/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2001 Thomas * 2005,2006 Wim Taymans * 2013 Sebastian Dröge * 2014 Collabora * Olivier Crete * * gstaudioaggregator.c: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION: gstaudioaggregator * @title: GstAudioAggregator * @short_description: Base class that manages a set of audio input pads * with the purpose of aggregating or mixing their raw audio input buffers * @see_also: #GstAggregator, #GstAudioMixer * * Subclasses must use (a subclass of) #GstAudioAggregatorPad for both * their source and sink pads, * gst_element_class_add_static_pad_template_with_gtype() is a convenient * helper. * * #GstAudioAggregator can perform conversion on the data arriving * on its sink pads, based on the format expected downstream: in order * to enable that behaviour, the GType of the sink pads must either be * a (subclass of) #GstAudioAggregatorConvertPad to use the default * #GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad * implementing #GstAudioAggregatorPadClass.convert_buffer. * * To allow for the output caps to change, the mechanism is the same as * above, with the GType of the source pad. * * See #GstAudioMixer for an example. * * When conversion is enabled, #GstAudioAggregator will accept * any type of raw audio caps and perform conversion * on the data arriving on its sink pads, with whatever downstream * expects as the target format. * * In case downstream caps are not fully fixated, it will use * the first configured sink pad to finish fixating its source pad * caps. * * A notable exception for now is the sample rate, sink pads must * have the same sample rate as either the downstream requirement, * or the first configured pad, or a combination of both (when * downstream specifies a range or a set of acceptable rates). * * The #GstAggregator::samples-selected signal is provided with some * additional information about the output buffer: * - "offset" G_TYPE_UINT64 Offset in samples since segment start * for the position that is next to be filled in the output buffer. * - "frames" G_TYPE_UINT Number of frames per output buffer. * * In addition the gst_aggregator_peek_next_sample() function returns * additional information in the info #GstStructure of the returned sample: * - "output-offset" G_TYPE_UINT64 Sample offset in output segment relative to * the output segment's start where the current position of this input * buffer would be placed * - "position" G_TYPE_UINT current position in the input buffer in samples * - "size" G_TYPE_UINT size of the input buffer in samples */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "gstaudioaggregator.h" #include GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug); #define GST_CAT_DEFAULT audio_aggregator_debug enum { PROP_PAD_0, PROP_PAD_QOS_MESSAGES, }; struct _GstAudioAggregatorPadPrivate { /* All members are protected by the pad object lock */ GstBuffer *buffer; /* current buffer we're mixing, for comparison with a new input buffer from aggregator to see if we need to update our cached values. */ guint position, size; /* position in the input buffer and size of the input buffer in number of samples */ guint64 output_offset; /* Sample offset in output segment relative to srcpad.segment.start where the current position of this input_buffer would be placed. */ guint64 next_offset; /* Next expected sample offset relative to pad.segment.start */ /* Last time we noticed a discont */ GstClockTime discont_time; /* A new unhandled segment event has been received */ gboolean new_segment; guint64 processed; /* Number of samples processed since the element came out of READY */ guint64 dropped; /* Number of sampels dropped since the element came out of READY */ gboolean qos_messages; /* Property to decide to send QoS messages or not */ }; /***************************************** * GstAudioAggregatorPad implementation * *****************************************/ G_DEFINE_TYPE_WITH_PRIVATE (GstAudioAggregatorPad, gst_audio_aggregator_pad, GST_TYPE_AGGREGATOR_PAD); static GstFlowReturn gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad, GstAggregator * aggregator); static void gst_audio_aggregator_pad_finalize (GObject * object) { GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) object; gst_buffer_replace (&pad->priv->buffer, NULL); G_OBJECT_CLASS (gst_audio_aggregator_pad_parent_class)->finalize (object); } static void gst_audio_aggregator_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (object); switch (prop_id) { case PROP_PAD_QOS_MESSAGES: GST_OBJECT_LOCK (pad); g_value_set_boolean (value, pad->priv->qos_messages); GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_aggregator_pad_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (object); switch (prop_id) { case PROP_PAD_QOS_MESSAGES: GST_OBJECT_LOCK (pad); pad->priv->qos_messages = g_value_get_boolean (value); GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_aggregator_pad_class_init (GstAudioAggregatorPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass; gobject_class->set_property = gst_audio_aggregator_pad_set_property; gobject_class->get_property = gst_audio_aggregator_pad_get_property; gobject_class->finalize = gst_audio_aggregator_pad_finalize; aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audio_aggregator_pad_flush_pad); /** * GstAudioAggregatorPad:qos-messages: * * Emit QoS messages when dropping buffers. * * Since: 1.20 */ g_object_class_install_property (gobject_class, PROP_PAD_QOS_MESSAGES, g_param_spec_boolean ("qos-messages", "Quality of Service Messages", "Emit QoS messages when dropping buffers", FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } static void gst_audio_aggregator_pad_init (GstAudioAggregatorPad * pad) { pad->priv = gst_audio_aggregator_pad_get_instance_private (pad); gst_audio_info_init (&pad->info); pad->priv->buffer = NULL; pad->priv->position = 0; pad->priv->size = 0; pad->priv->output_offset = -1; pad->priv->next_offset = -1; pad->priv->discont_time = GST_CLOCK_TIME_NONE; } /* Must be called from srcpad thread or when it is stopped */ static void gst_audio_aggregator_pad_reset_qos (GstAudioAggregatorPad * pad) { pad->priv->dropped = 0; pad->priv->processed = 0; } static GstFlowReturn gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad, GstAggregator * aggregator) { GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad); GST_OBJECT_LOCK (aggpad); pad->priv->position = pad->priv->size = 0; pad->priv->output_offset = pad->priv->next_offset = -1; pad->priv->discont_time = GST_CLOCK_TIME_NONE; gst_buffer_replace (&pad->priv->buffer, NULL); gst_audio_aggregator_pad_reset_qos (pad); GST_OBJECT_UNLOCK (aggpad); return GST_FLOW_OK; } enum { PROP_CONVERT_PAD_0, PROP_CONVERT_PAD_CONVERTER_CONFIG }; struct _GstAudioAggregatorConvertPadPrivate { /* All members are protected by the pad object lock */ GstAudioConverter *converter; GstStructure *converter_config; gboolean converter_config_changed; }; G_DEFINE_TYPE_WITH_PRIVATE (GstAudioAggregatorConvertPad, gst_audio_aggregator_convert_pad, GST_TYPE_AUDIO_AGGREGATOR_PAD); static void gst_audio_aggregator_convert_pad_update_converter (GstAudioAggregatorConvertPad * aaggcpad, GstAudioInfo * in_info, GstAudioInfo * out_info) { GstStructure *config = aaggcpad->priv->converter_config; GstAudioConverter *converter; if (!aaggcpad->priv->converter_config_changed) return; g_clear_pointer (&aaggcpad->priv->converter, gst_audio_converter_free); aaggcpad->priv->converter_config_changed = FALSE; if (in_info->finfo->format == GST_AUDIO_FORMAT_UNKNOWN) { /* If we haven't received caps yet, this pad should not have * a buffer to convert anyway */ return; } converter = gst_audio_converter_new (GST_AUDIO_CONVERTER_FLAG_NONE, in_info, out_info, config ? gst_structure_copy (config) : NULL); if (converter == NULL) { /* FIXME: Not converting when we need to but the config is invalid (e.g. * because the mix-matrix is not the right size) produces garbage. An * invalid config should cause a GST_FLOW_NOT_NEGOTIATED. */ return; } if (!gst_audio_converter_is_passthrough (converter)) aaggcpad->priv->converter = converter; else gst_audio_converter_free (converter); } static void gst_audio_aggregator_pad_update_conversion_info (GstAudioAggregatorPad * aaggpad) { GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad)->priv->converter_config_changed = TRUE; } static GstBuffer * gst_audio_aggregator_convert_pad_convert_buffer (GstAudioAggregatorPad * aaggpad, GstAudioInfo * in_info, GstAudioInfo * out_info, GstBuffer * input_buffer) { GstBuffer *res; GstAudioAggregatorConvertPad *aaggcpad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (aaggpad); gst_audio_aggregator_convert_pad_update_converter (aaggcpad, in_info, out_info); if (aaggcpad->priv->converter) { gint insize = gst_buffer_get_size (input_buffer); gsize insamples = insize / in_info->bpf; gsize outsamples = gst_audio_converter_get_out_frames (aaggcpad->priv->converter, insamples); gint outsize = outsamples * out_info->bpf; GstMapInfo inmap, outmap; res = gst_buffer_new_allocate (NULL, outsize, NULL); /* We create a perfectly similar buffer, except obviously for * its converted contents */ gst_buffer_copy_into (res, input_buffer, GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_META, 0, -1); gst_buffer_map (input_buffer, &inmap, GST_MAP_READ); gst_buffer_map (res, &outmap, GST_MAP_WRITE); gst_audio_converter_samples (aaggcpad->priv->converter, GST_AUDIO_CONVERTER_FLAG_NONE, (gpointer *) & inmap.data, insamples, (gpointer *) & outmap.data, outsamples); gst_buffer_unmap (input_buffer, &inmap); gst_buffer_unmap (res, &outmap); } else { res = gst_buffer_ref (input_buffer); } return res; } static void gst_audio_aggregator_convert_pad_finalize (GObject * object) { GstAudioAggregatorConvertPad *pad = (GstAudioAggregatorConvertPad *) object; if (pad->priv->converter) gst_audio_converter_free (pad->priv->converter); if (pad->priv->converter_config) gst_structure_free (pad->priv->converter_config); G_OBJECT_CLASS (gst_audio_aggregator_convert_pad_parent_class)->finalize (object); } static void gst_audio_aggregator_convert_pad_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object); switch (prop_id) { case PROP_CONVERT_PAD_CONVERTER_CONFIG: GST_OBJECT_LOCK (pad); if (pad->priv->converter_config) g_value_set_boxed (value, pad->priv->converter_config); GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_aggregator_convert_pad_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioAggregatorConvertPad *pad = GST_AUDIO_AGGREGATOR_CONVERT_PAD (object); switch (prop_id) { case PROP_CONVERT_PAD_CONVERTER_CONFIG: GST_OBJECT_LOCK (pad); if (pad->priv->converter_config) gst_structure_free (pad->priv->converter_config); pad->priv->converter_config = g_value_dup_boxed (value); pad->priv->converter_config_changed = TRUE; GST_OBJECT_UNLOCK (pad); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_aggregator_convert_pad_class_init (GstAudioAggregatorConvertPadClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstAudioAggregatorPadClass *aaggpad_class = (GstAudioAggregatorPadClass *) klass; gobject_class->set_property = gst_audio_aggregator_convert_pad_set_property; gobject_class->get_property = gst_audio_aggregator_convert_pad_get_property; g_object_class_install_property (gobject_class, PROP_CONVERT_PAD_CONVERTER_CONFIG, g_param_spec_boxed ("converter-config", "Converter configuration", "A GstStructure describing the configuration that should be used " "when converting this pad's audio buffers", GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); aaggpad_class->convert_buffer = gst_audio_aggregator_convert_pad_convert_buffer; aaggpad_class->update_conversion_info = gst_audio_aggregator_pad_update_conversion_info; gobject_class->finalize = gst_audio_aggregator_convert_pad_finalize; } static void gst_audio_aggregator_convert_pad_init (GstAudioAggregatorConvertPad * pad) { pad->priv = gst_audio_aggregator_convert_pad_get_instance_private (pad); } /************************************** * GstAudioAggregator implementation * **************************************/ struct _GstAudioAggregatorPrivate { GMutex mutex; /* All three properties are unprotected, can't be modified while streaming */ /* Size in frames that is output per buffer */ GstClockTime alignment_threshold; GstClockTime discont_wait; gint output_buffer_duration_n; gint output_buffer_duration_d; guint samples_per_buffer; guint error_per_buffer; guint accumulated_error; guint current_blocksize; /* Protected by srcpad stream clock */ /* Output buffer starting at offset containing blocksize frames (calculated * from output_buffer_duration) */ GstBuffer *current_buffer; /* counters to keep track of timestamps */ /* Readable with object lock, writable with both aag lock and object lock */ /* Sample offset starting from 0 at aggregator.segment.start */ gint64 offset; /* 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 (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 = end_offset; } 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 && pad->priv->qos_messages) { 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 (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 || start_output_offset < aagg->priv->offset) pad->priv->output_offset = aagg->priv->offset; else if (pad->priv->output_offset != -1) pad->priv->output_offset = MAX (pad->priv->output_offset, start_output_offset); else pad->priv->output_offset = start_output_offset; GST_DEBUG_OBJECT (pad, "Buffer resynced: Pad offset %" G_GUINT64_FORMAT ", current audio aggregator offset %" G_GINT64_FORMAT, pad->priv->output_offset, aagg->priv->offset); } done: GST_LOG_OBJECT (pad, "Queued new buffer at offset %" G_GUINT64_FORMAT, pad->priv->output_offset); return TRUE; } /* Called with pad object lock held */ static gboolean gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad, GstBuffer * inbuf, GstBuffer * outbuf, guint blocksize) { guint overlap; guint out_start; gboolean filled; guint in_offset; gboolean pad_changed = FALSE; /* Overlap => mix */ if (aagg->priv->offset < pad->priv->output_offset) out_start = pad->priv->output_offset - aagg->priv->offset; else out_start = 0; overlap = pad->priv->size - pad->priv->position; if (overlap > blocksize - out_start) overlap = blocksize - out_start; if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) { /* skip gap buffer */ GST_LOG_OBJECT (pad, "skipping GAP buffer"); pad->priv->output_offset += pad->priv->size - pad->priv->position; pad->priv->position = pad->priv->size; gst_buffer_replace (&pad->priv->buffer, NULL); 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 && pad->priv->qos_messages) { 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; } }