/* GStreamer * Copyright (C) 1999,2000 Erik Walthinsen * 2001 Thomas * 2005,2006 Wim Taymans * 2013 Sebastian Dröge * 2014 Collabora * Olivier Crete * * gstaudioaggregator.c: * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION: gstaudioaggregator * @short_description: manages a set of pads with the purpose of * aggregating their buffers for raw audio * @see_also: #GstAggregator * */ #ifdef HAVE_CONFIG_H # include "config.h" #endif #include "gstaudioaggregator.h" #include GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug); #define GST_CAT_DEFAULT audio_aggregator_debug struct _GstAudioAggregatorPadPrivate { /* All members are protected by the pad object lock */ GstBuffer *buffer; /* current buffer we're mixing, for comparison with collect.buffer to see if we need to update our cached values. */ guint position, size; guint64 output_offset; /* Offset in output segment that collect.pos refers to in the current buffer. */ guint64 next_offset; /* Next expected offset in the input segment */ /* Last time we noticed a discont */ GstClockTime discont_time; /* A new unhandled segment event has been received */ gboolean new_segment; }; /***************************************** * GstAudioAggregatorPad implementation * *****************************************/ G_DEFINE_TYPE (GstAudioAggregatorPad, gst_audio_aggregator_pad, GST_TYPE_AGGREGATOR_PAD); static gboolean gst_audio_aggregator_pad_flush_pad (GstAggregatorPad * aggpad, GstAggregator * aggregator); static void gst_audio_aggregator_pad_class_init (GstAudioAggregatorPadClass * klass) { GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass; g_type_class_add_private (klass, sizeof (GstAudioAggregatorPadPrivate)); aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audio_aggregator_pad_flush_pad); } static void gst_audio_aggregator_pad_init (GstAudioAggregatorPad * pad) { pad->priv = G_TYPE_INSTANCE_GET_PRIVATE (pad, GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadPrivate); gst_audio_info_init (&pad->info); pad->priv->buffer = NULL; pad->priv->position = 0; pad->priv->size = 0; pad->priv->output_offset = -1; pad->priv->next_offset = -1; pad->priv->discont_time = GST_CLOCK_TIME_NONE; } static gboolean 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_OBJECT_UNLOCK (aggpad); return TRUE; } /************************************** * GstAudioAggregator implementation * **************************************/ struct _GstAudioAggregatorPrivate { GMutex mutex; gboolean send_caps; /* aagg lock */ /* All three properties are unprotected, can't be modified while streaming */ /* Size in frames that is output per buffer */ GstClockTime output_buffer_duration; GstClockTime alignment_threshold; GstClockTime discont_wait; /* Protected by srcpad stream clock */ /* Buffer starting at offset containing block_size frames */ GstBuffer *current_buffer; /* counters to keep track of timestamps */ /* Readable with object lock, writable with both aag lock and object lock */ gint64 offset; }; #define GST_AUDIO_AGGREGATOR_LOCK(self) g_mutex_lock (&(self)->priv->mutex); #define GST_AUDIO_AGGREGATOR_UNLOCK(self) g_mutex_unlock (&(self)->priv->mutex); static void gst_audio_aggregator_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_audio_aggregator_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_audio_aggregator_dispose (GObject * object); static gboolean gst_audio_aggregator_src_event (GstAggregator * agg, GstEvent * event); static gboolean gst_audio_aggregator_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad, GstEvent * event); static gboolean gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query); static gboolean gst_audio_aggregator_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 GstFlowReturn gst_audio_aggregator_do_clip (GstAggregator * agg, GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** outbuf); static GstFlowReturn gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout); #define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND) #define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND) #define DEFAULT_DISCONT_WAIT (1 * GST_SECOND) enum { PROP_0, PROP_OUTPUT_BUFFER_DURATION, PROP_ALIGNMENT_THRESHOLD, PROP_DISCONT_WAIT, }; G_DEFINE_ABSTRACT_TYPE (GstAudioAggregator, gst_audio_aggregator, GST_TYPE_AGGREGATOR); static GstClockTime gst_audio_aggregator_get_next_time (GstAggregator * agg) { GstClockTime next_time; GST_OBJECT_LOCK (agg); if (agg->segment.position == -1) next_time = agg->segment.start; else next_time = agg->segment.position; GST_OBJECT_UNLOCK (agg); return next_time; } static void gst_audio_aggregator_class_init (GstAudioAggregatorClass * klass) { GObjectClass *gobject_class = (GObjectClass *) klass; GstAggregatorClass *gstaggregator_class = (GstAggregatorClass *) klass; g_type_class_add_private (klass, sizeof (GstAudioAggregatorPrivate)); gobject_class->set_property = gst_audio_aggregator_set_property; gobject_class->get_property = gst_audio_aggregator_get_property; gobject_class->dispose = gst_audio_aggregator_dispose; gstaggregator_class->src_event = GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_event); gstaggregator_class->sink_event = GST_DEBUG_FUNCPTR (gst_audio_aggregator_sink_event); gstaggregator_class->src_query = GST_DEBUG_FUNCPTR (gst_audio_aggregator_src_query); gstaggregator_class->start = gst_audio_aggregator_start; gstaggregator_class->stop = gst_audio_aggregator_stop; gstaggregator_class->flush = gst_audio_aggregator_flush; gstaggregator_class->aggregate = GST_DEBUG_FUNCPTR (gst_audio_aggregator_aggregate); gstaggregator_class->clip = GST_DEBUG_FUNCPTR (gst_audio_aggregator_do_clip); gstaggregator_class->get_next_time = gst_audio_aggregator_get_next_time; 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)); g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD, g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold", "Timestamp alignment threshold in nanoseconds", 0, G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DISCONT_WAIT, g_param_spec_uint64 ("discont-wait", "Discont Wait", "Window of time in nanoseconds to wait before " "creating a discontinuity", 0, G_MAXUINT64 - 1, DEFAULT_DISCONT_WAIT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); } static void gst_audio_aggregator_init (GstAudioAggregator * aagg) { aagg->priv = G_TYPE_INSTANCE_GET_PRIVATE (aagg, GST_TYPE_AUDIO_AGGREGATOR, GstAudioAggregatorPrivate); g_mutex_init (&aagg->priv->mutex); aagg->priv->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION; aagg->priv->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD; aagg->priv->discont_wait = DEFAULT_DISCONT_WAIT; aagg->current_caps = NULL; gst_audio_info_init (&aagg->info); gst_aggregator_set_latency (GST_AGGREGATOR (aagg), aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration); } static void gst_audio_aggregator_dispose (GObject * object) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object); gst_caps_replace (&aagg->current_caps, NULL); g_mutex_clear (&aagg->priv->mutex); G_OBJECT_CLASS (gst_audio_aggregator_parent_class)->dispose (object); } static void gst_audio_aggregator_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object); switch (prop_id) { case PROP_OUTPUT_BUFFER_DURATION: aagg->priv->output_buffer_duration = g_value_get_uint64 (value); gst_aggregator_set_latency (GST_AGGREGATOR (aagg), aagg->priv->output_buffer_duration, aagg->priv->output_buffer_duration); break; case PROP_ALIGNMENT_THRESHOLD: aagg->priv->alignment_threshold = g_value_get_uint64 (value); break; case PROP_DISCONT_WAIT: aagg->priv->discont_wait = g_value_get_uint64 (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_audio_aggregator_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (object); switch (prop_id) { case PROP_OUTPUT_BUFFER_DURATION: g_value_set_uint64 (value, aagg->priv->output_buffer_duration); break; case PROP_ALIGNMENT_THRESHOLD: g_value_set_uint64 (value, aagg->priv->alignment_threshold); break; case PROP_DISCONT_WAIT: g_value_set_uint64 (value, aagg->priv->discont_wait); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } /* 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 parametters before linking up */ if ((start_type != GST_SEEK_TYPE_NONE) && (start_type != GST_SEEK_TYPE_SET)) { result = FALSE; GST_DEBUG_OBJECT (aagg, "seeking failed, unhandled seek type for start: %d", start_type); goto done; } if ((stop_type != GST_SEEK_TYPE_NONE) && (stop_type != GST_SEEK_TYPE_SET)) { result = FALSE; GST_DEBUG_OBJECT (aagg, "seeking failed, unhandled seek type for end: %d", stop_type); goto done; } GST_OBJECT_LOCK (agg); dest_format = agg->segment.format; GST_OBJECT_UNLOCK (agg); if (seek_format != dest_format) { result = FALSE; GST_DEBUG_OBJECT (aagg, "seeking failed, unhandled seek format: %s", gst_format_get_name (seek_format)); goto done; } } break; default: break; } return GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_event (agg, event); done: return result; } static gboolean gst_audio_aggregator_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad, GstEvent * event) { gboolean res = TRUE; GST_DEBUG_OBJECT (aggpad, "Got %s event on sink pad", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEGMENT: { const GstSegment *segment; gst_event_parse_segment (event, &segment); if (segment->format != GST_FORMAT_TIME) { GST_ERROR_OBJECT (agg, "Segment of type %s are not supported," " only TIME segments are supported", gst_format_get_name (segment->format)); gst_event_unref (event); event = NULL; res = FALSE; break; } GST_OBJECT_LOCK (agg); if (segment->rate != agg->segment.rate) { GST_ERROR_OBJECT (aggpad, "Got segment event with wrong rate %lf, expected %lf", segment->rate, agg->segment.rate); res = FALSE; gst_event_unref (event); event = NULL; } else if (segment->rate < 0.0) { GST_ERROR_OBJECT (aggpad, "Negative rates not supported yet"); res = FALSE; gst_event_unref (event); event = NULL; } else { GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (aggpad); GST_OBJECT_LOCK (pad); pad->priv->new_segment = TRUE; GST_OBJECT_UNLOCK (pad); } GST_OBJECT_UNLOCK (agg); break; } default: break; } if (event != NULL) return GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->sink_event (agg, aggpad, event); return res; } /* FIXME, the duration query should reflect how long you will produce * data, that is the amount of stream time until you will emit EOS. * * For synchronized mixing this is always the max of all the durations * of upstream since we emit EOS when all of them finished. * * We don't do synchronized mixing so this really depends on where the * streams where punched in and what their relative offsets are against * eachother which we can get from the first timestamps we see. * * When we add a new stream (or remove a stream) the duration might * also become invalid again and we need to post a new DURATION * message to notify this fact to the parent. * For now we take the max of all the upstream elements so the simple * cases work at least somewhat. */ static gboolean gst_audio_aggregator_query_duration (GstAudioAggregator * aagg, GstQuery * query) { gint64 max; gboolean res; GstFormat format; GstIterator *it; gboolean done; GValue item = { 0, }; /* parse format */ gst_query_parse_duration (query, &format, NULL); max = -1; res = TRUE; done = FALSE; it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (aagg)); while (!done) { GstIteratorResult ires; ires = gst_iterator_next (it, &item); switch (ires) { case GST_ITERATOR_DONE: done = TRUE; break; case GST_ITERATOR_OK: { GstPad *pad = g_value_get_object (&item); gint64 duration; /* ask sink peer for duration */ res &= gst_pad_peer_query_duration (pad, format, &duration); /* take max from all valid return values */ if (res) { /* valid unknown length, stop searching */ if (duration == -1) { max = duration; done = TRUE; } /* else see if bigger than current max */ else if (duration > max) max = duration; } g_value_reset (&item); break; } case GST_ITERATOR_RESYNC: max = -1; res = TRUE; gst_iterator_resync (it); break; default: res = FALSE; done = TRUE; break; } } g_value_unset (&item); gst_iterator_free (it); if (res) { /* and store the max */ GST_DEBUG_OBJECT (aagg, "Total duration in format %s: %" GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max)); gst_query_set_duration (query, format, max); } return res; } static gboolean gst_audio_aggregator_src_query (GstAggregator * agg, GstQuery * query) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); gboolean res = FALSE; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_DURATION: res = gst_audio_aggregator_query_duration (aagg, query); break; case GST_QUERY_POSITION: { GstFormat format; gst_query_parse_position (query, &format, NULL); GST_OBJECT_LOCK (aagg); switch (format) { case GST_FORMAT_TIME: /* FIXME, bring to stream time, might be tricky */ gst_query_set_position (query, format, agg->segment.position); res = TRUE; break; case GST_FORMAT_BYTES: if (GST_AUDIO_INFO_BPF (&aagg->info)) { gst_query_set_position (query, format, aagg->priv->offset * GST_AUDIO_INFO_BPF (&aagg->info)); res = TRUE; } break; case GST_FORMAT_DEFAULT: gst_query_set_position (query, format, aagg->priv->offset); res = TRUE; break; default: break; } GST_OBJECT_UNLOCK (aagg); break; } default: res = GST_AGGREGATOR_CLASS (gst_audio_aggregator_parent_class)->src_query (agg, query); break; } return res; } void gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad, GstCaps * caps) { gboolean valid; GST_OBJECT_LOCK (pad); valid = gst_audio_info_from_caps (&pad->info, caps); GST_OBJECT_UNLOCK (pad); g_assert (valid); } gboolean gst_audio_aggregator_set_src_caps (GstAudioAggregator * aagg, GstCaps * caps) { GstAudioInfo info; 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, &aagg->info)) { GST_INFO_OBJECT (aagg, "setting caps to %" GST_PTR_FORMAT, caps); gst_caps_replace (&aagg->current_caps, caps); memcpy (&aagg->info, &info, sizeof (info)); aagg->priv->send_caps = TRUE; } GST_OBJECT_UNLOCK (aagg); GST_AUDIO_AGGREGATOR_UNLOCK (aagg); /* send caps event later, after stream-start event */ return TRUE; } /* Must hold object lock and aagg lock to call */ static void gst_audio_aggregator_reset (GstAudioAggregator * aagg) { GstAggregator *agg = GST_AGGREGATOR (aagg); GST_AUDIO_AGGREGATOR_LOCK (aagg); GST_OBJECT_LOCK (aagg); agg->segment.position = -1; aagg->priv->offset = 0; gst_audio_info_init (&aagg->info); gst_caps_replace (&aagg->current_caps, NULL); gst_buffer_replace (&aagg->priv->current_buffer, NULL); GST_OBJECT_UNLOCK (aagg); GST_AUDIO_AGGREGATOR_UNLOCK (aagg); } static gboolean gst_audio_aggregator_start (GstAggregator * agg) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); gst_audio_aggregator_reset (aagg); return TRUE; } static gboolean gst_audio_aggregator_stop (GstAggregator * agg) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); gst_audio_aggregator_reset (aagg); return TRUE; } static GstFlowReturn gst_audio_aggregator_flush (GstAggregator * agg) { GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (agg); GST_AUDIO_AGGREGATOR_LOCK (aagg); GST_OBJECT_LOCK (aagg); agg->segment.position = -1; aagg->priv->offset = 0; gst_buffer_replace (&aagg->priv->current_buffer, NULL); GST_OBJECT_UNLOCK (aagg); GST_AUDIO_AGGREGATOR_UNLOCK (aagg); return GST_FLOW_OK; } static GstFlowReturn gst_audio_aggregator_do_clip (GstAggregator * agg, GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** out) { GstAudioAggregatorPad *pad = GST_AUDIO_AGGREGATOR_PAD (bpad); gint rate, bpf; rate = GST_AUDIO_INFO_RATE (&pad->info); bpf = GST_AUDIO_INFO_BPF (&pad->info); GST_OBJECT_LOCK (bpad); *out = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf); GST_OBJECT_UNLOCK (bpad); return GST_FLOW_OK; } /* Called with the object lock for both the element and pad held, * as well as the aagg lock */ static gboolean gst_audio_aggregator_fill_buffer (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad, GstBuffer * inbuf) { GstClockTime start_time, end_time; gboolean discont = FALSE; guint64 start_offset, end_offset; gint rate, bpf; GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad); g_assert (pad->priv->buffer == NULL); 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 (inbuf) / bpf; if (!GST_BUFFER_PTS_IS_VALID (inbuf)) { 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 (inbuf); end_time = start_time + gst_util_uint64_scale_ceil (pad->priv->size, GST_SECOND, rate); start_offset = gst_util_uint64_scale (start_time, rate, GST_SECOND); end_offset = start_offset + pad->priv->size; if (GST_BUFFER_IS_DISCONT (inbuf) || GST_BUFFER_FLAG_IS_SET (inbuf, 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_INFO_OBJECT (pad, "Have discont. Expected %" G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT, pad->priv->next_offset, start_offset); pad->priv->output_offset = -1; pad->priv->next_offset = end_offset; } else { pad->priv->next_offset += pad->priv->size; } if (pad->priv->output_offset == -1) { GstClockTime start_running_time; GstClockTime end_running_time; guint64 start_running_time_offset; guint64 end_running_time_offset; 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); start_running_time_offset = gst_util_uint64_scale (start_running_time, rate, GST_SECOND); end_running_time_offset = gst_util_uint64_scale (end_running_time, rate, GST_SECOND); if (end_running_time_offset < aagg->priv->offset) { /* Before output segment, drop */ gst_buffer_unref (inbuf); pad->priv->buffer = NULL; pad->priv->position = 0; pad->priv->size = 0; pad->priv->output_offset = -1; GST_DEBUG_OBJECT (pad, "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %" G_GUINT64_FORMAT, end_running_time_offset, aagg->priv->offset); return FALSE; } if (start_running_time_offset < aagg->priv->offset) { guint diff = aagg->priv->offset - start_running_time_offset; pad->priv->position += diff; if (pad->priv->position >= pad->priv->size) { /* Empty buffer, drop */ gst_buffer_unref (inbuf); pad->priv->buffer = NULL; pad->priv->position = 0; pad->priv->size = 0; pad->priv->output_offset = -1; GST_DEBUG_OBJECT (pad, "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %" G_GUINT64_FORMAT, end_running_time_offset, aagg->priv->offset); return FALSE; } } pad->priv->output_offset = MAX (start_running_time_offset, aagg->priv->offset); GST_DEBUG_OBJECT (pad, "Buffer resynced: Pad offset %" G_GUINT64_FORMAT ", current audio aggregator offset %" G_GUINT64_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); pad->priv->buffer = inbuf; return TRUE; } /* Called with pad object lock held */ static gboolean gst_audio_aggregator_mix_buffer (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad, GstBuffer * inbuf, GstBuffer * outbuf) { guint overlap; guint out_start; gboolean filled; guint blocksize; blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration, GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND); blocksize = MAX (1, blocksize); /* 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->size; gst_buffer_replace (&pad->priv->buffer, NULL); return FALSE; } filled = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->aggregate_one_buffer (aagg, pad, inbuf, pad->priv->position, outbuf, out_start, overlap); if (filled) GST_BUFFER_FLAG_UNSET (outbuf, GST_BUFFER_FLAG_GAP); 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_DEBUG_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) { GstBuffer *outbuf = gst_buffer_new_allocate (NULL, num_frames * GST_AUDIO_INFO_BPF (&aagg->info), NULL); GstMapInfo outmap; gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE); gst_audio_format_fill_silence (aagg->info.finfo, outmap.data, outmap.size); gst_buffer_unmap (outbuf, &outmap); return outbuf; } static gboolean sync_pad_values (GstAudioAggregator * aagg, GstAudioAggregatorPad * pad) { GstAggregatorPad *bpad = GST_AGGREGATOR_PAD (pad); GstClockTime timestamp, stream_time; if (pad->priv->buffer == NULL) return TRUE; timestamp = GST_BUFFER_PTS (pad->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 (pad), stream_time); return TRUE; } static GstFlowReturn gst_audio_aggregator_aggregate (GstAggregator * agg, gboolean timeout) { /* Get all pads that have data for us and store them in a * new list. * * Calculate the current output offset/timestamp and * offset_end/timestamp_end. Allocate a silence buffer * for this and store it. * * For all pads: * 1) Once per input buffer (cached) * 1) Check discont (flag and timestamp with tolerance) * 2) If discont or new, resync. That means: * 1) Drop all start data of the buffer that comes before * the current position/offset. * 2) Calculate the offset (output segment!) that the first * frame of the input buffer corresponds to. Base this on * the running time. * * 2) If the current pad's offset/offset_end overlaps with the output * offset/offset_end, mix it at the appropiate position in the output * buffer and advance the pad's position. Remember if this pad needs * a new buffer to advance behind the output offset_end. * * 3) If we had no pad with a buffer, go EOS. * * 4) If we had at least one pad that did not advance behind output * offset_end, let collected be called again for the current * output offset/offset_end. */ GstElement *element; GstAudioAggregator *aagg; GList *iter; GstFlowReturn ret; GstBuffer *outbuf = NULL; gint64 next_offset; gint64 next_timestamp; gint rate, bpf; gboolean dropped = FALSE; gboolean is_eos = TRUE; gboolean is_done = TRUE; guint blocksize; element = GST_ELEMENT (agg); aagg = GST_AUDIO_AGGREGATOR (agg); /* Sync pad properties to the stream time */ gst_aggregator_iterate_sinkpads (agg, (GstAggregatorPadForeachFunc) GST_DEBUG_FUNCPTR (sync_pad_values), NULL); GST_AUDIO_AGGREGATOR_LOCK (aagg); GST_OBJECT_LOCK (agg); /* Update position from the segment start/stop if needed */ if (agg->segment.position == -1) { if (agg->segment.rate > 0.0) agg->segment.position = agg->segment.start; else agg->segment.position = agg->segment.stop; } if (G_UNLIKELY (aagg->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) { if (timeout) { GST_DEBUG_OBJECT (aagg, "Got timeout before receiving any caps, don't output anything"); /* Advance position */ if (agg->segment.rate > 0.0) agg->segment.position += aagg->priv->output_buffer_duration; else if (agg->segment.position > aagg->priv->output_buffer_duration) agg->segment.position -= aagg->priv->output_buffer_duration; else agg->segment.position = 0; GST_OBJECT_UNLOCK (agg); GST_AUDIO_AGGREGATOR_UNLOCK (aagg); return GST_FLOW_OK; } else { GST_OBJECT_UNLOCK (agg); goto not_negotiated; } } if (aagg->priv->send_caps) { GST_OBJECT_UNLOCK (agg); gst_aggregator_set_src_caps (agg, aagg->current_caps); GST_OBJECT_LOCK (agg); aagg->priv->offset = gst_util_uint64_scale (agg->segment.position, GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND); aagg->priv->send_caps = FALSE; } rate = GST_AUDIO_INFO_RATE (&aagg->info); bpf = GST_AUDIO_INFO_BPF (&aagg->info); blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration, GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND); blocksize = MAX (1, blocksize); /* for the next timestamp, use the sample counter, which will * never accumulate rounding errors */ /* 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; } next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate); if (aagg->priv->current_buffer == NULL) { GST_OBJECT_UNLOCK (agg); aagg->priv->current_buffer = GST_AUDIO_AGGREGATOR_GET_CLASS (aagg)->create_output_buffer (aagg, blocksize); /* Be careful, some things could have changed ? */ GST_OBJECT_LOCK (agg); GST_BUFFER_FLAG_SET (aagg->priv->current_buffer, GST_BUFFER_FLAG_GAP); } outbuf = aagg->priv->current_buffer; GST_LOG_OBJECT (agg, "Starting to mix %u samples for offset %" G_GUINT64_FORMAT " with timestamp %" GST_TIME_FORMAT, blocksize, aagg->priv->offset, GST_TIME_ARGS (agg->segment.position)); for (iter = element->sinkpads; iter; iter = iter->next) { GstBuffer *inbuf; GstAudioAggregatorPad *pad = (GstAudioAggregatorPad *) iter->data; GstAggregatorPad *aggpad = (GstAggregatorPad *) iter->data; gboolean drop_buf = FALSE; gboolean pad_eos = gst_aggregator_pad_is_eos (aggpad); if (!pad_eos) is_eos = FALSE; inbuf = gst_aggregator_pad_get_buffer (aggpad); GST_OBJECT_LOCK (pad); if (!inbuf) { if (timeout) { if (pad->priv->output_offset < next_offset) { gint64 diff = next_offset - pad->priv->output_offset; GST_LOG_OBJECT (pad, "Timeout, missing %" G_GINT64_FORMAT " frames (%" GST_TIME_FORMAT ")", diff, GST_TIME_ARGS (gst_util_uint64_scale (diff, GST_SECOND, GST_AUDIO_INFO_RATE (&aagg->info)))); } } else if (!pad_eos) { is_done = FALSE; } GST_OBJECT_UNLOCK (pad); continue; } g_assert (!pad->priv->buffer || pad->priv->buffer == inbuf); /* New buffer? */ if (!pad->priv->buffer) { /* Takes ownership of buffer */ if (!gst_audio_aggregator_fill_buffer (aagg, pad, inbuf)) { dropped = TRUE; GST_OBJECT_UNLOCK (pad); gst_aggregator_pad_drop_buffer (aggpad); continue; } } else { gst_buffer_unref (inbuf); } 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 lacking behind, we need to update the offset * and maybe drop the current buffer */ if (pad->priv->output_offset < aagg->priv->offset) { gint64 diff = aagg->priv->offset - pad->priv->output_offset; gint64 odiff = diff; if (pad->priv->position + diff > pad->priv->size) diff = pad->priv->size - pad->priv->position; pad->priv->position += diff; pad->priv->output_offset += diff; if (pad->priv->position == pad->priv->size) { GST_LOG_OBJECT (pad, "Buffer was late by %" GST_TIME_FORMAT ", dropping %" GST_PTR_FORMAT, GST_TIME_ARGS (gst_util_uint64_scale (odiff, GST_SECOND, GST_AUDIO_INFO_RATE (&aagg->info))), pad->priv->buffer); /* Buffer done, drop it */ gst_buffer_replace (&pad->priv->buffer, NULL); dropped = TRUE; GST_OBJECT_UNLOCK (pad); gst_aggregator_pad_drop_buffer (aggpad); continue; } } if (pad->priv->output_offset >= aagg->priv->offset && pad->priv->output_offset < aagg->priv->offset + blocksize && pad->priv->buffer) { GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset"); drop_buf = !gst_audio_aggregator_mix_buffer (aagg, pad, pad->priv->buffer, outbuf); if (pad->priv->output_offset >= next_offset) { GST_DEBUG_OBJECT (pad, "Pad is after current offset: %" G_GUINT64_FORMAT " >= %" G_GUINT64_FORMAT, pad->priv->output_offset, next_offset); } else { is_done = FALSE; } } GST_OBJECT_UNLOCK (pad); if (drop_buf) gst_aggregator_pad_drop_buffer (aggpad); } GST_OBJECT_UNLOCK (agg); if (dropped) { /* We dropped a buffer, retry */ GST_INFO_OBJECT (aagg, "A pad dropped a buffer, wait for the next one"); GST_AUDIO_AGGREGATOR_UNLOCK (aagg); return GST_FLOW_OK; } if (!is_done && !is_eos) { /* Get more buffers */ GST_INFO_OBJECT (aagg, "We're not done yet for the current offset," " waiting for more data"); GST_AUDIO_AGGREGATOR_UNLOCK (aagg); return GST_FLOW_OK; } 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_GUINT64_FORMAT, max_offset, next_offset); next_offset = max_offset; next_timestamp = 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; } aagg->priv->offset = next_offset; 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, aagg->priv->current_buffer); aagg->priv->current_buffer = NULL; GST_LOG_OBJECT (aagg, "pushed outbuf, result = %s", gst_flow_get_name (ret)); 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; } }