include $(top_srcdir)/common/orc.mak
-libgstaudiomixer_la_SOURCES = gstaudiomixer.c
+libgstaudiomixer_la_SOURCES = gstaudiomixer.c gstaudioaggregator.c
nodist_libgstaudiomixer_la_SOURCES = $(ORC_NODIST_SOURCES)
libgstaudiomixer_la_CFLAGS = \
-I$(top_srcdir)/gst-libs \
$(GST_BASE_LIBS) $(GST_LIBS) $(ORC_LIBS)
libgstaudiomixer_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
-noinst_HEADERS = gstaudiomixer.h
+noinst_HEADERS = gstaudiomixer.h gstaudioaggregator.h
--- /dev/null
+/* GStreamer
+ * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
+ * 2001 Thomas <thomas@apestaart.org>
+ * 2005,2006 Wim Taymans <wim@fluendo.com>
+ * 2013 Sebastian Dröge <sebastian@centricular.com>
+ * 2014 Collabora
+ * Olivier Crete <olivier.crete@collabora.com>
+ *
+ * gstaudioaggregator.c:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION: gstaudioaggregator
+ * @short_description: manages a set of pads with the purpose of
+ * aggregating their buffers for raw audio
+ * @see_also: #GstAggregator
+ *
+ */
+
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include "gstaudioaggregator.h"
+
+#include <string.h>
+
+GST_DEBUG_CATEGORY_STATIC (audio_aggregator_debug);
+#define GST_CAT_DEFAULT audio_aggregator_debug
+
+struct _GstAudioAggregatorPadPrivate
+{
+ /* All members are protected by the pad object lock */
+
+ GstBuffer *buffer; /* current buffer we're mixing,
+ for comparison with 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)
+{
+ GST_OBJECT_LOCK (pad);
+ gst_audio_info_from_caps (&pad->info, caps);
+ GST_OBJECT_UNLOCK (pad);
+}
+
+
+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);
+
+ 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;
+ GstClockTime timestamp, stream_time = GST_CLOCK_TIME_NONE;
+ 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;
+ }
+
+ timestamp = GST_BUFFER_PTS (inbuf);
+ stream_time = gst_segment_to_stream_time (&aggpad->segment, GST_FORMAT_TIME,
+ timestamp);
+
+ /* 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);
+
+ 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_and_alloc (num_frames *
+ GST_AUDIO_INFO_BPF (&aagg->info));
+ 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 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);
+
+ blocksize = gst_util_uint64_scale (aagg->priv->output_buffer_duration,
+ GST_AUDIO_INFO_RATE (&aagg->info), GST_SECOND);
+ blocksize = MAX (1, blocksize);
+
+ 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);
+
+
+ /* 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;
+
+ 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) {
+ /* 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;
+ }
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) 2014 Collabora
+ * Author: Olivier Crete <olivier.crete@collabora.com>
+ *
+ * gstaudioaggregator.h:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_AUDIO_AGGREGATOR_H__
+#define __GST_AUDIO_AGGREGATOR_H__
+
+#ifndef GST_USE_UNSTABLE_API
+#warning "The Base library from gst-plugins-bad is unstable API and may change in future."
+#warning "You can define GST_USE_UNSTABLE_API to avoid this warning."
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstaggregator.h>
+#include <gst/audio/audio.h>
+
+G_BEGIN_DECLS
+
+/*******************************
+ * GstAudioAggregator Structs *
+ *******************************/
+
+typedef struct _GstAudioAggregator GstAudioAggregator;
+typedef struct _GstAudioAggregatorPrivate GstAudioAggregatorPrivate;
+typedef struct _GstAudioAggregatorClass GstAudioAggregatorClass;
+
+
+/************************
+ * GstAudioAggregatorPad API *
+ ***********************/
+
+#define GST_TYPE_AUDIO_AGGREGATOR_PAD (gst_audio_aggregator_pad_get_type())
+#define GST_AUDIO_AGGREGATOR_PAD(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPad))
+#define GST_AUDIO_AGGREGATOR_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadClass))
+#define GST_AUDIO_AGGREGATOR_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD, GstAudioAggregatorPadClass))
+#define GST_IS_AUDIO_AGGREGATOR_PAD(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR_PAD))
+#define GST_IS_AUDIO_AGGREGATOR_PAD_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR_PAD))
+
+/****************************
+ * GstAudioAggregatorPad Structs *
+ ***************************/
+
+typedef struct _GstAudioAggregatorPad GstAudioAggregatorPad;
+typedef struct _GstAudioAggregatorPadClass GstAudioAggregatorPadClass;
+typedef struct _GstAudioAggregatorPadPrivate GstAudioAggregatorPadPrivate;
+
+/**
+ * GstAudioAggregatorPad:
+ * @parent: The parent #GstAggregatorPad
+ * @info: The audio info for this pad set from the incoming caps
+ *
+ * The implementation the GstPad to use with #GstAudioAggregator
+ */
+struct _GstAudioAggregatorPad
+{
+ GstAggregatorPad parent;
+
+ GstAudioInfo info;
+
+ /*< private >*/
+ GstAudioAggregatorPadPrivate * priv;
+
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstAudioAggregatorPadClass:
+ *
+ */
+struct _GstAudioAggregatorPadClass
+{
+ GstAggregatorPadClass parent_class;
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+GType gst_audio_aggregator_pad_get_type (void);
+
+/**************************
+ * GstAudioAggregator API *
+ **************************/
+
+#define GST_TYPE_AUDIO_AGGREGATOR (gst_audio_aggregator_get_type())
+#define GST_AUDIO_AGGREGATOR(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregator))
+#define GST_AUDIO_AGGREGATOR_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregatorClass))
+#define GST_AUDIO_AGGREGATOR_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_AUDIO_AGGREGATOR,GstAudioAggregatorClass))
+#define GST_IS_AUDIO_AGGREGATOR(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_AGGREGATOR))
+#define GST_IS_AUDIO_AGGREGATOR_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIO_AGGREGATOR))
+
+#define GST_FLOW_CUSTOM_SUCCESS GST_FLOW_NOT_HANDLED
+
+/**
+ * GstAudioAggregator:
+ * @parent: The parent #GstAggregator
+ * @info: The information parsed from the current caps
+ * @current_caps: The caps set by the subclass
+ *
+ * GstAudioAggregator object
+ */
+struct _GstAudioAggregator
+{
+ GstAggregator parent;
+
+ /* All member are read only for subclasses, must hold OBJECT lock */
+ GstAudioInfo info;
+
+ GstCaps *current_caps;
+
+ /*< private >*/
+ GstAudioAggregatorPrivate *priv;
+
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstAudioAggregatorClass:
+ * @create_output_buffer: Create a new output buffer contains num_frames frames.
+ * @aggregate_one_buffer: Aggregates one input buffer to the output
+ * buffer. The in_offset and out_offset are in "frames", which is
+ * the size of a sample times the number of channels. Returns TRUE if
+ * any non-silence was added to the buffer
+ */
+struct _GstAudioAggregatorClass {
+ GstAggregatorClass parent_class;
+
+ GstBuffer * (* create_output_buffer) (GstAudioAggregator * aagg,
+ guint num_frames);
+ gboolean (* aggregate_one_buffer) (GstAudioAggregator * aagg,
+ GstAudioAggregatorPad * pad, GstBuffer * inbuf, guint in_offset,
+ GstBuffer * outbuf, guint out_offset, guint num_frames);
+
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING];
+};
+
+/*************************
+ * GstAggregator methods *
+ ************************/
+
+GType gst_audio_aggregator_get_type(void);
+
+void
+gst_audio_aggregator_set_sink_caps (GstAudioAggregator * aagg,
+ GstAudioAggregatorPad * pad, GstCaps * caps);
+
+gboolean
+gst_audio_aggregator_set_src_caps (GstAudioAggregator * aagg, GstCaps * caps);
+
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_AGGREGATOR_H__ */
PROP_PAD_MUTE
};
-G_DEFINE_TYPE (GstAudioMixerPad, gst_audiomixer_pad, GST_TYPE_AGGREGATOR_PAD);
+G_DEFINE_TYPE (GstAudioMixerPad, gst_audiomixer_pad,
+ GST_TYPE_AUDIO_AGGREGATOR_PAD);
static void
gst_audiomixer_pad_get_property (GObject * object, guint prop_id,
}
}
-static gboolean
-gst_audiomixer_pad_flush_pad (GstAggregatorPad * aggpad,
- GstAggregator * aggregator)
-{
- GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aggpad);
-
- GST_OBJECT_LOCK (aggpad);
- pad->position = pad->size = 0;
- pad->output_offset = pad->next_offset = -1;
- pad->discont_time = GST_CLOCK_TIME_NONE;
- gst_buffer_replace (&pad->buffer, NULL);
- GST_OBJECT_UNLOCK (aggpad);
-
- return TRUE;
-}
-
static void
gst_audiomixer_pad_class_init (GstAudioMixerPadClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
- GstAggregatorPadClass *aggpadclass = (GstAggregatorPadClass *) klass;
gobject_class->set_property = gst_audiomixer_pad_set_property;
gobject_class->get_property = gst_audiomixer_pad_get_property;
g_param_spec_boolean ("mute", "Mute", "Mute this pad",
DEFAULT_PAD_MUTE,
G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
-
- aggpadclass->flush = GST_DEBUG_FUNCPTR (gst_audiomixer_pad_flush_pad);
}
static void
{
pad->volume = DEFAULT_PAD_VOLUME;
pad->mute = DEFAULT_PAD_MUTE;
-
- pad->buffer = NULL;
- pad->position = 0;
- pad->size = 0;
- pad->output_offset = -1;
- pad->next_offset = -1;
- pad->discont_time = GST_CLOCK_TIME_NONE;
}
-#define DEFAULT_ALIGNMENT_THRESHOLD (40 * GST_MSECOND)
-#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
-#define DEFAULT_OUTPUT_BUFFER_DURATION (10 * GST_MSECOND)
-
enum
{
PROP_0,
- PROP_FILTER_CAPS,
- PROP_ALIGNMENT_THRESHOLD,
- PROP_DISCONT_WAIT,
- PROP_OUTPUT_BUFFER_DURATION
+ PROP_FILTER_CAPS
};
/* elementfactory information */
gpointer iface_data);
#define gst_audiomixer_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstAudioMixer, gst_audiomixer, GST_TYPE_AGGREGATOR,
- G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
+G_DEFINE_TYPE_WITH_CODE (GstAudioMixer, gst_audiomixer,
+ GST_TYPE_AUDIO_AGGREGATOR, G_IMPLEMENT_INTERFACE (GST_TYPE_CHILD_PROXY,
gst_audiomixer_child_proxy_init));
static void gst_audiomixer_dispose (GObject * object);
GstPadTemplate * temp, const gchar * req_name, const GstCaps * caps);
static void gst_audiomixer_release_pad (GstElement * element, GstPad * pad);
-static GstFlowReturn
-gst_audiomixer_do_clip (GstAggregator * agg,
- GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** outbuf);
-static GstFlowReturn gst_audiomixer_aggregate (GstAggregator * agg,
- gboolean timeout);
+static gboolean
+gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg,
+ GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset,
+ GstBuffer * outbuf, guint out_offset, guint num_samples);
-static GstClockTime
-gst_audiomixer_get_next_time (GstAggregator * agg)
-{
- if (agg->segment.position == -1)
- return agg->segment.start;
- else
- return agg->segment.position;
-}
/* we can only accept caps that we and downstream can handle.
* if we have filtercaps set, use those to constrain the target caps.
*/
static GstCaps *
-gst_audiomixer_sink_getcaps (GstPad * pad, GstCaps * filter)
+gst_audiomixer_sink_getcaps (GstAggregator * agg, GstPad * pad,
+ GstCaps * filter)
{
- GstAggregator *agg;
+ GstAudioAggregator *aagg;
GstAudioMixer *audiomixer;
GstCaps *result, *peercaps, *current_caps, *filter_caps;
GstStructure *s;
gint i, n;
- audiomixer = GST_AUDIO_MIXER (GST_PAD_PARENT (pad));
- agg = GST_AGGREGATOR (audiomixer);
+ audiomixer = GST_AUDIO_MIXER (agg);
+ aagg = GST_AUDIO_AGGREGATOR (agg);
GST_OBJECT_LOCK (audiomixer);
/* take filter */
/* get the allowed caps on this sinkpad */
GST_OBJECT_LOCK (audiomixer);
- current_caps =
- audiomixer->current_caps ? gst_caps_ref (audiomixer->current_caps) : NULL;
+ current_caps = aagg->current_caps ? gst_caps_ref (aagg->current_caps) : NULL;
if (current_caps == NULL) {
current_caps = gst_pad_get_pad_template_caps (pad);
if (!current_caps)
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
- caps = gst_audiomixer_sink_getcaps (GST_PAD (aggpad), filter);
+ caps = gst_audiomixer_sink_getcaps (agg, GST_PAD (aggpad), filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
gst_audiomixer_setcaps (GstAudioMixer * audiomixer, GstPad * pad,
GstCaps * orig_caps)
{
+ GstAudioAggregator *aagg = GST_AUDIO_AGGREGATOR (audiomixer);
GstCaps *caps;
GstAudioInfo info;
GstStructure *s;
gint channels;
+ gboolean ret;
caps = gst_caps_copy (orig_caps);
* different upstream threads doing query_caps + accept_caps + sending
* (possibly different) CAPS events, but there's not much we can do about
* that, upstream needs to deal with it. */
- if (audiomixer->current_caps != NULL) {
- if (gst_audio_info_is_equal (&info, &audiomixer->info)) {
+ if (aagg->current_caps != NULL) {
+ if (gst_audio_info_is_equal (&info, &aagg->info)) {
GST_OBJECT_UNLOCK (audiomixer);
gst_caps_unref (caps);
+ gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
+ orig_caps);
return TRUE;
} else {
GST_DEBUG_OBJECT (pad, "got input caps %" GST_PTR_FORMAT ", but "
- "current caps are %" GST_PTR_FORMAT, caps, audiomixer->current_caps);
+ "current caps are %" GST_PTR_FORMAT, caps, aagg->current_caps);
GST_OBJECT_UNLOCK (audiomixer);
gst_pad_push_event (pad, gst_event_new_reconfigure ());
gst_caps_unref (caps);
return FALSE;
}
}
+ GST_OBJECT_UNLOCK (audiomixer);
- GST_INFO_OBJECT (pad, "setting caps to %" GST_PTR_FORMAT, caps);
- gst_caps_replace (&audiomixer->current_caps, caps);
+ ret = gst_audio_aggregator_set_src_caps (aagg, caps);
- memcpy (&audiomixer->info, &info, sizeof (info));
- audiomixer->send_caps = TRUE;
- GST_OBJECT_UNLOCK (audiomixer);
- /* send caps event later, after stream-start event */
+ if (ret)
+ gst_audio_aggregator_set_sink_caps (aagg, GST_AUDIO_AGGREGATOR_PAD (pad),
+ orig_caps);
GST_INFO_OBJECT (pad, "handle caps change to %" GST_PTR_FORMAT, caps);
gst_caps_unref (caps);
- return TRUE;
+ return ret;
/* ERRORS */
invalid_format:
}
}
-/* 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_audiomixer_query_duration (GstAudioMixer * audiomixer, 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 (audiomixer));
- 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 (audiomixer, "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_audiomixer_src_query (GstAggregator * agg, GstQuery * query)
-{
- GstAudioMixer *audiomixer = GST_AUDIO_MIXER (agg);
- gboolean res = FALSE;
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_POSITION:
- {
- GstFormat format;
-
- gst_query_parse_position (query, &format, NULL);
-
- 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_DEFAULT:
- gst_query_set_position (query, format, audiomixer->offset);
- res = TRUE;
- break;
- default:
- break;
- }
- break;
- }
- case GST_QUERY_DURATION:
- res = gst_audiomixer_query_duration (audiomixer, query);
- break;
- default:
- res =
- GST_AGGREGATOR_CLASS (gst_audiomixer_parent_class)->src_query
- (agg, query);
- break;
- }
-
- return res;
-}
-
-/* event handling */
-
-typedef struct
-{
- GstEvent *event;
- gboolean flush;
-} EventData;
-
-static gboolean
-gst_audiomixer_src_event (GstAggregator * agg, GstEvent * event)
-{
- gboolean result;
-
- GstAudioMixer *audiomixer = GST_AUDIO_MIXER (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 (audiomixer,
- "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 (audiomixer,
- "seeking failed, unhandled seek type for end: %d", stop_type);
- goto done;
- }
-
- dest_format = agg->segment.format;
- if (seek_format != dest_format) {
- result = FALSE;
- GST_DEBUG_OBJECT (audiomixer,
- "seeking failed, unhandled seek format: %d", seek_format);
- goto done;
- }
-
- /* Link up */
- result = GST_AGGREGATOR_CLASS (parent_class)->src_event (agg, event);
- goto done;
- }
- break;
- default:
- break;
- }
-
- return GST_AGGREGATOR_CLASS (parent_class)->src_event (agg, event);
-
-done:
- return result;
-}
-
static gboolean
gst_audiomixer_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
GstEvent * event)
event = NULL;
break;
}
- case GST_EVENT_SEGMENT:
- {
- const GstSegment *segment;
-
- gst_event_parse_segment (event, &segment);
- 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 {
- GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aggpad);
-
- /* Ideally, this should only be set when the new segment causes running
- * times to change, and hence needs discont calculation in fill_buffer */
- GST_OBJECT_LOCK (pad);
- pad->new_segment = TRUE;
- GST_OBJECT_UNLOCK (pad);
- }
- break;
- }
default:
break;
}
}
static void
-gst_audiomixer_reset (GstAudioMixer * audiomixer)
-{
- GstAggregator *agg = GST_AGGREGATOR (audiomixer);
-
- audiomixer->offset = 0;
- agg->segment.position = -1;
-
- gst_audio_info_init (&audiomixer->info);
- gst_caps_replace (&audiomixer->current_caps, NULL);
- gst_buffer_replace (&audiomixer->current_buffer, NULL);
-}
-
-static gboolean
-gst_audiomixer_start (GstAggregator * agg)
-{
- GstAudioMixer *audiomixer = GST_AUDIO_MIXER (agg);
-
- gst_audiomixer_reset (audiomixer);
-
- return TRUE;
-}
-
-static gboolean
-gst_audiomixer_stop (GstAggregator * agg)
-{
- GstAudioMixer *audiomixer = GST_AUDIO_MIXER (agg);
-
- gst_audiomixer_reset (audiomixer);
-
- return TRUE;
-}
-
-static GstFlowReturn
-gst_audiomixer_flush (GstAggregator * agg)
-{
- GstAudioMixer *audiomixer = GST_AUDIO_MIXER (agg);
-
- audiomixer->offset = 0;
- agg->segment.position = -1;
- gst_buffer_replace (&audiomixer->current_buffer, NULL);
-
- return GST_FLOW_OK;
-}
-
-static void
gst_audiomixer_class_init (GstAudioMixerClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
GstAggregatorClass *agg_class = (GstAggregatorClass *) klass;
+ GstAudioAggregatorClass *aagg_class = (GstAudioAggregatorClass *) klass;
gobject_class->set_property = gst_audiomixer_set_property;
gobject_class->get_property = gst_audiomixer_get_property;
"Setting this property takes a reference to the supplied GstCaps "
"object", GST_TYPE_CAPS, 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));
-
- 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));
-
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_audiomixer_src_template));
gst_element_class_add_pad_template (gstelement_class,
GST_DEBUG_FUNCPTR (gst_audiomixer_release_pad);
agg_class->sinkpads_type = GST_TYPE_AUDIO_MIXER_PAD;
- agg_class->start = gst_audiomixer_start;
- agg_class->stop = gst_audiomixer_stop;
-
- agg_class->get_next_time = gst_audiomixer_get_next_time;
agg_class->sink_query = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_query);
agg_class->sink_event = GST_DEBUG_FUNCPTR (gst_audiomixer_sink_event);
- agg_class->aggregate = GST_DEBUG_FUNCPTR (gst_audiomixer_aggregate);
- agg_class->clip = GST_DEBUG_FUNCPTR (gst_audiomixer_do_clip);
-
- agg_class->src_event = GST_DEBUG_FUNCPTR (gst_audiomixer_src_event);
- agg_class->src_query = GST_DEBUG_FUNCPTR (gst_audiomixer_src_query);
-
- agg_class->flush = GST_DEBUG_FUNCPTR (gst_audiomixer_flush);
+ aagg_class->aggregate_one_buffer = gst_audiomixer_aggregate_one_buffer;
}
static void
gst_audiomixer_init (GstAudioMixer * audiomixer)
{
- audiomixer->current_caps = NULL;
- gst_audio_info_init (&audiomixer->info);
-
audiomixer->filter_caps = NULL;
- audiomixer->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
- audiomixer->discont_wait = DEFAULT_DISCONT_WAIT;
- audiomixer->output_buffer_duration = DEFAULT_OUTPUT_BUFFER_DURATION;
- gst_aggregator_set_latency (GST_AGGREGATOR (audiomixer),
- audiomixer->output_buffer_duration, audiomixer->output_buffer_duration);
}
static void
GstAudioMixer *audiomixer = GST_AUDIO_MIXER (object);
gst_caps_replace (&audiomixer->filter_caps, NULL);
- gst_caps_replace (&audiomixer->current_caps, NULL);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
GST_DEBUG_OBJECT (audiomixer, "set new caps %" GST_PTR_FORMAT, new_caps);
break;
}
- case PROP_ALIGNMENT_THRESHOLD:
- audiomixer->alignment_threshold = g_value_get_uint64 (value);
- break;
- case PROP_DISCONT_WAIT:
- audiomixer->discont_wait = g_value_get_uint64 (value);
- break;
- case PROP_OUTPUT_BUFFER_DURATION:
- audiomixer->output_buffer_duration = g_value_get_uint64 (value);
- gst_aggregator_set_latency (GST_AGGREGATOR (audiomixer),
- audiomixer->output_buffer_duration,
- audiomixer->output_buffer_duration);
- break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
gst_value_set_caps (value, audiomixer->filter_caps);
GST_OBJECT_UNLOCK (audiomixer);
break;
- case PROP_ALIGNMENT_THRESHOLD:
- g_value_set_uint64 (value, audiomixer->alignment_threshold);
- break;
- case PROP_DISCONT_WAIT:
- g_value_set_uint64 (value, audiomixer->discont_wait);
- break;
- case PROP_OUTPUT_BUFFER_DURATION:
- g_value_set_uint64 (value, audiomixer->output_buffer_duration);
- break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
GST_ELEMENT_CLASS (parent_class)->release_pad (element, pad);
}
-static GstFlowReturn
-gst_audiomixer_do_clip (GstAggregator * agg,
- GstAggregatorPad * bpad, GstBuffer * buffer, GstBuffer ** out)
-{
- GstAudioMixer *audiomixer = GST_AUDIO_MIXER (agg);
- gint rate, bpf;
-
- rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
- bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
-
- buffer = gst_audio_buffer_clip (buffer, &bpad->segment, rate, bpf);
-
- *out = buffer;
- return GST_FLOW_OK;
-}
+/* Called with object lock and pad object lock held */
static gboolean
-gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
- GstBuffer * inbuf)
+gst_audiomixer_aggregate_one_buffer (GstAudioAggregator * aagg,
+ GstAudioAggregatorPad * aaggpad, GstBuffer * inbuf, guint in_offset,
+ GstBuffer * outbuf, guint out_offset, guint num_frames)
{
- GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
- GstClockTime start_time, end_time;
- gboolean discont = FALSE;
- guint64 start_offset, end_offset;
- GstClockTime timestamp, stream_time;
- gint rate, bpf;
-
- g_assert (pad->buffer == NULL);
-
- rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
- bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
-
- timestamp = GST_BUFFER_PTS (inbuf);
- stream_time = gst_segment_to_stream_time (&aggpad->segment, GST_FORMAT_TIME,
- timestamp);
-
- /* 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);
-
- GST_OBJECT_LOCK (pad);
- pad->position = 0;
- pad->size = gst_buffer_get_size (inbuf);
-
- start_time = GST_BUFFER_PTS (inbuf);
- end_time =
- start_time + gst_util_uint64_scale_ceil (pad->size / bpf,
- GST_SECOND, rate);
-
- start_offset = gst_util_uint64_scale (start_time, rate, GST_SECOND);
- end_offset = start_offset + pad->size / bpf;
-
- if (GST_BUFFER_IS_DISCONT (inbuf)
- || GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_RESYNC)
- || pad->new_segment || pad->next_offset == -1) {
- discont = TRUE;
- pad->new_segment = FALSE;
- } else {
- guint64 diff, max_sample_diff;
-
- /* Check discont, based on audiobasesink */
- if (start_offset <= pad->next_offset)
- diff = pad->next_offset - start_offset;
- else
- diff = start_offset - pad->next_offset;
-
- max_sample_diff =
- gst_util_uint64_scale_int (audiomixer->alignment_threshold, rate,
- GST_SECOND);
-
- /* Discont! */
- if (G_UNLIKELY (diff >= max_sample_diff)) {
- if (audiomixer->discont_wait > 0) {
- if (pad->discont_time == GST_CLOCK_TIME_NONE) {
- pad->discont_time = start_time;
- } else if (start_time - pad->discont_time >= audiomixer->discont_wait) {
- discont = TRUE;
- pad->discont_time = GST_CLOCK_TIME_NONE;
- }
- } else {
- discont = TRUE;
- }
- } else if (G_UNLIKELY (pad->discont_time != GST_CLOCK_TIME_NONE)) {
- /* we have had a discont, but are now back on track! */
- pad->discont_time = GST_CLOCK_TIME_NONE;
- }
- }
-
- if (discont) {
- /* Have discont, need resync */
- if (pad->next_offset != -1)
- GST_INFO_OBJECT (pad, "Have discont. Expected %"
- G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
- pad->next_offset, start_offset);
- pad->output_offset = -1;
- pad->next_offset = end_offset;
- } else {
- pad->next_offset += pad->size / bpf;
- }
-
- if (pad->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 < audiomixer->offset) {
- /* Before output segment, drop */
- gst_buffer_unref (inbuf);
- pad->buffer = NULL;
- gst_aggregator_pad_drop_buffer (aggpad);
- pad->position = 0;
- pad->size = 0;
- pad->output_offset = -1;
- GST_DEBUG_OBJECT (pad,
- "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
- G_GUINT64_FORMAT, end_running_time_offset, audiomixer->offset);
- GST_OBJECT_UNLOCK (pad);
- return FALSE;
- }
-
- if (start_running_time_offset < audiomixer->offset) {
- guint diff = (audiomixer->offset - start_running_time_offset) * bpf;
-
- pad->position += diff;
- if (pad->position >= pad->size) {
- /* Empty buffer, drop */
- gst_buffer_unref (inbuf);
- pad->buffer = NULL;
- gst_aggregator_pad_drop_buffer (aggpad);
- pad->position = 0;
- pad->size = 0;
- pad->output_offset = -1;
- GST_DEBUG_OBJECT (pad,
- "Buffer before segment or current position: %" G_GUINT64_FORMAT
- " < %" G_GUINT64_FORMAT, end_running_time_offset,
- audiomixer->offset);
- GST_OBJECT_UNLOCK (pad);
- return FALSE;
- }
- }
-
- pad->output_offset = MAX (start_running_time_offset, audiomixer->offset);
- GST_DEBUG_OBJECT (pad,
- "Buffer resynced: Pad offset %" G_GUINT64_FORMAT
- ", current mixer offset %" G_GUINT64_FORMAT, pad->output_offset,
- audiomixer->offset);
- }
-
- GST_LOG_OBJECT (pad,
- "Queued new buffer at offset %" G_GUINT64_FORMAT, pad->output_offset);
- pad->buffer = inbuf;
-
- GST_OBJECT_UNLOCK (pad);
- return TRUE;
-}
-
-static void
-gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
- GstMapInfo * outmap)
-{
- guint overlap;
- guint out_start;
- GstBuffer *inbuf;
+ GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aaggpad);
GstMapInfo inmap;
+ GstMapInfo outmap;
gint bpf;
- guint blocksize;
- gboolean drop_buf = FALSE;
-
- GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (pad);
-
- blocksize =
- gst_util_uint64_scale (audiomixer->output_buffer_duration,
- GST_AUDIO_INFO_RATE (&audiomixer->info), GST_SECOND);
- blocksize = MAX (1, blocksize);
-
- bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
-
- GST_OBJECT_LOCK (pad);
- /* Overlap => mix */
- if (audiomixer->offset < pad->output_offset)
- out_start = pad->output_offset - audiomixer->offset;
- else
- out_start = 0;
-
- overlap = pad->size / bpf - pad->position / bpf;
- if (overlap > blocksize - out_start)
- overlap = blocksize - out_start;
-
- inbuf = gst_aggregator_pad_get_buffer (aggpad);
- if (inbuf == NULL) {
- GST_OBJECT_UNLOCK (pad);
- return;
- }
if (pad->mute || pad->volume < G_MINDOUBLE) {
GST_DEBUG_OBJECT (pad, "Skipping muted pad");
- gst_buffer_unref (inbuf);
- pad->position += overlap * bpf;
- pad->output_offset += overlap;
- if (pad->position >= pad->size) {
- /* Buffer done, drop it */
- gst_buffer_replace (&pad->buffer, NULL);
- drop_buf = TRUE;
- }
- GST_OBJECT_UNLOCK (pad);
- if (drop_buf)
- gst_aggregator_pad_drop_buffer (aggpad);
- return;
+ return FALSE;
}
- if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) {
- /* skip gap buffer */
- GST_LOG_OBJECT (pad, "skipping GAP buffer");
- gst_buffer_unref (inbuf);
- pad->output_offset += pad->size / bpf;
- /* Buffer done, drop it */
- gst_buffer_replace (&pad->buffer, NULL);
- GST_OBJECT_UNLOCK (pad);
- gst_aggregator_pad_drop_buffer (aggpad);
- return;
- }
+ bpf = GST_AUDIO_INFO_BPF (&aagg->info);
+ gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
GST_LOG_OBJECT (pad, "mixing %u bytes at offset %u from offset %u",
- overlap * bpf, out_start * bpf, pad->position);
+ num_frames * bpf, out_offset * bpf, in_offset * bpf);
+
/* further buffers, need to add them */
if (pad->volume == 1.0) {
- switch (audiomixer->info.finfo->format) {
+ switch (aagg->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
- audiomixer_orc_add_u8 ((gpointer) (outmap->data + out_start * bpf),
- (gpointer) (inmap.data + pad->position),
- overlap * audiomixer->info.channels);
+ audiomixer_orc_add_u8 ((gpointer) (outmap.data + out_offset * bpf),
+ (gpointer) (inmap.data + in_offset * bpf),
+ num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S8:
- audiomixer_orc_add_s8 ((gpointer) (outmap->data + out_start * bpf),
- (gpointer) (inmap.data + pad->position),
- overlap * audiomixer->info.channels);
+ audiomixer_orc_add_s8 ((gpointer) (outmap.data + out_offset * bpf),
+ (gpointer) (inmap.data + in_offset * bpf),
+ num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_U16:
- audiomixer_orc_add_u16 ((gpointer) (outmap->data + out_start * bpf),
- (gpointer) (inmap.data + pad->position),
- overlap * audiomixer->info.channels);
+ audiomixer_orc_add_u16 ((gpointer) (outmap.data + out_offset * bpf),
+ (gpointer) (inmap.data + in_offset * bpf),
+ num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S16:
- audiomixer_orc_add_s16 ((gpointer) (outmap->data + out_start * bpf),
- (gpointer) (inmap.data + pad->position),
- overlap * audiomixer->info.channels);
+ audiomixer_orc_add_s16 ((gpointer) (outmap.data + out_offset * bpf),
+ (gpointer) (inmap.data + in_offset * bpf),
+ num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_U32:
- audiomixer_orc_add_u32 ((gpointer) (outmap->data + out_start * bpf),
- (gpointer) (inmap.data + pad->position),
- overlap * audiomixer->info.channels);
+ audiomixer_orc_add_u32 ((gpointer) (outmap.data + out_offset * bpf),
+ (gpointer) (inmap.data + in_offset * bpf),
+ num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S32:
- audiomixer_orc_add_s32 ((gpointer) (outmap->data + out_start * bpf),
- (gpointer) (inmap.data + pad->position),
- overlap * audiomixer->info.channels);
+ audiomixer_orc_add_s32 ((gpointer) (outmap.data + out_offset * bpf),
+ (gpointer) (inmap.data + in_offset * bpf),
+ num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_F32:
- audiomixer_orc_add_f32 ((gpointer) (outmap->data + out_start * bpf),
- (gpointer) (inmap.data + pad->position),
- overlap * audiomixer->info.channels);
+ audiomixer_orc_add_f32 ((gpointer) (outmap.data + out_offset * bpf),
+ (gpointer) (inmap.data + in_offset * bpf),
+ num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_F64:
- audiomixer_orc_add_f64 ((gpointer) (outmap->data + out_start * bpf),
- (gpointer) (inmap.data + pad->position),
- overlap * audiomixer->info.channels);
+ audiomixer_orc_add_f64 ((gpointer) (outmap.data + out_offset * bpf),
+ (gpointer) (inmap.data + in_offset * bpf),
+ num_frames * aagg->info.channels);
break;
default:
g_assert_not_reached ();
break;
}
} else {
- switch (audiomixer->info.finfo->format) {
+ switch (aagg->info.finfo->format) {
case GST_AUDIO_FORMAT_U8:
- audiomixer_orc_add_volume_u8 ((gpointer) (outmap->data +
- out_start * bpf), (gpointer) (inmap.data + pad->position),
- pad->volume_i8, overlap * audiomixer->info.channels);
+ audiomixer_orc_add_volume_u8 ((gpointer) (outmap.data +
+ out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
+ pad->volume_i8, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S8:
- audiomixer_orc_add_volume_s8 ((gpointer) (outmap->data +
- out_start * bpf), (gpointer) (inmap.data + pad->position),
- pad->volume_i8, overlap * audiomixer->info.channels);
+ audiomixer_orc_add_volume_s8 ((gpointer) (outmap.data +
+ out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
+ pad->volume_i8, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_U16:
- audiomixer_orc_add_volume_u16 ((gpointer) (outmap->data +
- out_start * bpf), (gpointer) (inmap.data + pad->position),
- pad->volume_i16, overlap * audiomixer->info.channels);
+ audiomixer_orc_add_volume_u16 ((gpointer) (outmap.data +
+ out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
+ pad->volume_i16, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S16:
- audiomixer_orc_add_volume_s16 ((gpointer) (outmap->data +
- out_start * bpf), (gpointer) (inmap.data + pad->position),
- pad->volume_i16, overlap * audiomixer->info.channels);
+ audiomixer_orc_add_volume_s16 ((gpointer) (outmap.data +
+ out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
+ pad->volume_i16, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_U32:
- audiomixer_orc_add_volume_u32 ((gpointer) (outmap->data +
- out_start * bpf), (gpointer) (inmap.data + pad->position),
- pad->volume_i32, overlap * audiomixer->info.channels);
+ audiomixer_orc_add_volume_u32 ((gpointer) (outmap.data +
+ out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
+ pad->volume_i32, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_S32:
- audiomixer_orc_add_volume_s32 ((gpointer) (outmap->data +
- out_start * bpf), (gpointer) (inmap.data + pad->position),
- pad->volume_i32, overlap * audiomixer->info.channels);
+ audiomixer_orc_add_volume_s32 ((gpointer) (outmap.data +
+ out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
+ pad->volume_i32, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_F32:
- audiomixer_orc_add_volume_f32 ((gpointer) (outmap->data +
- out_start * bpf), (gpointer) (inmap.data + pad->position),
- pad->volume, overlap * audiomixer->info.channels);
+ audiomixer_orc_add_volume_f32 ((gpointer) (outmap.data +
+ out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
+ pad->volume, num_frames * aagg->info.channels);
break;
case GST_AUDIO_FORMAT_F64:
- audiomixer_orc_add_volume_f64 ((gpointer) (outmap->data +
- out_start * bpf), (gpointer) (inmap.data + pad->position),
- pad->volume, overlap * audiomixer->info.channels);
+ audiomixer_orc_add_volume_f64 ((gpointer) (outmap.data +
+ out_offset * bpf), (gpointer) (inmap.data + in_offset * bpf),
+ pad->volume, num_frames * aagg->info.channels);
break;
default:
g_assert_not_reached ();
}
}
gst_buffer_unmap (inbuf, &inmap);
- gst_buffer_unref (inbuf);
-
- pad->position += overlap * bpf;
- pad->output_offset += overlap;
-
- if (pad->position == pad->size) {
- /* Buffer done, drop it */
- gst_buffer_replace (&pad->buffer, NULL);
- GST_DEBUG_OBJECT (pad, "Finished mixing buffer, waiting for next");
- drop_buf = TRUE;
- }
-
- GST_OBJECT_UNLOCK (pad);
-
- if (drop_buf)
- gst_aggregator_pad_drop_buffer (aggpad);
-}
-
-static GstFlowReturn
-gst_audiomixer_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.
- */
- GstAudioMixer *audiomixer;
- GList *iter;
- GstFlowReturn ret;
- GstBuffer *outbuf = NULL;
- GstMapInfo outmap;
- gint64 next_offset;
- gint64 next_timestamp;
- gint rate, bpf;
- gboolean dropped = FALSE;
- gboolean is_eos = TRUE;
- gboolean is_done = TRUE;
- guint blocksize;
-
- audiomixer = GST_AUDIO_MIXER (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 (audiomixer->info.finfo->format == GST_AUDIO_FORMAT_UNKNOWN)) {
- if (timeout) {
- GST_DEBUG_OBJECT (audiomixer,
- "Got timeout before receiving any caps, don't output anything");
-
- /* Advance position */
- if (agg->segment.rate > 0.0)
- agg->segment.position += audiomixer->output_buffer_duration;
- else if (agg->segment.position > audiomixer->output_buffer_duration)
- agg->segment.position -= audiomixer->output_buffer_duration;
- else
- agg->segment.position = 0;
-
- return GST_FLOW_OK;
- } else {
- goto not_negotiated;
- }
- }
-
- blocksize =
- gst_util_uint64_scale (audiomixer->output_buffer_duration,
- GST_AUDIO_INFO_RATE (&audiomixer->info), GST_SECOND);
- blocksize = MAX (1, blocksize);
-
- if (audiomixer->send_caps) {
- gst_aggregator_set_src_caps (agg, audiomixer->current_caps);
-
- audiomixer->offset = gst_util_uint64_scale (agg->segment.position,
- GST_AUDIO_INFO_RATE (&audiomixer->info), GST_SECOND);
-
- audiomixer->send_caps = FALSE;
- }
-
- rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
- bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
-
- /* 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 = audiomixer->offset + blocksize;
- } else {
- next_offset = audiomixer->offset - blocksize;
- }
- next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
-
- if (audiomixer->current_buffer) {
- outbuf = audiomixer->current_buffer;
- } else {
- outbuf = gst_buffer_new_and_alloc (blocksize * bpf);
- gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
- gst_audio_format_fill_silence (audiomixer->info.finfo, outmap.data,
- outmap.size);
- gst_buffer_unmap (outbuf, &outmap);
- audiomixer->current_buffer = outbuf;
- }
-
- GST_LOG_OBJECT (agg,
- "Starting to mix %u samples for offset %" G_GUINT64_FORMAT
- " with timestamp %" GST_TIME_FORMAT, blocksize,
- audiomixer->offset, GST_TIME_ARGS (agg->segment.position));
-
- gst_buffer_map (outbuf, &outmap, GST_MAP_READWRITE);
-
- GST_OBJECT_LOCK (agg);
- for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) {
- GstBuffer *inbuf;
- GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (iter->data);
- GstAggregatorPad *aggpad = GST_AGGREGATOR_PAD (iter->data);
-
- if (!gst_aggregator_pad_is_eos (aggpad))
- is_eos = FALSE;
-
- inbuf = gst_aggregator_pad_get_buffer (aggpad);
- if (!inbuf) {
- if (timeout) {
- if (pad->output_offset < next_offset) {
- gint64 diff = next_offset - pad->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, rate)));
- }
- } else if (!gst_aggregator_pad_is_eos (aggpad)) {
- is_done = FALSE;
- }
- continue;
- }
-
- g_assert (!pad->buffer || pad->buffer == inbuf);
-
- /* New buffer? */
- if (!pad->buffer) {
- /* Takes ownership of buffer */
- if (!gst_audio_mixer_fill_buffer (audiomixer, pad, inbuf)) {
- dropped = TRUE;
- continue;
- }
- } else {
- gst_buffer_unref (inbuf);
- }
-
- if (!pad->buffer && !dropped && gst_aggregator_pad_is_eos (aggpad)) {
- GST_DEBUG_OBJECT (aggpad, "Pad is in EOS state");
- continue;
- }
-
- g_assert (pad->buffer);
-
- /* This pad is lacking behind, we need to update the offset
- * and maybe drop the current buffer */
- if (pad->output_offset < audiomixer->offset) {
- gint64 diff = audiomixer->offset - pad->output_offset;
- gint bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
-
- if (pad->position + (diff * bpf) > pad->size)
- diff = (pad->size - pad->position) / bpf;
- pad->position += diff * bpf;
- pad->output_offset += diff;
-
- if (pad->position == pad->size) {
- /* Buffer done, drop it */
- gst_buffer_replace (&pad->buffer, NULL);
- gst_aggregator_pad_drop_buffer (aggpad);
- dropped = TRUE;
- continue;
- }
- }
-
- if (pad->output_offset >= audiomixer->offset
- && pad->output_offset < audiomixer->offset + blocksize && pad->buffer) {
- GST_LOG_OBJECT (aggpad, "Mixing buffer for current offset");
- gst_audio_mixer_mix_buffer (audiomixer, pad, &outmap);
-
- if (pad->output_offset >= next_offset) {
- GST_DEBUG_OBJECT (pad,
- "Pad is after current offset: %" G_GUINT64_FORMAT " >= %"
- G_GUINT64_FORMAT, pad->output_offset, next_offset);
- } else {
- is_done = FALSE;
- }
- }
- }
- GST_OBJECT_UNLOCK (agg);
-
gst_buffer_unmap (outbuf, &outmap);
- if (dropped) {
- /* We dropped a buffer, retry */
- GST_INFO_OBJECT (audiomixer,
- "A pad dropped a buffer, wait for the next one");
- return GST_FLOW_OK;
- }
-
- if (!is_done && !is_eos) {
- /* Get more buffers */
- GST_INFO_OBJECT (audiomixer,
- "We're not done yet for the current offset, waiting for more data");
- return GST_FLOW_OK;
- }
-
- if (is_eos) {
- gint64 max_offset = 0;
- gboolean empty_buffer = TRUE;
-
- GST_DEBUG_OBJECT (audiomixer, "We're EOS");
-
- GST_OBJECT_LOCK (agg);
- for (iter = GST_ELEMENT (agg)->sinkpads; iter; iter = iter->next) {
- GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (iter->data);
-
- max_offset = MAX ((gint64) max_offset, (gint64) pad->output_offset);
- if (pad->output_offset > audiomixer->offset)
- empty_buffer = FALSE;
- }
- GST_OBJECT_UNLOCK (agg);
-
- /* This means EOS or no pads at all */
- if (empty_buffer) {
- gst_buffer_replace (&audiomixer->current_buffer, NULL);
- goto eos;
- }
-
- if (max_offset <= next_offset) {
- GST_DEBUG_OBJECT (audiomixer,
- "Last buffer is incomplete: %" G_GUINT64_FORMAT " <= %"
- G_GUINT64_FORMAT, max_offset, next_offset);
- next_offset = max_offset;
- if (next_offset > audiomixer->offset)
- gst_buffer_resize (outbuf, 0, (next_offset - audiomixer->offset) * bpf);
-
- next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
- }
- }
-
- /* set timestamps on the output buffer */
- if (agg->segment.rate > 0.0) {
- GST_BUFFER_TIMESTAMP (outbuf) = agg->segment.position;
- GST_BUFFER_OFFSET (outbuf) = audiomixer->offset;
- GST_BUFFER_OFFSET_END (outbuf) = next_offset;
- GST_BUFFER_DURATION (outbuf) = next_timestamp - agg->segment.position;
- } else {
- GST_BUFFER_TIMESTAMP (outbuf) = next_timestamp;
- GST_BUFFER_OFFSET (outbuf) = next_offset;
- GST_BUFFER_OFFSET_END (outbuf) = audiomixer->offset;
- GST_BUFFER_DURATION (outbuf) = agg->segment.position - next_timestamp;
- }
-
- audiomixer->offset = next_offset;
- agg->segment.position = next_timestamp;
-
- /* send it out */
- GST_LOG_OBJECT (audiomixer,
- "pushing outbuf %p, timestamp %" GST_TIME_FORMAT " offset %"
- G_GINT64_FORMAT, outbuf, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
- GST_BUFFER_OFFSET (outbuf));
-
- ret = gst_aggregator_finish_buffer (agg, audiomixer->current_buffer);
- audiomixer->current_buffer = NULL;
-
- GST_LOG_OBJECT (audiomixer, "pushed outbuf, result = %s",
- gst_flow_get_name (ret));
-
- if (ret == GST_FLOW_OK && is_eos)
- goto eos;
-
- return ret;
- /* ERRORS */
-not_negotiated:
- {
- GST_ELEMENT_ERROR (audiomixer, STREAM, FORMAT, (NULL),
- ("Unknown data received, not negotiated"));
- return GST_FLOW_NOT_NEGOTIATED;
- }
-
-eos:
- {
- GST_DEBUG_OBJECT (audiomixer, "EOS");
- return GST_FLOW_EOS;
- }
+ return TRUE;
}
+
/* GstChildProxy implementation */
static GObject *
gst_audiomixer_child_proxy_get_child_by_index (GstChildProxy * child_proxy,
#define __GST_AUDIO_MIXER_H__
#include <gst/gst.h>
-#include <gst/base/gstaggregator.h>
#include <gst/audio/audio.h>
+#include "gstaudioaggregator.h"
G_BEGIN_DECLS
* The audiomixer object structure.
*/
struct _GstAudioMixer {
- GstAggregator aggregator;
-
- /* the next are valid for both int and float */
- GstAudioInfo info;
-
- /* counters to keep track of timestamps */
- gint64 offset;
- /* Buffer starting at offset containing block_size samples */
- GstBuffer *current_buffer;
-
- /* current caps */
- GstCaps *current_caps;
- gboolean send_caps;
+ GstAudioAggregator element;
/* target caps (set via property) */
GstCaps *filter_caps;
-
- GstClockTime alignment_threshold;
- GstClockTime discont_wait;
-
- /* Duration of every output buffer */
- GstClockTime output_buffer_duration;
};
struct _GstAudioMixerClass {
- GstAggregatorClass parent_class;
+ GstAudioAggregatorClass parent_class;
};
GType gst_audiomixer_get_type (void);
#define GST_AUDIO_MIXER_PAD_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_MIXER_PAD,GstAudioMixerPadClass))
struct _GstAudioMixerPad {
- GstAggregatorPad parent;
+ GstAudioAggregatorPad parent;
gdouble volume;
gint volume_i32;
gint volume_i16;
gint volume_i8;
gboolean mute;
-
- /* < private > */
- 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;
};
struct _GstAudioMixerPadClass {
- GstAggregatorPadClass parent_class;
+ GstAudioAggregatorPadClass parent_class;
};
GType gst_audiomixer_pad_get_type (void);
projects / platform / upstream / gstreamer.git / commitdiff