mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-20 23:36:38 +00:00
9243418a23
Previously one of the branches did not check for the property value. To avoid this in the future, check inside the QoS calculation function instead. As a side effect this now always prints the debug messages into the logs when samples are dropped, which is useful information even without the QoS messages. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1224>
2519 lines
80 KiB
C
2519 lines
80 KiB
C
/* GStreamer
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* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
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* 2001 Thomas <thomas@apestaart.org>
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* 2005,2006 Wim Taymans <wim@fluendo.com>
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* 2013 Sebastian Dröge <sebastian@centricular.com>
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* 2014 Collabora
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* Olivier Crete <olivier.crete@collabora.com>
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*
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* gstaudioaggregator.c:
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION: gstaudioaggregator
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* @title: GstAudioAggregator
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* @short_description: Base class that manages a set of audio input pads
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* with the purpose of aggregating or mixing their raw audio input buffers
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* @see_also: #GstAggregator, #GstAudioMixer
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*
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* Subclasses must use (a subclass of) #GstAudioAggregatorPad for both
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* their source and sink pads,
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* gst_element_class_add_static_pad_template_with_gtype() is a convenient
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* helper.
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*
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* #GstAudioAggregator can perform conversion on the data arriving
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* on its sink pads, based on the format expected downstream: in order
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* to enable that behaviour, the GType of the sink pads must either be
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* a (subclass of) #GstAudioAggregatorConvertPad to use the default
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* #GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
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* implementing #GstAudioAggregatorPadClass.convert_buffer.
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*
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* To allow for the output caps to change, the mechanism is the same as
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* above, with the GType of the source pad.
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*
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* See #GstAudioMixer for an example.
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*
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* When conversion is enabled, #GstAudioAggregator will accept
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* any type of raw audio caps and perform conversion
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* on the data arriving on its sink pads, with whatever downstream
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* expects as the target format.
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*
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* In case downstream caps are not fully fixated, it will use
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* the first configured sink pad to finish fixating its source pad
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* caps.
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*
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* A notable exception for now is the sample rate, sink pads must
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* have the same sample rate as either the downstream requirement,
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* or the first configured pad, or a combination of both (when
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* downstream specifies a range or a set of acceptable rates).
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*
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* The #GstAggregator::samples-selected signal is provided with some
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* additional information about the output buffer:
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* - "offset" G_TYPE_UINT64 Offset in samples since segment start
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* for the position that is next to be filled in the output buffer.
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* - "frames" G_TYPE_UINT Number of frames per output buffer.
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*
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* In addition the gst_aggregator_peek_next_sample() function returns
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* additional information in the info #GstStructure of the returned sample:
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* - "output-offset" G_TYPE_UINT64 Sample offset in output segment relative to
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* the output segment's start where the current position of this input
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* buffer would be placed
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* - "position" G_TYPE_UINT current position in the input buffer in samples
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* - "size" G_TYPE_UINT size of the input buffer in samples
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "gstaudioaggregator.h"
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#include <string.h>
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GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug);
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#define GST_CAT_DEFAULT audio_aggregator_debug
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enum
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{
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PROP_PAD_0,
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PROP_PAD_QOS_MESSAGES,
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};
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struct _GstAudioAggregatorPadPrivate
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{
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/* All members are protected by the pad object lock */
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GstBuffer *buffer; /* current buffer we're mixing, for
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comparison with a new input buffer from
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aggregator to see if we need to update our
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cached values. */
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guint position, size; /* position in the input buffer and size of the
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input buffer in number of samples */
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guint64 output_offset; /* Sample offset in output segment relative to
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srcpad.segment.start where the current position
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of this input_buffer would be placed. */
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guint64 next_offset; /* Next expected sample offset relative to
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pad.segment.start. This is -1 when resyncing is
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needed, e.g. because of a previous discont. */
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/* Last time we noticed a discont */
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GstClockTime discont_time;
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/* A new unhandled segment event has been received */
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gboolean new_segment;
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guint64 processed; /* Number of samples processed since the element came out of READY */
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guint64 dropped; /* Number of sampels dropped since the element came out of READY */
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gboolean qos_messages; /* Property to decide to send QoS messages or not */
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};
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/*****************************************
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* GstAudioAggregatorPad implementation *
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*****************************************/
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G_DEFINE_TYPE_WITH_PRIVATE (GstAudioAggregatorPad, gst_audio_aggregator_pad,
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GST_TYPE_AGGREGATOR_PAD);
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static GstFlowReturn
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gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
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GstAggregator * aggregator);
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static void
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gst_audio_aggregator_pad_finalize (GObject * object)
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{
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GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) object;
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gst_buffer_replace (&pad->priv->buffer, NULL);
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G_OBJECT_CLASS (gst_audio_aggregator_pad_parent_class)->finalize (object);
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}
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static void
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gst_audio_aggregator_pad_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (object);
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switch (prop_id) {
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case PROP_PAD_QOS_MESSAGES:
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GST_OBJECT_LOCK (pad);
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g_value_set_boolean (value, pad->priv->qos_messages);
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GST_OBJECT_UNLOCK (pad);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_aggregator_pad_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (object);
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switch (prop_id) {
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case PROP_PAD_QOS_MESSAGES:
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GST_OBJECT_LOCK (pad);
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pad->priv->qos_messages = g_value_get_boolean (value);
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GST_OBJECT_UNLOCK (pad);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_aggregator_pad_class_init (GstAudioAggregatorPadClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass;
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gobject_class->set_property = gst_audio_aggregator_pad_set_property;
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gobject_class->get_property = gst_audio_aggregator_pad_get_property;
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gobject_class->finalize = gst_audio_aggregator_pad_finalize;
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aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audio_aggregator_pad_flush_pad);
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/**
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* GstAudioAggregatorPad:qos-messages:
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*
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* Emit QoS messages when dropping buffers.
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*
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* Since: 1.20
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*/
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g_object_class_install_property (gobject_class,
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PROP_PAD_QOS_MESSAGES, g_param_spec_boolean ("qos-messages",
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"Quality of Service Messages",
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"Emit QoS messages when dropping buffers", FALSE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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}
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static void
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gst_audio_aggregator_pad_init (GstAudioAggregatorPad * pad)
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{
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pad->priv = gst_audio_aggregator_pad_get_instance_private (pad);
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gst_audio_info_init (&pad->info);
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pad->priv->buffer = NULL;
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pad->priv->position = 0;
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pad->priv->size = 0;
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pad->priv->output_offset = -1;
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pad->priv->next_offset = -1;
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pad->priv->discont_time = GST_CLOCK_TIME_NONE;
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}
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/* Must be called from srcpad thread or when it is stopped */
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static void
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gst_audio_aggregator_pad_reset_qos (GstAudioAggregatorPad * pad)
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{
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pad->priv->dropped = 0;
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pad->priv->processed = 0;
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}
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static GstFlowReturn
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gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad,
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GstAggregator * aggregator)
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{
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GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
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GST_OBJECT_LOCK (aggpad);
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pad->priv->position = pad->priv->size = 0;
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pad->priv->output_offset = pad->priv->next_offset = -1;
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pad->priv->discont_time = GST_CLOCK_TIME_NONE;
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gst_buffer_replace (&pad->priv->buffer, NULL);
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gst_audio_aggregator_pad_reset_qos (pad);
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GST_OBJECT_UNLOCK (aggpad);
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return GST_FLOW_OK;
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}
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enum
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{
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PROP_CONVERT_PAD_0,
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PROP_CONVERT_PAD_CONVERTER_CONFIG
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};
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struct _GstAudioAggregatorConvertPadPrivate
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{
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/* All members are protected by the pad object lock */
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GstAudioConverter *converter;
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GstStructure *converter_config;
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gboolean converter_config_changed;
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};
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G_DEFINE_TYPE_WITH_PRIVATE (GstAudioAggregatorConvertPad,
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gst_audio_aggregator_convert_pad, GST_TYPE_AUDIO_AGGREGATOR_PAD);
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static void
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gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad
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* aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info)
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{
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GstStructure *config = aaggcpad->priv->converter_config;
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GstAudioConverter *converter;
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if (!aaggcpad->priv->converter_config_changed)
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return;
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g_clear_pointer (&aaggcpad->priv->converter, gst_audio_converter_free);
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aaggcpad->priv->converter_config_changed = FALSE;
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if (in_info->finfo->format == GST_AUDIO_FORMAT_UNKNOWN) {
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/* If we haven't received caps yet, this pad should not have
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* a buffer to convert anyway */
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return;
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}
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converter =
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gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE, in_info, out_info,
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config ? gst_structure_copy (config) : NULL);
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if (converter == NULL) {
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/* FIXME: Not converting when we need to but the config is invalid (e.g.
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* because the mix-matrix is not the right size) produces garbage. An
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* invalid config should cause a GST_FLOW_NOT_NEGOTIATED. */
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return;
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}
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if (!gst_audio_converter_is_passthrough (converter))
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aaggcpad->priv->converter = converter;
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else
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gst_audio_converter_free (converter);
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}
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static void
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gst_audio_aggregator_pad_update_conversion_info (GstAudioAggregatorPad *
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aaggpad)
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{
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GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->priv->converter_config_changed =
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TRUE;
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}
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static GstBuffer *
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gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorPad *
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aaggpad, GstAudioInfo * in_info, GstAudioInfo * out_info,
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GstBuffer * input_buffer)
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{
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GstBuffer *res;
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GstAudioAggregatorConvertPad *aaggcpad =
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GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad);
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gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info,
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out_info);
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if (aaggcpad->priv->converter) {
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gint insize = gst_buffer_get_size (input_buffer);
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gsize insamples = insize / in_info->bpf;
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gsize outsamples =
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gst_audio_converter_get_out_frames (aaggcpad->priv->converter,
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insamples);
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gint outsize = outsamples * out_info->bpf;
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GstMapInfo inmap, outmap;
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res = gst_buffer_new_allocate (NULL, outsize, NULL);
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/* We create a perfectly similar buffer, except obviously for
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* its converted contents */
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gst_buffer_copy_into (res, input_buffer,
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GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS |
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GST_BUFFER_COPY_META, 0, -1);
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gst_buffer_map (input_buffer, &inmap, GST_MAP_READ);
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gst_buffer_map (res, &outmap, GST_MAP_WRITE);
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gst_audio_converter_samples (aaggcpad->priv->converter,
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GST_AUDIO_CONVERTER_FLAG_NONE,
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(gpointer *) & inmap.data, insamples,
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(gpointer *) & outmap.data, outsamples);
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gst_buffer_unmap (input_buffer, &inmap);
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gst_buffer_unmap (res, &outmap);
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} else {
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res = gst_buffer_ref (input_buffer);
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}
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return res;
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}
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static void
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gst_audio_aggregator_convert_pad_finalize (GObject * object)
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{
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GstAudioAggregatorConvertPad *pad = (GstAudioAggregatorConvertPad *) object;
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if (pad->priv->converter)
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gst_audio_converter_free (pad->priv->converter);
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if (pad->priv->converter_config)
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gst_structure_free (pad->priv->converter_config);
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G_OBJECT_CLASS (gst_audio_aggregator_convert_pad_parent_class)->finalize
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(object);
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}
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static void
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gst_audio_aggregator_convert_pad_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object);
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switch (prop_id) {
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case PROP_CONVERT_PAD_CONVERTER_CONFIG:
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GST_OBJECT_LOCK (pad);
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if (pad->priv->converter_config)
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g_value_set_boxed (value, pad->priv->converter_config);
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GST_OBJECT_UNLOCK (pad);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_aggregator_convert_pad_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object);
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switch (prop_id) {
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case PROP_CONVERT_PAD_CONVERTER_CONFIG:
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GST_OBJECT_LOCK (pad);
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if (pad->priv->converter_config)
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gst_structure_free (pad->priv->converter_config);
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pad->priv->converter_config = g_value_dup_boxed (value);
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pad->priv->converter_config_changed = TRUE;
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GST_OBJECT_UNLOCK (pad);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
|
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|
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static void
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gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass *
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klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstAudioAggregatorPadClass *aaggpad_class =
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(GstAudioAggregatorPadClass *) klass;
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gobject_class->set_property = gst_audio_aggregator_convert_pad_set_property;
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gobject_class->get_property = gst_audio_aggregator_convert_pad_get_property;
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g_object_class_install_property (gobject_class,
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PROP_CONVERT_PAD_CONVERTER_CONFIG,
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g_param_spec_boxed ("converter-config", "Converter configuration",
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"A GstStructure describing the configuration that should be used "
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"when converting this pad's audio buffers",
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GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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aaggpad_class->convert_buffer =
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gst_audio_aggregator_convert_pad_convert_buffer;
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aaggpad_class->update_conversion_info =
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gst_audio_aggregator_pad_update_conversion_info;
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gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize;
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}
|
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|
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static void
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gst_audio_aggregator_convert_pad_init (GstAudioAggregatorConvertPad * pad)
|
|
{
|
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pad->priv = gst_audio_aggregator_convert_pad_get_instance_private (pad);
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}
|
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|
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/**************************************
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* GstAudioAggregator implementation *
|
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**************************************/
|
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|
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struct _GstAudioAggregatorPrivate
|
|
{
|
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GMutex mutex;
|
|
|
|
/* All three properties are unprotected, can't be modified while streaming */
|
|
/* Size in frames that is output per buffer */
|
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GstClockTime alignment_threshold;
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GstClockTime discont_wait;
|
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|
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gint output_buffer_duration_n;
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gint output_buffer_duration_d;
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|
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guint samples_per_buffer;
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guint error_per_buffer;
|
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guint accumulated_error;
|
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guint current_blocksize;
|
|
|
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/* Protected by srcpad stream clock */
|
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/* Output buffer starting at offset containing blocksize frames (calculated
|
|
* from output_buffer_duration) */
|
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GstBuffer *current_buffer;
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|
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/* counters to keep track of timestamps */
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|
/* Readable with object lock, writable with both aag lock and object lock */
|
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|
|
/* Sample offset starting from 0 at aggregator.segment.start */
|
|
gint64 offset;
|
|
|
|
/* info structure passed to selected-samples signal, must only be accessed
|
|
* from the aggregate thread */
|
|
GstStructure *selected_samples_info;
|
|
|
|
/* Only access from src thread */
|
|
/* Messages to post after releasing locks */
|
|
GQueue messages;
|
|
};
|
|
|
|
#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);
|
|
static GstSample *gst_audio_aggregator_peek_next_sample (GstAggregator * agg,
|
|
GstAggregatorPad * aggpad);
|
|
|
|
#define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND)
|
|
#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
|
|
#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
|
|
#define DEFAULT_OUTPUT_BUFFER_DURATION_N (1)
|
|
#define DEFAULT_OUTPUT_BUFFER_DURATION_D (100)
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_OUTPUT_BUFFER_DURATION,
|
|
PROP_ALIGNMENT_THRESHOLD,
|
|
PROP_DISCONT_WAIT,
|
|
PROP_OUTPUT_BUFFER_DURATION_FRACTION,
|
|
};
|
|
|
|
G_DEFINE_ABSTRACT_TYPE_WITH_PRIVATE (GstAudioAggregator, gst_audio_aggregator,
|
|
GST_TYPE_AGGREGATOR);
|
|
|
|
static GstBuffer *
|
|
gst_audio_aggregator_convert_buffer (GstAudioAggregator * aagg, GstPad * pad,
|
|
GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * buffer)
|
|
{
|
|
GstAudioAggregatorPadClass *klass = GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad);
|
|
GstAudioAggregatorPad *aaggpad = GST_AUDIO_AGGREGATOR_PAD (pad);
|
|
|
|
g_assert (klass->convert_buffer);
|
|
|
|
return klass->convert_buffer (aaggpad, in_info, out_info, buffer);
|
|
}
|
|
|
|
static void
|
|
gst_audio_aggregator_translate_output_buffer_duration (GstAudioAggregator *
|
|
aagg, GstClockTime duration)
|
|
{
|
|
gint gcd;
|
|
|
|
aagg->priv->output_buffer_duration_n = duration;
|
|
aagg->priv->output_buffer_duration_d = GST_SECOND;
|
|
|
|
gcd = gst_util_greatest_common_divisor (aagg->priv->output_buffer_duration_n,
|
|
aagg->priv->output_buffer_duration_d);
|
|
|
|
if (gcd) {
|
|
aagg->priv->output_buffer_duration_n /= gcd;
|
|
aagg->priv->output_buffer_duration_d /= gcd;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_audio_aggregator_update_samples_per_buffer (GstAudioAggregator * aagg)
|
|
{
|
|
gboolean ret = TRUE;
|
|
GstAudioAggregatorPad *srcpad =
|
|
GST_AUDIO_AGGREGATOR_PAD (GST_AGGREGATOR_SRC_PAD (aagg));
|
|
|
|
if (!srcpad->info.finfo
|
|
|| GST_AUDIO_INFO_FORMAT (&srcpad->info) == GST_AUDIO_FORMAT_UNKNOWN) {
|
|
ret = FALSE;
|
|
goto out;
|
|
}
|
|
|
|
aagg->priv->samples_per_buffer =
|
|
(((guint64) GST_AUDIO_INFO_RATE (&srcpad->info)) *
|
|
aagg->priv->output_buffer_duration_n) /
|
|
aagg->priv->output_buffer_duration_d;
|
|
|
|
if (aagg->priv->samples_per_buffer == 0) {
|
|
ret = FALSE;
|
|
goto out;
|
|
}
|
|
|
|
aagg->priv->error_per_buffer =
|
|
(((guint64) GST_AUDIO_INFO_RATE (&srcpad->info)) *
|
|
aagg->priv->output_buffer_duration_n) %
|
|
aagg->priv->output_buffer_duration_d;
|
|
aagg->priv->accumulated_error = 0;
|
|
|
|
GST_DEBUG_OBJECT (aagg, "Buffer duration: %u/%u",
|
|
aagg->priv->output_buffer_duration_n,
|
|
aagg->priv->output_buffer_duration_d);
|
|
GST_DEBUG_OBJECT (aagg, "Samples per buffer: %u (error: %u/%u)",
|
|
aagg->priv->samples_per_buffer, aagg->priv->error_per_buffer,
|
|
aagg->priv->output_buffer_duration_d);
|
|
|
|
out:
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_audio_aggregator_recalculate_latency (GstAudioAggregator * aagg)
|
|
{
|
|
guint64 latency = gst_util_uint64_scale_int (GST_SECOND,
|
|
aagg->priv->output_buffer_duration_n,
|
|
aagg->priv->output_buffer_duration_d);
|
|
|
|
gst_aggregator_set_latency (GST_AGGREGATOR (aagg), latency, latency);
|
|
|
|
GST_OBJECT_LOCK (aagg);
|
|
/* Force recalculating in aggregate */
|
|
aagg->priv->samples_per_buffer = 0;
|
|
GST_OBJECT_UNLOCK (aagg);
|
|
}
|
|
|
|
static void
|
|
gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = (GObjectClass *) klass;
|
|
GstAggregatorClass *gstaggregator_class = (GstAggregatorClass *) klass;
|
|
|
|
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_aggregator_simple_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;
|
|
gstaggregator_class->peek_next_sample = gst_audio_aggregator_peek_next_sample;
|
|
|
|
klass->create_output_buffer = gst_audio_aggregator_create_output_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));
|
|
|
|
/**
|
|
* GstAudioAggregator:output-buffer-duration-fraction:
|
|
*
|
|
* Output block size in nanoseconds, expressed as a fraction.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_OUTPUT_BUFFER_DURATION_FRACTION,
|
|
gst_param_spec_fraction ("output-buffer-duration-fraction",
|
|
"Output buffer duration fraction",
|
|
"Output block size in nanoseconds, expressed as a fraction", 1,
|
|
G_MAXINT, G_MAXINT, 1, DEFAULT_OUTPUT_BUFFER_DURATION_N,
|
|
DEFAULT_OUTPUT_BUFFER_DURATION_D,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
|
|
GST_PARAM_MUTABLE_READY));
|
|
|
|
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 = gst_audio_aggregator_get_instance_private (aagg);
|
|
|
|
g_mutex_init (&aagg->priv->mutex);
|
|
|
|
aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
|
|
aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT;
|
|
|
|
gst_audio_aggregator_translate_output_buffer_duration (aagg,
|
|
DEFAULT_OUTPUT_BUFFER_DURATION);
|
|
gst_audio_aggregator_recalculate_latency (aagg);
|
|
|
|
aagg->current_caps = NULL;
|
|
|
|
aagg->priv->selected_samples_info =
|
|
gst_structure_new_empty ("GstAudioAggregatorSelectedSamplesInfo");
|
|
|
|
g_queue_init (&aagg->priv->messages);
|
|
}
|
|
|
|
static void
|
|
gst_audio_aggregator_dispose (GObject * object)
|
|
{
|
|
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object);
|
|
|
|
gst_caps_replace (&aagg->current_caps, NULL);
|
|
|
|
gst_clear_structure (&aagg->priv->selected_samples_info);
|
|
|
|
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:
|
|
gst_audio_aggregator_translate_output_buffer_duration (aagg,
|
|
g_value_get_uint64 (value));
|
|
g_object_notify (object, "output-buffer-duration-fraction");
|
|
gst_audio_aggregator_recalculate_latency (aagg);
|
|
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;
|
|
case PROP_OUTPUT_BUFFER_DURATION_FRACTION:
|
|
aagg->priv->output_buffer_duration_n =
|
|
gst_value_get_fraction_numerator (value);
|
|
aagg->priv->output_buffer_duration_d =
|
|
gst_value_get_fraction_denominator (value);
|
|
g_object_notify (object, "output-buffer-duration");
|
|
gst_audio_aggregator_recalculate_latency (aagg);
|
|
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, gst_util_uint64_scale_int (GST_SECOND,
|
|
aagg->priv->output_buffer_duration_n,
|
|
aagg->priv->output_buffer_duration_d));
|
|
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;
|
|
case PROP_OUTPUT_BUFFER_DURATION_FRACTION:
|
|
gst_value_set_fraction (value, aagg->priv->output_buffer_duration_n,
|
|
aagg->priv->output_buffer_duration_d);
|
|
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;
|
|
|
|
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);
|
|
|
|
/* If we already have a configured pad, assume that we can only configure
|
|
* to the very same format filtered with the template caps and continue
|
|
* with the result of that as the template caps */
|
|
|
|
if (first_configured_pad) {
|
|
GstCaps *first_configured_caps =
|
|
gst_audio_info_to_caps (&first_configured_pad->info);
|
|
GstCaps *tmp;
|
|
|
|
tmp =
|
|
gst_caps_intersect_full (sink_template_caps, first_configured_caps,
|
|
GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (first_configured_caps);
|
|
gst_caps_unref (sink_template_caps);
|
|
sink_template_caps = tmp;
|
|
|
|
gst_object_unref (first_configured_pad);
|
|
}
|
|
|
|
/* If we have downstream caps, filter them against our template caps or
|
|
* the filtered first configured pad caps from above */
|
|
if (downstream_caps) {
|
|
sink_caps =
|
|
gst_caps_intersect_full (sink_template_caps, downstream_caps,
|
|
GST_CAPS_INTERSECT_FIRST);
|
|
} else {
|
|
sink_caps = gst_caps_ref (sink_template_caps);
|
|
}
|
|
|
|
if (filter) {
|
|
GstCaps *tmp = gst_caps_intersect_full (sink_caps, filter,
|
|
GST_CAPS_INTERSECT_FIRST);
|
|
|
|
gst_caps_unref (sink_caps);
|
|
sink_caps = tmp;
|
|
}
|
|
|
|
gst_caps_unref (sink_template_caps);
|
|
|
|
if (downstream_caps)
|
|
gst_caps_unref (downstream_caps);
|
|
|
|
GST_INFO_OBJECT (pad, "returned sink caps : %" GST_PTR_FORMAT, sink_caps);
|
|
|
|
return sink_caps;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_audio_aggregator_convert_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;
|
|
|
|
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);
|
|
|
|
/* We can convert between all formats except for the sample rate, which has
|
|
* to match. */
|
|
|
|
/* If we have a first configured pad, we can only convert everything except
|
|
* for the sample rate, so modify our template caps to have exactly that
|
|
* sample rate in all structures */
|
|
if (first_configured_pad) {
|
|
GST_INFO_OBJECT (pad, "first configured pad has sample rate %d",
|
|
first_configured_pad->info.rate);
|
|
sink_template_caps = gst_caps_make_writable (sink_template_caps);
|
|
gst_caps_set_simple (sink_template_caps, "rate", G_TYPE_INT,
|
|
first_configured_pad->info.rate, NULL);
|
|
gst_object_unref (first_configured_pad);
|
|
}
|
|
|
|
/* Now if we have downstream caps, filter against the template caps from
|
|
* above, i.e. with potentially fixated sample rate field already. This
|
|
* filters out any structures with unsupported rates.
|
|
*
|
|
* Afterwards we create new caps that only take over the rate fields of the
|
|
* remaining downstream caps, and filter that against the plain template
|
|
* caps to get the resulting allowed caps with conversion for everything but
|
|
* the rate */
|
|
if (downstream_caps) {
|
|
GstCaps *tmp;
|
|
guint i, n;
|
|
|
|
tmp =
|
|
gst_caps_intersect_full (sink_template_caps, downstream_caps,
|
|
GST_CAPS_INTERSECT_FIRST);
|
|
|
|
n = gst_caps_get_size (tmp);
|
|
sink_caps = gst_caps_new_empty ();
|
|
for (i = 0; i < n; i++) {
|
|
GstStructure *s = gst_caps_get_structure (tmp, i);
|
|
GstStructure *new_s =
|
|
gst_structure_new_empty (gst_structure_get_name (s));
|
|
gst_structure_set_value (new_s, "rate", gst_structure_get_value (s,
|
|
"rate"));
|
|
sink_caps = gst_caps_merge_structure (sink_caps, new_s);
|
|
}
|
|
gst_caps_unref (tmp);
|
|
tmp = sink_caps;
|
|
|
|
sink_caps =
|
|
gst_caps_intersect_full (sink_template_caps, tmp,
|
|
GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (tmp);
|
|
} else {
|
|
sink_caps = gst_caps_ref (sink_template_caps);
|
|
}
|
|
|
|
/* And finally filter anything that remains against the filter caps */
|
|
if (filter) {
|
|
GstCaps *tmp =
|
|
gst_caps_intersect_full (filter, sink_caps, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (sink_caps);
|
|
sink_caps = tmp;
|
|
}
|
|
|
|
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);
|
|
GstAudioInfo info;
|
|
gboolean ret = TRUE;
|
|
gboolean downstream_supports_rate = TRUE;
|
|
|
|
if (!gst_audio_info_from_caps (&info, caps)) {
|
|
GST_WARNING_OBJECT (aaggpad, "Rejecting invalid caps: %" GST_PTR_FORMAT,
|
|
caps);
|
|
return FALSE;
|
|
}
|
|
|
|
/* 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
|
|
*/
|
|
|
|
/* Only check against the downstream caps if we didn't configure any caps
|
|
* so far. Otherwise we already know that downstream supports the rate
|
|
* because we negotiated with downstream */
|
|
if (!first_configured_pad) {
|
|
GstCaps *downstream_caps = gst_pad_get_allowed_caps (agg->srcpad);
|
|
|
|
/* Returns NULL if there is no downstream peer */
|
|
if (downstream_caps) {
|
|
GstCaps *rate_caps =
|
|
gst_caps_new_simple ("audio/x-raw", "rate", G_TYPE_INT, info.rate,
|
|
NULL);
|
|
|
|
gst_caps_set_features_simple (rate_caps,
|
|
gst_caps_features_copy (GST_CAPS_FEATURES_ANY));
|
|
|
|
downstream_supports_rate =
|
|
gst_caps_can_intersect (rate_caps, downstream_caps);
|
|
gst_caps_unref (rate_caps);
|
|
gst_caps_unref (downstream_caps);
|
|
}
|
|
}
|
|
|
|
if (!downstream_supports_rate || (first_configured_pad
|
|
&& info.rate != first_configured_pad->info.rate)) {
|
|
GST_WARNING_OBJECT (aaggpad,
|
|
"Sample rate %d can't be configured (downstream supported: %d, configured rate: %d)",
|
|
info.rate, downstream_supports_rate,
|
|
first_configured_pad ? first_configured_pad->info.rate : 0);
|
|
gst_pad_push_event (GST_PAD (aaggpad), gst_event_new_reconfigure ());
|
|
ret = FALSE;
|
|
} else {
|
|
GstAudioAggregatorPadClass *klass =
|
|
GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
|
|
GST_OBJECT_LOCK (aaggpad);
|
|
aaggpad->info = info;
|
|
if (klass->update_conversion_info)
|
|
klass->update_conversion_info (aaggpad);
|
|
GST_OBJECT_UNLOCK (aaggpad);
|
|
}
|
|
|
|
if (first_configured_pad)
|
|
gst_object_unref (first_configured_pad);
|
|
|
|
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)
|
|
{
|
|
GstAudioAggregatorPad *first_configured_pad = NULL;
|
|
|
|
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad)->convert_buffer)
|
|
first_configured_pad = gst_audio_aggregator_get_first_configured_pad (agg);
|
|
|
|
caps = gst_caps_make_writable (caps);
|
|
|
|
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;
|
|
gint channels;
|
|
|
|
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_structure_get_int (s, "channels", &channels);
|
|
|
|
if (!gst_structure_has_field (s, "channel-mask") && channels > 2) {
|
|
guint64 mask;
|
|
|
|
if (!gst_structure_get (s2, "channel-mask", GST_TYPE_BITMASK, &mask,
|
|
NULL)) {
|
|
mask = gst_audio_channel_get_fallback_mask (channels);
|
|
}
|
|
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK, mask, NULL);
|
|
}
|
|
|
|
gst_caps_unref (first_configured_caps);
|
|
gst_object_unref (first_configured_pad);
|
|
} else {
|
|
GstStructure *s;
|
|
gint channels;
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
gst_structure_fixate_field_nearest_int (s, "rate", GST_AUDIO_DEF_RATE);
|
|
gst_structure_fixate_field_string (s, "format", GST_AUDIO_NE ("S16"));
|
|
gst_structure_fixate_field_string (s, "layout", "interleaved");
|
|
gst_structure_fixate_field_nearest_int (s, "channels", 2);
|
|
|
|
if (gst_structure_get_int (s, "channels", &channels) && channels > 2) {
|
|
if (!gst_structure_has_field_typed (s, "channel-mask", GST_TYPE_BITMASK))
|
|
gst_structure_set (s, "channel-mask", GST_TYPE_BITMASK, 0ULL, NULL);
|
|
}
|
|
}
|
|
|
|
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, GstAudioInfo * old_info)
|
|
{
|
|
GList *l;
|
|
|
|
for (l = GST_ELEMENT (aagg)->sinkpads; l; l = l->next) {
|
|
GstAudioAggregatorPad *aaggpad = l->data;
|
|
GstAudioAggregatorPadClass *klass =
|
|
GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (aaggpad);
|
|
|
|
if (klass->update_conversion_info)
|
|
klass->update_conversion_info (aaggpad);
|
|
|
|
/* If we currently were mixing a buffer, we need to convert it to the new
|
|
* format */
|
|
if (aaggpad->priv->buffer) {
|
|
GstBuffer *new_converted_buffer =
|
|
gst_audio_aggregator_convert_buffer (aagg, GST_PAD (aaggpad),
|
|
old_info, new_info, aaggpad->priv->buffer);
|
|
gst_buffer_replace (&aaggpad->priv->buffer, new_converted_buffer);
|
|
gst_buffer_unref (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);
|
|
GstAudioInfo info;
|
|
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
|
|
|
|
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 (!gst_audio_info_is_equal (&info, &srcpad->info)) {
|
|
GstAudioInfo old_info = srcpad->info;
|
|
GstAudioAggregatorPadClass *srcpad_klass =
|
|
GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (agg->srcpad);
|
|
|
|
GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps);
|
|
gst_caps_replace (&aagg->current_caps, caps);
|
|
|
|
if (old_info.rate != info.rate)
|
|
aagg->priv->offset = -1;
|
|
|
|
memcpy (&srcpad->info, &info, sizeof (info));
|
|
|
|
gst_audio_aggregator_update_converters (aagg, &info, &old_info);
|
|
|
|
if (srcpad_klass->update_conversion_info)
|
|
srcpad_klass->update_conversion_info (GST_AUDIO_AGGREGATOR_PAD (agg->
|
|
srcpad));
|
|
|
|
if (aagg->priv->current_buffer) {
|
|
GstBuffer *converted;
|
|
|
|
converted =
|
|
gst_audio_aggregator_convert_buffer (aagg, agg->srcpad, &old_info,
|
|
&info, aagg->priv->current_buffer);
|
|
gst_buffer_unref (aagg->priv->current_buffer);
|
|
aagg->priv->current_buffer = converted;
|
|
}
|
|
|
|
/* Force recalculating in aggregate */
|
|
aagg->priv->samples_per_buffer = 0;
|
|
}
|
|
|
|
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 = GST_AGGREGATOR_PAD (agg->srcpad)->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 != GST_AGGREGATOR_PAD (agg->srcpad)->segment.rate) {
|
|
GST_ERROR_OBJECT (aggpad,
|
|
"Got segment event with wrong rate %lf, expected %lf",
|
|
segment->rate, GST_AGGREGATOR_PAD (agg->srcpad)->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_audio_aggregator_pad_reset_qos (pad);
|
|
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 (!res) {
|
|
if (event)
|
|
gst_event_unref (event);
|
|
return res;
|
|
}
|
|
|
|
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);
|
|
if (GST_IS_AUDIO_AGGREGATOR_CONVERT_PAD (aggpad)) {
|
|
caps =
|
|
gst_audio_aggregator_convert_sink_getcaps (GST_PAD (aggpad), agg,
|
|
filter);
|
|
} else {
|
|
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
|
|
* each other 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);
|
|
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
|
|
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 (&GST_AGGREGATOR_PAD (agg->srcpad)->
|
|
segment, GST_FORMAT_TIME,
|
|
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position));
|
|
res = TRUE;
|
|
break;
|
|
case GST_FORMAT_BYTES:
|
|
if (GST_AUDIO_INFO_BPF (&srcpad->info)) {
|
|
gst_query_set_position (query, format, aagg->priv->offset *
|
|
GST_AUDIO_INFO_BPF (&srcpad->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);
|
|
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
|
|
aagg->priv->offset = -1;
|
|
gst_audio_info_init (&GST_AUDIO_AGGREGATOR_PAD (agg->srcpad)->info);
|
|
gst_caps_replace (&aagg->current_caps, NULL);
|
|
gst_buffer_replace (&aagg->priv->current_buffer, NULL);
|
|
aagg->priv->accumulated_error = 0;
|
|
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);
|
|
GST_AGGREGATOR_PAD (agg->srcpad)->segment.position = -1;
|
|
aagg->priv->offset = -1;
|
|
aagg->priv->accumulated_error = 0;
|
|
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;
|
|
|
|
/* Guard against invalid audio info, we just don't clip here then */
|
|
if (!GST_AUDIO_INFO_IS_VALID (&pad->info))
|
|
return buffer;
|
|
|
|
GST_OBJECT_LOCK (bpad);
|
|
rate = GST_AUDIO_INFO_RATE (&pad->info);
|
|
bpf = GST_AUDIO_INFO_BPF (&pad->info);
|
|
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 audio aggregator lock.
|
|
* Should only be called on the output queue.
|
|
*/
|
|
static GstClockTime
|
|
gst_audio_aggregator_pad_enqueue_qos_message (GstAudioAggregatorPad * pad,
|
|
GstAudioAggregator * aagg, guint64 samples)
|
|
{
|
|
GstAggregator *agg = GST_AGGREGATOR (aagg);
|
|
GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
|
|
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
|
|
|
|
guint rate_output = GST_AUDIO_INFO_RATE (&srcpad->info);
|
|
GstClockTime offset = gst_util_uint64_scale (GST_SECOND, pad->priv->position,
|
|
rate_output);
|
|
GstClockTime timestamp = GST_BUFFER_PTS (pad->priv->buffer) + offset;
|
|
GstClockTime running_time =
|
|
gst_segment_to_running_time (&aggpad->segment, GST_FORMAT_TIME,
|
|
timestamp);
|
|
GstClockTime stream_time = gst_segment_to_stream_time (&aggpad->segment,
|
|
GST_FORMAT_TIME, timestamp);
|
|
GstClockTime duration;
|
|
guint rate_input;
|
|
guint64 processed, dropped;
|
|
GstMessage *msg;
|
|
|
|
if (!pad->priv->qos_messages)
|
|
return running_time;
|
|
|
|
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
|
|
rate_input = GST_AUDIO_INFO_RATE (&srcpad->info);
|
|
else
|
|
rate_input = GST_AUDIO_INFO_RATE (&pad->info);
|
|
|
|
duration = gst_util_uint64_scale (samples, GST_SECOND, rate_input);
|
|
|
|
processed = gst_util_uint64_scale (pad->priv->processed, rate_input,
|
|
rate_output);
|
|
dropped = gst_util_uint64_scale (pad->priv->dropped, rate_output,
|
|
rate_output);
|
|
|
|
msg = gst_message_new_qos (GST_OBJECT (aggpad), TRUE, running_time,
|
|
stream_time, timestamp, duration);
|
|
gst_message_set_qos_stats (msg, GST_FORMAT_DEFAULT, processed, dropped);
|
|
|
|
g_queue_push_tail (&aagg->priv->messages, msg);
|
|
|
|
return running_time;
|
|
}
|
|
|
|
static void
|
|
gst_audio_aggregator_post_messages (GstAudioAggregator * aagg)
|
|
{
|
|
if (g_queue_get_length (&aagg->priv->messages) != 0) {
|
|
GstClockTime latency = gst_aggregator_get_latency (GST_AGGREGATOR (aagg));
|
|
gboolean is_live = GST_CLOCK_TIME_IS_VALID (latency);
|
|
GstElement *e = GST_ELEMENT (aagg);
|
|
GstMessage *msg;
|
|
|
|
while ((msg = g_queue_pop_head (&aagg->priv->messages))) {
|
|
if (is_live) {
|
|
GstStructure *s = gst_message_writable_structure (msg);
|
|
gst_structure_set (s, "live", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
}
|
|
|
|
gst_element_post_message (e, msg);
|
|
}
|
|
}
|
|
}
|
|
|
|
/* 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)
|
|
{
|
|
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);
|
|
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
|
|
|
|
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer) {
|
|
rate = GST_AUDIO_INFO_RATE (&srcpad->info);
|
|
bpf = GST_AUDIO_INFO_BPF (&srcpad->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->next_offset = -1;
|
|
} else {
|
|
pad->priv->next_offset += pad->priv->size;
|
|
}
|
|
|
|
if (pad->priv->output_offset == -1 || discont) {
|
|
GstClockTime start_running_time;
|
|
GstClockTime end_running_time;
|
|
GstClockTime segment_pos;
|
|
guint64 start_output_offset = -1;
|
|
guint64 end_output_offset = -1;
|
|
GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
|
|
|
|
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;
|
|
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) {
|
|
GstClockTime rt;
|
|
|
|
pad->priv->dropped += pad->priv->size;
|
|
rt = gst_audio_aggregator_pad_enqueue_qos_message (pad, aagg,
|
|
pad->priv->size);
|
|
GST_DEBUG_OBJECT (pad, "Dropped buffer of %u samples at running time %"
|
|
GST_TIME_FORMAT " because input buffer is entirely before current"
|
|
" output offset", pad->priv->size, GST_TIME_ARGS (rt));
|
|
|
|
pad->priv->position = 0;
|
|
pad->priv->size = 0;
|
|
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 != -1 &&
|
|
start_output_offset < pad->priv->output_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 if (pad->priv->output_offset != -1 &&
|
|
start_output_offset < pad->priv->output_offset) {
|
|
diff = pad->priv->output_offset - start_output_offset;
|
|
} else {
|
|
diff = aagg->priv->offset - start_output_offset;
|
|
}
|
|
|
|
pad->priv->dropped += MIN (diff, pad->priv->size);
|
|
if (diff != 0) {
|
|
GstClockTime rt;
|
|
|
|
rt = gst_audio_aggregator_pad_enqueue_qos_message (pad, aagg, diff);
|
|
GST_DEBUG_OBJECT (pad, "Dropped %u samples at running time %"
|
|
GST_TIME_FORMAT " because input buffer starts before current"
|
|
" output offset", diff, GST_TIME_ARGS (rt));
|
|
}
|
|
|
|
pad->priv->position += diff;
|
|
if (start_output_offset != -1)
|
|
start_output_offset += diff;
|
|
if (pad->priv->position >= pad->priv->size) {
|
|
/* Empty buffer, drop */
|
|
pad->priv->dropped += pad->priv->size;
|
|
pad->priv->position = 0;
|
|
pad->priv->size = 0;
|
|
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)
|
|
pad->priv->output_offset = aagg->priv->offset;
|
|
else
|
|
pad->priv->output_offset = start_output_offset;
|
|
|
|
if (pad->priv->next_offset == -1)
|
|
pad->priv->next_offset = end_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);
|
|
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->processed += overlap;
|
|
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_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;
|
|
GstAggregator *agg = GST_AGGREGATOR (aagg);
|
|
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
|
|
|
|
gst_aggregator_get_allocator (GST_AGGREGATOR (aagg), &allocator, ¶ms);
|
|
|
|
GST_DEBUG ("Creating output buffer with size %d",
|
|
num_frames * GST_AUDIO_INFO_BPF (&srcpad->info));
|
|
|
|
outbuf = gst_buffer_new_allocate (allocator, num_frames *
|
|
GST_AUDIO_INFO_BPF (&srcpad->info), ¶ms);
|
|
|
|
if (allocator)
|
|
gst_object_unref (allocator);
|
|
|
|
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
|
|
gst_audio_format_info_fill_silence (srcpad->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 GstSample *
|
|
gst_audio_aggregator_peek_next_sample (GstAggregator * agg,
|
|
GstAggregatorPad * aggpad)
|
|
{
|
|
GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg);
|
|
GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad);
|
|
GstSample *sample = NULL;
|
|
|
|
if (pad->priv->buffer && pad->priv->output_offset >= aagg->priv->offset
|
|
&& pad->priv->output_offset <
|
|
aagg->priv->offset + aagg->priv->samples_per_buffer) {
|
|
GstCaps *caps = gst_pad_get_current_caps (GST_PAD (aggpad));
|
|
GstStructure *info =
|
|
gst_structure_new ("GstAudioAggregatorPadNextSampleInfo",
|
|
"output-offset", G_TYPE_UINT64, pad->priv->output_offset,
|
|
"position", G_TYPE_UINT, pad->priv->position,
|
|
"size", G_TYPE_UINT, pad->priv->size,
|
|
NULL);
|
|
|
|
sample = gst_sample_new (pad->priv->buffer, caps, &aggpad->segment, info);
|
|
gst_caps_unref (caps);
|
|
gst_structure_free (info);
|
|
}
|
|
|
|
return sample;
|
|
}
|
|
|
|
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 appropriate 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;
|
|
GstAudioAggregatorPad *srcpad = GST_AUDIO_AGGREGATOR_PAD (agg->srcpad);
|
|
GstSegment *agg_segment = &GST_AGGREGATOR_PAD (agg->srcpad)->segment;
|
|
|
|
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);
|
|
|
|
if (aagg->priv->samples_per_buffer == 0) {
|
|
if (!gst_audio_aggregator_update_samples_per_buffer (aagg)) {
|
|
GST_ERROR_OBJECT (aagg,
|
|
"Failed to calculate the number of samples per buffer");
|
|
GST_OBJECT_UNLOCK (agg);
|
|
goto not_negotiated;
|
|
}
|
|
}
|
|
|
|
/* 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;
|
|
}
|
|
|
|
rate = GST_AUDIO_INFO_RATE (&srcpad->info);
|
|
bpf = GST_AUDIO_INFO_BPF (&srcpad->info);
|
|
|
|
if (G_UNLIKELY (srcpad->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
|
|
if (timeout) {
|
|
GstClockTime output_buffer_duration;
|
|
GST_DEBUG_OBJECT (aagg,
|
|
"Got timeout before receiving any caps, don't output anything");
|
|
|
|
blocksize = aagg->priv->samples_per_buffer;
|
|
if (aagg->priv->error_per_buffer + aagg->priv->accumulated_error >=
|
|
aagg->priv->output_buffer_duration_d)
|
|
blocksize += 1;
|
|
aagg->priv->accumulated_error =
|
|
(aagg->priv->accumulated_error +
|
|
aagg->priv->error_per_buffer) % aagg->priv->output_buffer_duration_d;
|
|
|
|
output_buffer_duration =
|
|
gst_util_uint64_scale (blocksize, GST_SECOND, rate);
|
|
|
|
/* Advance position */
|
|
if (agg_segment->rate > 0.0)
|
|
agg_segment->position += output_buffer_duration;
|
|
else if (agg_segment->position > output_buffer_duration)
|
|
agg_segment->position -= 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;
|
|
}
|
|
}
|
|
|
|
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);
|
|
}
|
|
|
|
if (aagg->priv->current_buffer == NULL) {
|
|
blocksize = aagg->priv->samples_per_buffer;
|
|
|
|
if (aagg->priv->error_per_buffer + aagg->priv->accumulated_error >=
|
|
aagg->priv->output_buffer_duration_d)
|
|
blocksize += 1;
|
|
|
|
aagg->priv->current_blocksize = blocksize;
|
|
|
|
aagg->priv->accumulated_error =
|
|
(aagg->priv->accumulated_error +
|
|
aagg->priv->error_per_buffer) % aagg->priv->output_buffer_duration_d;
|
|
|
|
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);
|
|
} else {
|
|
blocksize = aagg->priv->current_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);
|
|
|
|
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);
|
|
GstBuffer *input_buffer;
|
|
|
|
if (!pad_eos)
|
|
is_eos = FALSE;
|
|
|
|
input_buffer = gst_aggregator_pad_peek_buffer (aggpad);
|
|
|
|
GST_OBJECT_LOCK (pad);
|
|
if (!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 (&srcpad->info))));
|
|
}
|
|
} else if (!pad_eos) {
|
|
is_done = FALSE;
|
|
}
|
|
GST_OBJECT_UNLOCK (pad);
|
|
continue;
|
|
} else if (!GST_AUDIO_INFO_IS_VALID (&pad->info)) {
|
|
GST_OBJECT_UNLOCK (pad);
|
|
GST_OBJECT_UNLOCK (agg);
|
|
goto not_negotiated;
|
|
}
|
|
|
|
/* New buffer? */
|
|
if (!pad->priv->buffer) {
|
|
if (GST_AUDIO_AGGREGATOR_PAD_GET_CLASS (pad)->convert_buffer)
|
|
pad->priv->buffer =
|
|
gst_audio_aggregator_convert_buffer
|
|
(aagg, GST_PAD (pad), &pad->info, &srcpad->info, input_buffer);
|
|
else
|
|
pad->priv->buffer = gst_buffer_ref (input_buffer);
|
|
|
|
if (!gst_audio_aggregator_fill_buffer (aagg, pad)) {
|
|
gst_buffer_replace (&pad->priv->buffer, NULL);
|
|
gst_buffer_unref (input_buffer);
|
|
dropped = TRUE;
|
|
GST_OBJECT_UNLOCK (pad);
|
|
|
|
gst_aggregator_pad_drop_buffer (aggpad);
|
|
continue;
|
|
}
|
|
}
|
|
gst_buffer_unref (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->dropped += diff;
|
|
if (diff != 0) {
|
|
GstClockTime rt;
|
|
|
|
rt = gst_audio_aggregator_pad_enqueue_qos_message (pad, aagg, diff);
|
|
GST_DEBUG_OBJECT (pad, "Dropped %" G_GINT64_FORMAT " samples at"
|
|
" running time %" GST_TIME_FORMAT " because input buffer is before"
|
|
" output offset", diff, GST_TIME_ARGS (rt));
|
|
}
|
|
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 (&srcpad->info))), pad->priv->buffer);
|
|
/* Buffer done, drop it */
|
|
gst_buffer_replace (&pad->priv->buffer, NULL);
|
|
dropped = TRUE;
|
|
GST_OBJECT_UNLOCK (pad);
|
|
gst_aggregator_pad_drop_buffer (aggpad);
|
|
continue;
|
|
}
|
|
}
|
|
|
|
g_assert (pad->priv->buffer);
|
|
GST_OBJECT_UNLOCK (pad);
|
|
}
|
|
GST_OBJECT_UNLOCK (agg);
|
|
|
|
gst_audio_aggregator_post_messages (aagg);
|
|
|
|
{
|
|
gst_structure_set (aagg->priv->selected_samples_info, "offset",
|
|
G_TYPE_UINT64, aagg->priv->offset, "frames", G_TYPE_UINT, blocksize,
|
|
NULL);
|
|
gst_aggregator_selected_samples (agg, agg_segment->position,
|
|
GST_CLOCK_TIME_NONE, next_timestamp - agg_segment->position,
|
|
aagg->priv->selected_samples_info);
|
|
}
|
|
|
|
GST_OBJECT_LOCK (agg);
|
|
for (iter = element->sinkpads; iter; iter = iter->next) {
|
|
GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data;
|
|
GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data;
|
|
|
|
GST_OBJECT_LOCK (pad);
|
|
|
|
if (pad->priv->buffer && 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;
|
|
}
|
|
}
|