst_buffer_replace (&aagg->priv->current_buffer, NULL); - GST_OBJECT_UNLOCK (aagg); - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - - return GST_FLOW_OK; -} - -static GstBuffer * -gst_audio_aggregator_do_clip (GstAggregator * agg, - GstAggregatorPad * bpad, GstBuffer * buffer) -{ - GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad); - gint rate, bpf; - - rate = GST_AUDIO_INFO_RATE (&pad->info); - bpf = GST_AUDIO_INFO_BPF (&pad->info); - - GST_OBJECT_LOCK (bpad); - buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf); - GST_OBJECT_UNLOCK (bpad); - - return buffer; -} - -/* Called with the object lock for both the element and pad held, - * as well as the aagg lock - * - * Replace the current buffer with input and update GstAudioAggregatorPadPrivate - * values. - */ -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; - gint rate, bpf; - - GstAggregator *agg = GST_AGGREGATOR (aagg); - GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad); - - if (aaggclass->convert_buffer) { - rate = GST_AUDIO_INFO_RATE (&aagg->info); - bpf = GST_AUDIO_INFO_BPF (&aagg->info); - } else { - rate = GST_AUDIO_INFO_RATE (&pad->info); - bpf = GST_AUDIO_INFO_BPF (&pad->info); - } - - pad->priv->position = 0; - pad->priv->size = gst_buffer_get_size (pad->priv->buffer) / bpf; - - if (pad->priv->size == 0) { - if (!GST_BUFFER_DURATION_IS_VALID (pad->priv->buffer) || - !GST_BUFFER_FLAG_IS_SET (pad->priv->buffer, GST_BUFFER_FLAG_GAP)) { - GST_WARNING_OBJECT (pad, "Dropping 0-sized buffer missing either a" - " duration or a GAP flag: %" GST_PTR_FORMAT, pad->priv->buffer); - return FALSE; - } - - pad->priv->size = - gst_util_uint64_scale (GST_BUFFER_DURATION (pad->priv->buffer), rate, - GST_SECOND); - } - - if (!GST_BUFFER_PTS_IS_VALID (pad->priv->buffer)) { - if (pad->priv->output_offset == -1) - pad->priv->output_offset = aagg->priv->offset; - if (pad->priv->next_offset == -1) - pad->priv->next_offset = pad->priv->size; - else - pad->priv->next_offset += pad->priv->size; - goto done; - } - - start_time = GST_BUFFER_PTS (pad->priv->buffer); - end_time = - start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND, - rate); - - /* Clipping should've ensured this */ - g_assert (start_time >= aggpad->segment.start); - - start_offset = - gst_util_uint64_scale (start_time - aggpad->segment.start, rate, - GST_SECOND); - end_offset = start_offset + pad->priv->size; - - if (GST_BUFFER_IS_DISCONT (pad->priv->buffer) - || GST_BUFFER_FLAG_IS_SET (pad->priv->buffer, GST_BUFFER_FLAG_RESYNC) - || pad->priv->new_segment || pad->priv->next_offset == -1) { - discont = TRUE; - pad->priv->new_segment = FALSE; - } else { - guint64 diff, max_sample_diff; - - /* Check discont, based on audiobasesink */ - if (start_offset <= pad->priv->next_offset) - diff = pad->priv->next_offset - start_offset; - else - diff = start_offset - pad->priv->next_offset; - - max_sample_diff = - gst_util_uint64_scale_int (aagg->priv->alignment_threshold, rate, - GST_SECOND); - - /* Discont! */ - if (G_UNLIKELY (diff >= max_sample_diff)) { - if (aagg->priv->discont_wait > 0) { - if (pad->priv->discont_time == GST_CLOCK_TIME_NONE) { - pad->priv->discont_time = start_time; - } else if (start_time - pad->priv->discont_time >= - aagg->priv->discont_wait) { - discont = TRUE; - pad->priv->discont_time = GST_CLOCK_TIME_NONE; - } - } else { - discont = TRUE; - } - } else if (G_UNLIKELY (pad->priv->discont_time != GST_CLOCK_TIME_NONE)) { - /* we have had a discont, but are now back on track! */ - pad->priv->discont_time = GST_CLOCK_TIME_NONE; - } - } - - if (discont) { - /* Have discont, need resync */ - if (pad->priv->next_offset != -1) - GST_DEBUG_OBJECT (pad, "Have discont. Expected %" - G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT, - pad->priv->next_offset, start_offset); - pad->priv->output_offset = -1; - pad->priv->next_offset = end_offset; - } else { - pad->priv->next_offset += pad->priv->size; - } - - if (pad->priv->output_offset == -1) { - GstClockTime start_running_time; - GstClockTime end_running_time; - GstClockTime segment_pos; - guint64 start_output_offset = -1; - guint64 end_output_offset = -1; - - start_running_time = - gst_segment_to_running_time (&aggpad->segment, - GST_FORMAT_TIME, start_time); - end_running_time = - gst_segment_to_running_time (&aggpad->segment, - GST_FORMAT_TIME, end_time); - - /* Convert to position in the output segment */ - segment_pos = - gst_segment_position_from_running_time (&agg->segment, GST_FORMAT_TIME, - start_running_time); - if (GST_CLOCK_TIME_IS_VALID (segment_pos)) - start_output_offset = - gst_util_uint64_scale (segment_pos - agg->segment.start, rate, - GST_SECOND); - - segment_pos = - gst_segment_position_from_running_time (&agg->segment, GST_FORMAT_TIME, - end_running_time); - if (GST_CLOCK_TIME_IS_VALID (segment_pos)) - end_output_offset = - gst_util_uint64_scale (segment_pos - agg->segment.start, rate, - GST_SECOND); - - if (start_output_offset == -1 && end_output_offset == -1) { - /* Outside output segment, drop */ - pad->priv->position = 0; - pad->priv->size = 0; - pad->priv->output_offset = -1; - GST_DEBUG_OBJECT (pad, "Buffer outside output segment"); - return FALSE; - } - - /* Calculate end_output_offset if it was outside the output segment */ - if (end_output_offset == -1) - end_output_offset = start_output_offset + pad->priv->size; - - if (end_output_offset < aagg->priv->offset) { - pad->priv->position = 0; - pad->priv->size = 0; - pad->priv->output_offset = -1; - GST_DEBUG_OBJECT (pad, - "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %" - G_GINT64_FORMAT, end_output_offset, aagg->priv->offset); - return FALSE; - } - - if (start_output_offset == -1 || start_output_offset < aagg->priv->offset) { - guint diff; - - if (start_output_offset == -1 && end_output_offset < pad->priv->size) { - diff = pad->priv->size - end_output_offset + aagg->priv->offset; - } else if (start_output_offset == -1) { - start_output_offset = end_output_offset - pad->priv->size; - - if (start_output_offset < aagg->priv->offset) - diff = aagg->priv->offset - start_output_offset; - else - diff = 0; - } else { - diff = aagg->priv->offset - start_output_offset; - } - - pad->priv->position += diff; - if (pad->priv->position >= pad->priv->size) { - /* Empty buffer, drop */ - pad->priv->position = 0; - pad->priv->size = 0; - pad->priv->output_offset = -1; - GST_DEBUG_OBJECT (pad, - "Buffer before segment or current position: %" G_GUINT64_FORMAT - " < %" G_GINT64_FORMAT, end_output_offset, aagg->priv->offset); - return FALSE; - } - } - - if (start_output_offset == -1 || start_output_offset < aagg->priv->offset) - pad->priv->output_offset = aagg->priv->offset; - else - pad->priv->output_offset = start_output_offset; - - GST_DEBUG_OBJECT (pad, - "Buffer resynced: Pad offset %" G_GUINT64_FORMAT - ", current audio aggregator offset %" G_GINT64_FORMAT, - pad->priv->output_offset, aagg->priv->offset); - } - -done: - - GST_LOG_OBJECT (pad, - "Queued new buffer at offset %" G_GUINT64_FORMAT, - pad->priv->output_offset); - - return TRUE; -} - -/* Called with pad object lock held */ - -static gboolean -gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg, - GstAudioAggregatorPad * pad, GstBuffer * inbuf, GstBuffer * outbuf, - guint blocksize) -{ - guint overlap; - guint out_start; - gboolean filled; - guint in_offset; - gboolean pad_changed = FALSE; - - /* Overlap => mix */ - if (aagg->priv->offset < pad->priv->output_offset) - out_start = pad->priv->output_offset - aagg->priv->offset; - else - out_start = 0; - - overlap = pad->priv->size - pad->priv->position; - if (overlap > blocksize - out_start) - overlap = blocksize - out_start; - - if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) { - /* skip gap buffer */ - GST_LOG_OBJECT (pad, "skipping GAP buffer"); - pad->priv->output_offset += pad->priv->size - pad->priv->position; - pad->priv->position = pad->priv->size; - - gst_buffer_replace (&pad->priv->buffer, NULL); - gst_buffer_replace (&pad->priv->input_buffer, NULL); - return FALSE; - } - - gst_buffer_ref (inbuf); - in_offset = pad->priv->position; - GST_OBJECT_UNLOCK (pad); - GST_OBJECT_UNLOCK (aagg); - - filled = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->aggregate_one_buffer (aagg, - pad, inbuf, in_offset, outbuf, out_start, overlap); - - GST_OBJECT_LOCK (aagg); - GST_OBJECT_LOCK (pad); - - pad_changed = (inbuf != pad->priv->buffer); - gst_buffer_unref (inbuf); - - if (filled) - GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_GAP); - - if (pad_changed) - return FALSE; - - pad->priv->position += overlap; - pad->priv->output_offset += overlap; - - if (pad->priv->position == pad->priv->size) { - /* Buffer done, drop it */ - gst_buffer_replace (&pad->priv->buffer, NULL); - gst_buffer_replace (&pad->priv->input_buffer, NULL); - GST_LOG_OBJECT (pad, "Finished mixing buffer, waiting for next"); - return FALSE; - } - - return TRUE; -} - -static GstBuffer * -gst_audio_aggregator_create_output_buffer (GstAudioAggregator * aagg, - guint num_frames) -{ - GstAllocator *allocator; - GstAllocationParams params; - GstBuffer *outbuf; - GstMapInfo outmap; - - gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, ¶ms); - - GST_DEBUG ("Creating output buffer with size %d", - num_frames * GST_AUDIO_INFO_BPF (&aagg->info)); - - outbuf = gst_buffer_new_allocate (allocator, num_frames * - GST_AUDIO_INFO_BPF (&aagg->info), ¶ms); - - if (allocator) - gst_object_unref (allocator); - - gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE); - gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size); - gst_buffer_unmap (outbuf, &outmap); - - return outbuf; -} - -static gboolean -sync_pad_values (GstElement * aagg, GstPad * pad, gpointer user_data) -{ - GstAudioAggregatorPad *aapad = GST_AUDIO_AGGREGATOR_PAD (pad); - GstAggregatorPad *bpad = GST_AGGREGATOR_PAD_CAST (pad); - GstClockTime timestamp, stream_time; - - if (aapad->priv->buffer == NULL) - return TRUE; - - timestamp = GST_BUFFER_PTS (aapad->priv->buffer); - GST_OBJECT_LOCK (bpad); - stream_time = gst_segment_to_stream_time (&bpad->segment, GST_FORMAT_TIME, - timestamp); - GST_OBJECT_UNLOCK (bpad); - - /* sync object properties on stream time */ - /* TODO: Ideally we would want to do that on every sample */ - if (GST_CLOCK_TIME_IS_VALID (stream_time)) - gst_object_sync_values (GST_OBJECT_CAST (pad), stream_time); - - return TRUE; -} - -static GstFlowReturn -gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout) -{ - /* Calculate the current output offset/timestamp and offset_end/timestamp_end. - * Allocate a silence buffer for this and store it. - * - * For all pads: - * 1) Once per input buffer (cached) - * 1) Check discont (flag and timestamp with tolerance) - * 2) If discont or new, resync. That means: - * 1) Drop all start data of the buffer that comes before - * the current position/offset. - * 2) Calculate the offset (output segment!) that the first - * frame of the input buffer corresponds to. Base this on - * the running time. - * - * 2) If the current pad's offset/offset_end overlaps with the output - * offset/offset_end, mix it at the appropiate position in the output - * buffer and advance the pad's position. Remember if this pad needs - * a new buffer to advance behind the output offset_end. - * - * If we had no pad with a buffer, go EOS. - * - * If we had at least one pad that did not advance behind output - * offset_end, let aggregate be called again for the current - * output offset/offset_end. - */ - GstElement *element; - GstAudioAggregator *aagg; - GList *iter; - GstFlowReturn ret; - GstBuffer *outbuf = NULL; - gint64 next_offset; - gint64 next_timestamp; - gint rate, bpf; - gboolean dropped = FALSE; - gboolean is_eos = TRUE; - gboolean is_done = TRUE; - guint blocksize; - - element = GST_ELEMENT (agg); - aagg = GST_AUDIO_AGGREGATOR (agg); - - /* Sync pad properties to the stream time */ - gst_element_foreach_sink_pad (element, sync_pad_values, NULL); - - GST_AUDIO_AGGREGATOR_LOCK (aagg); - GST_OBJECT_LOCK (agg); - - /* Update position from the segment start/stop if needed */ - if (agg->segment.position == -1) { - if (agg->segment.rate > 0.0) - agg->segment.position = agg->segment.start; - else - agg->segment.position = agg->segment.stop; - } - - if (G_UNLIKELY (aagg->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) { - if (timeout) { - GST_DEBUG_OBJECT (aagg, - "Got timeout before receiving any caps, don't output anything"); - - /* Advance position */ - if (agg->segment.rate > 0.0) - agg->segment.position += aagg->priv->output_buffer_duration; - else if (agg->segment.position > aagg->priv->output_buffer_duration) - agg->segment.position -= aagg->priv->output_buffer_duration; - else - agg->segment.position = 0; - - GST_OBJECT_UNLOCK (agg); - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - return GST_AGGREGATOR_FLOW_NEED_DATA; - } else { - GST_OBJECT_UNLOCK (agg); - goto not_negotiated; - } - } - - rate = GST_AUDIO_INFO_RATE (&aagg->info); - bpf = GST_AUDIO_INFO_BPF (&aagg->info); - - if (aagg->priv->offset == -1) { - aagg->priv->offset = - gst_util_uint64_scale (agg->segment.position - agg->segment.start, rate, - GST_SECOND); - GST_DEBUG_OBJECT (aagg, "Starting at offset %" G_GINT64_FORMAT, - aagg->priv->offset); - } - - blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration, - rate, GST_SECOND); - blocksize = MAX (1, blocksize); - - /* FIXME: Reverse mixing does not work at all yet */ - if (agg->segment.rate > 0.0) { - next_offset = aagg->priv->offset + blocksize; - } else { - next_offset = aagg->priv->offset - blocksize; - } - - /* Use the sample counter, which will never accumulate rounding errors */ - next_timestamp = - agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND, - rate); - - if (aagg->priv->current_buffer == NULL) { - GST_OBJECT_UNLOCK (agg); - aagg->priv->current_buffer = - GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg, - blocksize); - /* Be careful, some things could have changed ? */ - GST_OBJECT_LOCK (agg); - GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP); - } - outbuf = aagg->priv->current_buffer; - - GST_LOG_OBJECT (agg, - "Starting to mix %u samples for offset %" G_GINT64_FORMAT - " with timestamp %" GST_TIME_FORMAT, blocksize, - aagg->priv->offset, GST_TIME_ARGS (agg->segment.position)); - - for (iter = element->sinkpads; iter; iter = iter->next) { - GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data; - GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data; - gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad); - - if (!pad_eos) - is_eos = FALSE; - - pad->priv->input_buffer = gst_aggregator_pad_peek_buffer (aggpad); - - GST_OBJECT_LOCK (pad); - if (!pad->priv->input_buffer) { - if (timeout) { - if (pad->priv->output_offset < next_offset) { - gint64 diff = next_offset - pad->priv->output_offset; - GST_DEBUG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT - " frames (%" GST_TIME_FORMAT ")", diff, - GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND, - GST_AUDIO_INFO_RATE (&aagg->info)))); - } - } else if (!pad_eos) { - is_done = FALSE; - } - GST_OBJECT_UNLOCK (pad); - continue; - } - - /* New buffer? */ - if (!pad->priv->buffer) { - if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (pad)) - pad->priv->buffer = - gst_audio_aggregator_convert_buffer - (aagg, GST_PAD (pad), &pad->info, &aagg->info, - pad->priv->input_buffer); - else - pad->priv->buffer = gst_buffer_ref (pad->priv->input_buffer); - - if (!gst_audio_aggregator_fill_buffer (aagg, pad)) { - gst_buffer_replace (&pad->priv->buffer, NULL); - gst_buffer_replace (&pad->priv->input_buffer, NULL); - pad->priv->buffer = NULL; - dropped = TRUE; - GST_OBJECT_UNLOCK (pad); - - gst_aggregator_pad_drop_buffer (aggpad); - continue; - } - } else { - gst_buffer_unref (pad->priv->input_buffer); - } - - if (!pad->priv->buffer && !dropped && pad_eos) { - GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state"); - GST_OBJECT_UNLOCK (pad); - continue; - } - - g_assert (pad->priv->buffer); - - /* This pad is lagging behind, we need to update the offset - * and maybe drop the current buffer */ - if (pad->priv->output_offset < aagg->priv->offset) { - gint64 diff = aagg->priv->offset - pad->priv->output_offset; - gint64 odiff = diff; - - if (pad->priv->position + diff > pad->priv->size) - diff = pad->priv->size - pad->priv->position; - pad->priv->position += diff; - pad->priv->output_offset += diff; - - if (pad->priv->position == pad->priv->size) { - GST_DEBUG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT - ", dropping %" GST_PTR_FORMAT, - GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND, - GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer); - /* Buffer done, drop it */ - gst_buffer_replace (&pad->priv->buffer, NULL); - gst_buffer_replace (&pad->priv->input_buffer, NULL); - dropped = TRUE; - GST_OBJECT_UNLOCK (pad); - gst_aggregator_pad_drop_buffer (aggpad); - continue; - } - } - - g_assert (pad->priv->buffer); - - if (pad->priv->output_offset >= aagg->priv->offset - && pad->priv->output_offset < aagg->priv->offset + blocksize) { - gboolean drop_buf; - - GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset"); - drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer, - outbuf, blocksize); - if (pad->priv->output_offset >= next_offset) { - GST_LOG_OBJECT (pad, - "Pad is at or after current offset: %" G_GUINT64_FORMAT " >= %" - G_GINT64_FORMAT, pad->priv->output_offset, next_offset); - } else { - is_done = FALSE; - } - if (drop_buf) { - GST_OBJECT_UNLOCK (pad); - gst_aggregator_pad_drop_buffer (aggpad); - continue; - } - } - - GST_OBJECT_UNLOCK (pad); - } - GST_OBJECT_UNLOCK (agg); - - if (dropped) { - /* We dropped a buffer, retry */ - GST_LOG_OBJECT (aagg, "A pad dropped a buffer, wait for the next one"); - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - return GST_AGGREGATOR_FLOW_NEED_DATA; - } - - if (!is_done && !is_eos) { - /* Get more buffers */ - GST_LOG_OBJECT (aagg, - "We're not done yet for the current offset, waiting for more data"); - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - return GST_AGGREGATOR_FLOW_NEED_DATA; - } - - if (is_eos) { - gint64 max_offset = 0; - - GST_DEBUG_OBJECT (aagg, "We're EOS"); - - GST_OBJECT_LOCK (agg); - for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) { - GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data); - - max_offset = MAX ((gint64) max_offset, (gint64) pad->priv->output_offset); - } - GST_OBJECT_UNLOCK (agg); - - /* This means EOS or nothing mixed in at all */ - if (aagg->priv->offset == max_offset) { - gst_buffer_replace (&aagg->priv->current_buffer, NULL); - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - return GST_FLOW_EOS; - } - - if (max_offset <= next_offset) { - GST_DEBUG_OBJECT (aagg, - "Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %" - G_GINT64_FORMAT, max_offset, next_offset); - next_offset = max_offset; - next_timestamp = - agg->segment.start + gst_util_uint64_scale (next_offset, GST_SECOND, - rate); - - if (next_offset > aagg->priv->offset) - gst_buffer_resize (outbuf, 0, (next_offset - aagg->priv->offset) * bpf); - } - } - - /* set timestamps on the output buffer */ - GST_OBJECT_LOCK (agg); - if (agg->segment.rate > 0.0) { - GST_BUFFER_PTS (outbuf) = agg->segment.position; - GST_BUFFER_OFFSET (outbuf) = aagg->priv->offset; - GST_BUFFER_OFFSET_END (outbuf) = next_offset; - GST_BUFFER_DURATION (outbuf) = next_timestamp - agg->segment.position; - } else { - GST_BUFFER_PTS (outbuf) = next_timestamp; - GST_BUFFER_OFFSET (outbuf) = next_offset; - GST_BUFFER_OFFSET_END (outbuf) = aagg->priv->offset; - GST_BUFFER_DURATION (outbuf) = agg->segment.position - next_timestamp; - } - - GST_OBJECT_UNLOCK (agg); - - /* send it out */ - GST_LOG_OBJECT (aagg, - "pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %" - G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)), - GST_BUFFER_OFFSET (outbuf)); - - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - - ret = gst_aggregator_finish_buffer (agg, outbuf); - aagg->priv->current_buffer = NULL; - - GST_LOG_OBJECT (aagg, "pushed outbuf, result = %s", gst_flow_get_name (ret)); - - GST_AUDIO_AGGREGATOR_LOCK (aagg); - GST_OBJECT_LOCK (agg); - aagg->priv->offset = next_offset; - agg->segment.position = next_timestamp; - - /* If there was a timeout and there was a gap in data in out of the streams, - * then it's a very good time to for a resync with the timestamps. - */ - if (timeout) { - for (iter = element->sinkpads; iter; iter = iter->next) { - GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (iter->data); - - GST_OBJECT_LOCK (pad); - if (pad->priv->output_offset < aagg->priv->offset) - pad->priv->output_offset = -1; - GST_OBJECT_UNLOCK (pad); - } - } - GST_OBJECT_UNLOCK (agg); - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - - return ret; - /* ERRORS */ -not_negotiated: - { - GST_AUDIO_AGGREGATOR_UNLOCK (aagg); - GST_ELEMENT_ERROR (aagg, STREAM, FORMAT, (NULL), - ("Unknown data received, not negotiated")); - return GST_FLOW_NOT_NEGOTIATED; - } -} diff --git a/gst-libs/gst/audio/gstaudioaggregator.h b/gst-libs/gst/audio/gstaudioaggregator.h deleted file mode 100644 index b32630ee6..000000000 --- a/gst-libs/gst/audio/gstaudioaggregator.h +++ /dev/null @@ -1,228 +0,0 @@ -/* GStreamer - * Copyright (C) 2014 Collabora - * Author: Olivier Crete - * - * gstaudioaggregator.h: - * - * 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_AGGREGATOR_H__ -#define __GST_AUDIO_AGGREGATOR_H__ - -#ifndef GST_USE_UNSTABLE_API -#warning "The Base library from gst-plugins-bad is unstable API and may change in future." -#warning "You can define GST_USE_UNSTABLE_API to avoid this warning." -#endif - -#include -#include -#include - -G_BEGIN_DECLS - -/******************************* - * GstAudioAggregator Structs * - *******************************/ - -typedef struct _GstAudioAggregator GstAudioAggregator; -typedef struct _GstAudioAggregatorPrivate GstAudioAggregatorPrivate; -typedef struct _GstAudioAggregatorClass GstAudioAggregatorClass; - - -/************************ - * GstAudioAggregatorPad API * - ***********************/ - -#define GST_TYPE_AUDIO_AGGREGATOR_PAD (gst_audio_aggregator_pad_get_type()) -#define GST_AUDIO_AGGREGATOR_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPad)) -#define GST_AUDIO_AGGREGATOR_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadClass)) -#define GST_AUDIO_AGGREGATOR_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadClass)) -#define GST_IS_AUDIO_AGGREGATOR_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD)) -#define GST_IS_AUDIO_AGGREGATOR_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR_PAD)) - -/**************************** - * GstAudioAggregatorPad Structs * - ***************************/ - -typedef struct _GstAudioAggregatorPad GstAudioAggregatorPad; -typedef struct _GstAudioAggregatorPadClass GstAudioAggregatorPadClass; -typedef struct _GstAudioAggregatorPadPrivate GstAudioAggregatorPadPrivate; - -/** - * GstAudioAggregatorPad: - * @parent: The parent #GstAggregatorPad - * @info: The audio info for this pad set from the incoming caps - * - * The default implementation of GstPad used with #GstAudioAggregator - */ -struct _GstAudioAggregatorPad -{ - GstAggregatorPad parent; - - GstAudioInfo info; - - /*< private >*/ - GstAudioAggregatorPadPrivate * priv; - - gpointer _gst_reserved[GST_PADDING]; -}; - -/** - * GstAudioAggregatorPadClass: - * - */ -struct _GstAudioAggregatorPadClass - { - GstAggregatorPadClass parent_class; - - /*< private >*/ - gpointer _gst_reserved[GST_PADDING_LARGE]; -}; - -GST_EXPORT -GType gst_audio_aggregator_pad_get_type (void); - -#define GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD (gst_audio_aggregator_convert_pad_get_type()) -#define GST_AUDIO_AGGREGATOR_CONVERT_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD, GstAudioAggregatorConvertPad)) -#define GST_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD, GstAudioAggregatorConvertPadClass)) -#define GST_AUDIO_AGGREGATOR_CONVERT_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD, GstAudioAggregatorConvertPadClass)) -#define GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD)) -#define GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR_CONVERT_PAD)) - -/**************************** - * GstAudioAggregatorPad Structs * - ***************************/ - -typedef struct _GstAudioAggregatorConvertPad GstAudioAggregatorConvertPad; -typedef struct _GstAudioAggregatorConvertPadClass GstAudioAggregatorConvertPadClass; -typedef struct _GstAudioAggregatorConvertPadPrivate GstAudioAggregatorConvertPadPrivate; - -/** - * GstAudioAggregatorConvertPad: - * @parent: The parent #GstAudioAggregatorPad - * - * An implementation of GstPad that can be used with #GstAudioAggregator. - * - * See #GstAudioAggregator for more details. - */ -struct _GstAudioAggregatorConvertPad -{ - GstAudioAggregatorPad parent; - - /*< private >*/ - GstAudioAggregatorConvertPadPrivate * priv; - - gpointer _gst_reserved[GST_PADDING]; -}; - -/** - * GstAudioAggregatorConvertPadClass: - * - */ -struct _GstAudioAggregatorConvertPadClass -{ - GstAudioAggregatorPadClass parent_class; - - /*< private >*/ - gpointer _gst_reserved[GST_PADDING]; -}; - -GST_EXPORT -GType gst_audio_aggregator_convert_pad_get_type (void); - -/************************** - * GstAudioAggregator API * - **************************/ - -#define GST_TYPE_AUDIO_AGGREGATOR (gst_audio_aggregator_get_type()) -#define GST_AUDIO_AGGREGATOR(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregator)) -#define GST_AUDIO_AGGREGATOR_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregatorClass)) -#define GST_AUDIO_AGGREGATOR_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregatorClass)) -#define GST_IS_AUDIO_AGGREGATOR(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR)) -#define GST_IS_AUDIO_AGGREGATOR_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR)) - -/** - * GstAudioAggregator: - * @parent: The parent #GstAggregator - * @info: The information parsed from the current caps - * @current_caps: The caps set by the subclass - * - * GstAudioAggregator object - */ -struct _GstAudioAggregator -{ - GstAggregator parent; - - /* All member are read only for subclasses, must hold OBJECT lock */ - GstAudioInfo info; - - GstCaps *current_caps; - - /*< private >*/ - GstAudioAggregatorPrivate *priv; - - gpointer _gst_reserved[GST_PADDING]; -}; - -/** - * GstAudioAggregatorClass: - * @create_output_buffer: Create a new output buffer contains num_frames frames. - * @aggregate_one_buffer: Aggregates one input buffer to the output - * 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; - - GstBuffer * (* create_output_buffer) (GstAudioAggregator * aagg, - guint num_frames); - 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]; -}; - -/************************* - * GstAggregator methods * - ************************/ - -GST_EXPORT -GType gst_audio_aggregator_get_type(void); - -GST_EXPORT -void gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg, - GstAudioAggregatorPad * pad, - GstCaps * caps); - -GST_EXPORT -void gst_audio_aggregator_class_perform_conversion (GstAudioAggregatorClass * klass); - -G_END_DECLS - -#endif /* __GST_AUDIO_AGGREGATOR_H__ */ diff --git a/gst-libs/gst/audio/meson.build b/gst-libs/gst/audio/meson.build index ac4871903..e32bdf604 100644 --- a/gst-libs/gst/audio/meson.build +++ b/gst-libs/gst/audio/meson.build @@ -1,5 +1,5 @@ -badaudio_sources = ['gstaudioaggregator.c', 'gstnonstreamaudiodecoder.c'] -badaudio_headers = ['gstaudioaggregator.h', 'gstnonstreamaudiodecoder.h'] +badaudio_sources = ['gstnonstreamaudiodecoder.c'] +badaudio_headers = ['gstnonstreamaudiodecoder.h'] install_headers(badaudio_headers, subdir : 'gstreamer-1.0/gst/audio') -- cgit v1.2.1