+++ /dev/null
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- * 2001 Thomas <thomas@apestaart.org>
- * 2005,2006 Wim Taymans <wim@fluendo.com>
- * 2013 Sebastian Dröge <sebastian@centricular.com>
- * 2014 Collabora
- * Olivier Crete <olivier.crete@collabora.com>
- *
- * 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 <string.h>
-
-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;
- }
-}
projects / platform / upstream / gstreamer.git / commitdiff