audiobuffersplit: New element that splits raw audio buffers into equal-sized buffers

This is useful e.g. if audio buffers should be exactly the duration of a
video frame, or if a audio buffers should never be too large because of
latency constraints.

The element is taking a fractional buffer duration, to allow working
with e.g. 1001/30000 as output duration and it accumulates rounding
errors in the buffer durations and compensates for them by making some
buffers one sample larger than the others.

https://bugzilla.gnome.org/show_bug.cgi?id=774689

configure.ac

@@ -490,6 +490,7 @@ AG_GST_CHECK_PLUGIN(adpcmenc)
 AG_GST_CHECK_PLUGIN(aiff)
 AG_GST_CHECK_PLUGIN(videoframe_audiolevel)
 AG_GST_CHECK_PLUGIN(asfmux)
+AG_GST_CHECK_PLUGIN(audiobuffersplit)
 AG_GST_CHECK_PLUGIN(audiofxbad)
 AG_GST_CHECK_PLUGIN(audiomixer)
 AG_GST_CHECK_PLUGIN(compositor)
@@ -3678,6 +3679,7 @@ gst/adpcmenc/Makefile
 gst/aiff/Makefile
 gst/videoframe_audiolevel/Makefile
 gst/asfmux/Makefile
+gst/audiobuffersplit/Makefile
 gst/audiofxbad/Makefile
 gst/audiomixer/Makefile
 gst/audiovisualizers/Makefile

gst/audiobuffersplit/Makefile.am

@@ -0,0 +1,14 @@
+plugin_LTLIBRARIES = libgstaudiobuffersplit.la
+
+libgstaudiobuffersplit_la_SOURCES = gstaudiobuffersplit.c
+
+libgstaudiobuffersplit_la_CFLAGS = $(GST_PLUGINS_BAD_CFLAGS) \
+	$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) \
+	$(GST_CFLAGS)
+libgstaudiobuffersplit_la_LIBADD = \
+	$(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) \
+	$(GST_BASE_LIBS)  $(GST_LIBS) $(LIBM)
+libgstaudiobuffersplit_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstaudiobuffersplit_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
+
+noinst_HEADERS = gstaudiobuffersplit.h

gst/audiobuffersplit/gstaudiobuffersplit.c

@@ -0,0 +1,574 @@
+/*
+ * GStreamer
+ * Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * 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.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstaudiobuffersplit.h"
+
+#define GST_CAT_DEFAULT gst_audio_buffer_split_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw")
+    );
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw")
+    );
+
+enum
+{
+  PROP_0,
+  PROP_OUTPUT_BUFFER_DURATION,
+  PROP_ALIGNMENT_THRESHOLD,
+  PROP_DISCONT_WAIT,
+  LAST_PROP
+};
+
+#define DEFAULT_OUTPUT_BUFFER_DURATION_N (1)
+#define DEFAULT_OUTPUT_BUFFER_DURATION_D (50)
+#define DEFAULT_ALIGNMENT_THRESHOLD   (40 * GST_MSECOND)
+#define DEFAULT_DISCONT_WAIT (1 * GST_SECOND)
+
+#define parent_class gst_audio_buffer_split_parent_class
+G_DEFINE_TYPE (GstAudioBufferSplit, gst_audio_buffer_split, GST_TYPE_ELEMENT);
+
+static GstFlowReturn gst_audio_buffer_split_sink_chain (GstPad * pad,
+    GstObject * parent, GstBuffer * buffer);
+static gboolean gst_audio_buffer_split_sink_event (GstPad * pad,
+    GstObject * parent, GstEvent * event);
+static gboolean gst_audio_buffer_split_src_query (GstPad * pad,
+    GstObject * parent, GstQuery * query);
+
+static void gst_audio_buffer_split_finalize (GObject * object);
+static void gst_audio_buffer_split_get_property (GObject * object,
+    guint property_id, GValue * value, GParamSpec * pspec);
+static void gst_audio_buffer_split_set_property (GObject * object,
+    guint property_id, const GValue * value, GParamSpec * pspec);
+
+static GstStateChangeReturn gst_audio_buffer_split_change_state (GstElement *
+    element, GstStateChange transition);
+
+static void
+gst_audio_buffer_split_class_init (GstAudioBufferSplitClass * klass)
+{
+  GObjectClass *gobject_class = (GObjectClass *) klass;
+  GstElementClass *gstelement_class = (GstElementClass *) klass;
+
+  gobject_class->set_property = gst_audio_buffer_split_set_property;
+  gobject_class->get_property = gst_audio_buffer_split_get_property;
+  gobject_class->finalize = gst_audio_buffer_split_finalize;
+
+  g_object_class_install_property (gobject_class, PROP_OUTPUT_BUFFER_DURATION,
+      gst_param_spec_fraction ("output-buffer-duration",
+          "Output Buffer Duration", "Output block size in seconds", 1, G_MAXINT,
+          G_MAXINT, 1, DEFAULT_OUTPUT_BUFFER_DURATION_N,
+          DEFAULT_OUTPUT_BUFFER_DURATION_D,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+          GST_PARAM_MUTABLE_READY));
+
+  g_object_class_install_property (gobject_class, PROP_ALIGNMENT_THRESHOLD,
+      g_param_spec_uint64 ("alignment-threshold", "Alignment Threshold",
+          "Timestamp alignment threshold in nanoseconds", 0,
+          G_MAXUINT64 - 1, DEFAULT_ALIGNMENT_THRESHOLD,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
+          GST_PARAM_MUTABLE_READY));
+
+  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 |
+          GST_PARAM_MUTABLE_READY));
+
+  gst_element_class_set_static_metadata (gstelement_class,
+      "Audio Buffer Split", "Audio/Filter",
+      "Splits raw audio buffers into equal sized chunks",
+      "Sebastian Dröge <sebastian@centricular.com>");
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&src_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&sink_template));
+
+  gstelement_class->change_state = gst_audio_buffer_split_change_state;
+}
+
+static void
+gst_audio_buffer_split_init (GstAudioBufferSplit * self)
+{
+  self->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
+  gst_pad_set_chain_function (self->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_audio_buffer_split_sink_chain));
+  gst_pad_set_event_function (self->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_audio_buffer_split_sink_event));
+  GST_PAD_SET_PROXY_CAPS (self->sinkpad);
+  gst_element_add_pad (GST_ELEMENT (self), self->sinkpad);
+
+  self->srcpad = gst_pad_new_from_static_template (&src_template, "src");
+  gst_pad_set_query_function (self->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_audio_buffer_split_src_query));
+  GST_PAD_SET_PROXY_CAPS (self->srcpad);
+  gst_pad_use_fixed_caps (self->srcpad);
+  gst_element_add_pad (GST_ELEMENT (self), self->srcpad);
+
+  self->output_buffer_duration_n = DEFAULT_OUTPUT_BUFFER_DURATION_N;
+  self->output_buffer_duration_d = DEFAULT_OUTPUT_BUFFER_DURATION_D;
+  self->alignment_threshold = DEFAULT_ALIGNMENT_THRESHOLD;
+  self->discont_wait = DEFAULT_DISCONT_WAIT;
+
+  self->adapter = gst_adapter_new ();
+}
+
+static void
+gst_audio_buffer_split_finalize (GObject * object)
+{
+  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);
+
+  if (self->adapter) {
+    gst_object_unref (self->adapter);
+    self->adapter = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static void
+gst_audio_buffer_split_set_property (GObject * object, guint property_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);
+
+  switch (property_id) {
+    case PROP_OUTPUT_BUFFER_DURATION:
+      self->output_buffer_duration_n = gst_value_get_fraction_numerator (value);
+      self->output_buffer_duration_d =
+          gst_value_get_fraction_denominator (value);
+      break;
+    case PROP_ALIGNMENT_THRESHOLD:
+      self->alignment_threshold = g_value_get_uint64 (value);
+      break;
+    case PROP_DISCONT_WAIT:
+      self->discont_wait = g_value_get_uint64 (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_audio_buffer_split_get_property (GObject * object, guint property_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (object);
+
+  switch (property_id) {
+    case PROP_OUTPUT_BUFFER_DURATION:
+      gst_value_set_fraction (value, self->output_buffer_duration_n,
+          self->output_buffer_duration_d);
+      break;
+    case PROP_ALIGNMENT_THRESHOLD:
+      g_value_set_uint64 (value, self->alignment_threshold);
+      break;
+    case PROP_DISCONT_WAIT:
+      g_value_set_uint64 (value, self->discont_wait);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
+      break;
+  }
+}
+
+static GstStateChangeReturn
+gst_audio_buffer_split_change_state (GstElement * element,
+    GstStateChange transition)
+{
+  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (element);
+  GstStateChangeReturn state_ret;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      gst_audio_info_init (&self->info);
+      gst_segment_init (&self->segment, GST_FORMAT_TIME);
+      self->discont_time = GST_CLOCK_TIME_NONE;
+      self->next_offset = -1;
+      self->resync_time = GST_CLOCK_TIME_NONE;
+      self->current_offset = -1;
+      self->accumulated_error = 0;
+      break;
+    default:
+      break;
+  }
+
+  state_ret =
+      GST_ELEMENT_CLASS (gst_audio_buffer_split_parent_class)->change_state
+      (element, transition);
+  if (state_ret == GST_STATE_CHANGE_FAILURE)
+    return state_ret;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_adapter_clear (self->adapter);
+      break;
+    default:
+      break;
+  }
+
+  return state_ret;
+}
+
+static GstFlowReturn
+gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force)
+{
+  gint rate, bpf;
+  gint size, avail;
+  GstFlowReturn ret = GST_FLOW_OK;
+
+  rate = GST_AUDIO_INFO_RATE (&self->info);
+  bpf = GST_AUDIO_INFO_BPF (&self->info);
+
+  size = self->samples_per_buffer * bpf;
+
+  /* If we accumulated enough error for one sample, include one
+   * more sample in this buffer. Accumulated error is updated below */
+  if (self->error_per_buffer + self->accumulated_error >=
+      self->output_buffer_duration_d)
+    size += bpf;
+
+  while ((avail = gst_adapter_available (self->adapter)) >= size || (force
+          && avail > 0)) {
+    GstBuffer *buffer;
+    GstClockTime resync_time_diff;
+
+    size = MIN (size, avail);
+    buffer = gst_adapter_take_buffer (self->adapter, size);
+
+    resync_time_diff =
+        gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
+    if (self->segment.rate < 0.0) {
+      if (self->resync_time > resync_time_diff)
+        GST_BUFFER_TIMESTAMP (buffer) = self->resync_time - resync_time_diff;
+      else
+        GST_BUFFER_TIMESTAMP (buffer) = 0;
+      GST_BUFFER_DURATION (buffer) =
+          gst_util_uint64_scale (size / bpf, GST_SECOND, rate);
+
+      self->current_offset += size / bpf;
+    } else {
+      GST_BUFFER_TIMESTAMP (buffer) = self->resync_time + resync_time_diff;
+      self->current_offset += size / bpf;
+      resync_time_diff =
+          gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
+      GST_BUFFER_DURATION (buffer) =
+          resync_time_diff - (GST_BUFFER_TIMESTAMP (buffer) -
+          self->resync_time);
+    }
+
+    GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
+    GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
+
+    self->accumulated_error =
+        (self->accumulated_error +
+        self->error_per_buffer) % self->output_buffer_duration_d;
+
+    GST_LOG_OBJECT (self,
+        "Outputting buffer at timestamp %" GST_TIME_FORMAT " with duration %"
+        GST_TIME_FORMAT " (%u samples)",
+        GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
+        GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)), size / bpf);
+
+    ret = gst_pad_push (self->srcpad, buffer);
+    if (ret != GST_FLOW_OK)
+      break;
+  }
+
+  return ret;
+}
+
+static GstFlowReturn
+gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
+    GstBuffer * buffer)
+{
+  GstClockTime timestamp;
+  gsize size;
+  guint64 start_offset, end_offset;
+  gint rate, bpf;
+  gboolean discont = FALSE;
+  GstFlowReturn ret = GST_FLOW_OK;
+
+  timestamp = GST_BUFFER_TIMESTAMP (buffer);
+
+  rate = GST_AUDIO_INFO_RATE (&self->info);
+  bpf = GST_AUDIO_INFO_BPF (&self->info);
+  start_offset = gst_util_uint64_scale (timestamp, rate, GST_SECOND);
+  size = gst_buffer_get_size (buffer);
+  end_offset = start_offset + size / bpf;
+
+  if (self->segment.rate < 0.0) {
+    guint64 tmp = end_offset;
+    end_offset = start_offset;
+    start_offset = tmp;
+  }
+
+  if (GST_BUFFER_IS_DISCONT (buffer)
+      || GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_RESYNC)
+      || self->resync_time == GST_CLOCK_TIME_NONE) {
+    discont = TRUE;
+  } else {
+    guint64 diff, max_sample_diff;
+
+    /* Check discont, based on audiobasesink */
+    if (start_offset <= self->next_offset)
+      diff = self->next_offset - start_offset;
+    else
+      diff = start_offset - self->next_offset;
+
+    max_sample_diff =
+        gst_util_uint64_scale_int (self->alignment_threshold, rate, GST_SECOND);
+
+    /* Discont! */
+    if (G_UNLIKELY (diff >= max_sample_diff)) {
+      if (self->discont_wait > 0) {
+        if (self->discont_time == GST_CLOCK_TIME_NONE) {
+          self->discont_time = timestamp;
+        } else if (timestamp - self->discont_time >= self->discont_wait) {
+          discont = TRUE;
+          self->discont_time = GST_CLOCK_TIME_NONE;
+        }
+      } else {
+        discont = TRUE;
+      }
+    } else if (G_UNLIKELY (self->discont_time != GST_CLOCK_TIME_NONE)) {
+      /* we have had a discont, but are now back on track! */
+      self->discont_time = GST_CLOCK_TIME_NONE;
+    }
+  }
+
+  if (discont) {
+    /* Have discont, need resync */
+    if (self->next_offset != -1) {
+      GST_INFO_OBJECT (self, "Have discont. Expected %"
+          G_GUINT64_FORMAT ", got %" G_GUINT64_FORMAT,
+          self->next_offset, start_offset);
+      ret = gst_audio_buffer_split_output (self, TRUE);
+    }
+    self->next_offset = end_offset;
+    self->resync_time = timestamp;
+    self->current_offset = 0;
+    self->accumulated_error = 0;
+    gst_adapter_clear (self->adapter);
+  } else {
+    if (self->segment.rate < 0.0) {
+      if (self->next_offset > size / bpf)
+        self->next_offset -= size / bpf;
+      else
+        self->next_offset = 0;
+    } else {
+      self->next_offset += size / bpf;
+    }
+  }
+
+  return ret;
+}
+
+static GstBuffer *
+gst_audio_buffer_split_clip_buffer (GstAudioBufferSplit * self,
+    GstBuffer * buffer)
+{
+  return gst_audio_buffer_clip (buffer, &self->segment,
+      GST_AUDIO_INFO_RATE (&self->info), GST_AUDIO_INFO_BPF (&self->info));
+}
+
+static GstFlowReturn
+gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
+    GstBuffer * buffer)
+{
+  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
+  GstFlowReturn ret;
+
+  if (GST_AUDIO_INFO_FORMAT (&self->info) == GST_AUDIO_FORMAT_UNKNOWN) {
+    gst_buffer_unref (buffer);
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
+
+  buffer = gst_audio_buffer_split_clip_buffer (self, buffer);
+  if (!buffer)
+    return GST_FLOW_OK;
+
+  ret = gst_audio_buffer_split_handle_discont (self, buffer);
+  if (ret != GST_FLOW_OK) {
+    gst_buffer_unref (buffer);
+    return ret;
+  }
+
+  gst_adapter_push (self->adapter, buffer);
+
+  return gst_audio_buffer_split_output (self, FALSE);
+}
+
+static gboolean
+gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
+    GstEvent * event)
+{
+  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
+  gboolean ret = FALSE;
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_CAPS:{
+      GstCaps *caps;
+
+      gst_event_parse_caps (event, &caps);
+
+      ret = gst_audio_info_from_caps (&self->info, caps);
+
+      if (ret) {
+        self->samples_per_buffer =
+            (((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
+            self->output_buffer_duration_n) / self->output_buffer_duration_d;
+        if (self->samples_per_buffer == 0)
+          ret = FALSE;
+
+        self->error_per_buffer =
+            (((guint64) GST_AUDIO_INFO_RATE (&self->info)) *
+            self->output_buffer_duration_n) % self->output_buffer_duration_d;
+        self->accumulated_error = 0;
+
+        GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps);
+        GST_DEBUG_OBJECT (self, "Buffer duration: %u/%u",
+            self->output_buffer_duration_n, self->output_buffer_duration_d);
+        GST_DEBUG_OBJECT (self, "Samples per buffer: %u (error: %u/%u)",
+            self->samples_per_buffer, self->error_per_buffer,
+            self->output_buffer_duration_d);
+      } else {
+        ret = FALSE;
+      }
+
+      if (ret)
+        ret = gst_pad_event_default (pad, parent, event);
+      else
+        gst_event_unref (event);
+
+      break;
+    }
+    case GST_EVENT_FLUSH_STOP:
+      gst_segment_init (&self->segment, GST_FORMAT_TIME);
+      self->discont_time = GST_CLOCK_TIME_NONE;
+      self->next_offset = -1;
+      self->resync_time = GST_CLOCK_TIME_NONE;
+      self->current_offset = -1;
+      self->accumulated_error = 0;
+      gst_adapter_clear (self->adapter);
+      ret = gst_pad_event_default (pad, parent, event);
+      break;
+    case GST_EVENT_SEGMENT:
+      gst_event_copy_segment (event, &self->segment);
+      if (self->segment.format != GST_FORMAT_TIME) {
+        gst_event_unref (event);
+        ret = FALSE;
+      } else {
+        ret = gst_pad_event_default (pad, parent, event);
+      }
+      break;
+    case GST_EVENT_EOS:
+      gst_audio_buffer_split_output (self, TRUE);
+      ret = gst_pad_event_default (pad, parent, event);
+      break;
+    default:
+      ret = gst_pad_event_default (pad, parent, event);
+      break;
+  }
+
+  return ret;
+}
+
+static gboolean
+gst_audio_buffer_split_src_query (GstPad * pad,
+    GstObject * parent, GstQuery * query)
+{
+  GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
+  gboolean ret = FALSE;
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_LATENCY:{
+      if ((ret = gst_pad_peer_query (self->sinkpad, query))) {
+        GstClockTime latency;
+        GstClockTime min, max;
+        gboolean live;
+
+        gst_query_parse_latency (query, &live, &min, &max);
+
+        GST_DEBUG_OBJECT (self, "Peer latency: min %"
+            GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+            GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+
+        latency =
+            gst_util_uint64_scale (GST_SECOND, self->output_buffer_duration_n,
+            self->output_buffer_duration_d);
+
+        GST_DEBUG_OBJECT (self, "Our latency: min %" GST_TIME_FORMAT
+            ", max %" GST_TIME_FORMAT,
+            GST_TIME_ARGS (latency), GST_TIME_ARGS (latency));
+
+        min += latency;
+        if (max != GST_CLOCK_TIME_NONE)
+          max += latency;
+
+        GST_DEBUG_OBJECT (self, "Calculated total latency : min %"
+            GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
+            GST_TIME_ARGS (min), GST_TIME_ARGS (max));
+
+        gst_query_set_latency (query, live, min, max);
+      }
+
+      break;
+    }
+    default:
+      ret = gst_pad_query_default (pad, parent, query);
+      break;
+  }
+
+  return ret;
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  GST_DEBUG_CATEGORY_INIT (gst_audio_buffer_split_debug, "audiobuffersplit",
+      0, "Audio buffer splitter");
+
+  gst_element_register (plugin, "audiobuffersplit", GST_RANK_NONE,
+      GST_TYPE_AUDIO_BUFFER_SPLIT);
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    audiobuffersplit,
+    "Audio buffer splitter",
+    plugin_init, VERSION, "LGPL", PACKAGE_NAME, GST_PACKAGE_ORIGIN)

gst/audiobuffersplit/gstaudiobuffersplit.h

@@ -0,0 +1,76 @@
+/* 
+ * GStreamer
+ * Copyright (C) 2016 Sebastian Dröge <sebastian@centricular.com>
+ *
+ * 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_BUFFER_SPLIT_H__
+#define __GST_AUDIO_BUFFER_SPLIT_H__
+
+#include <gst/gst.h>
+#include <gst/base/base.h>
+#include <gst/audio/audio.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_AUDIO_BUFFER_SPLIT            (gst_audio_buffer_split_get_type())
+#define GST_AUDIO_BUFFER_SPLIT(obj)            (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BUFFER_SPLIT,GstAudioBufferSplit))
+#define GST_IS_AUDIO_BUFFER_SPLIT(obj)         (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BUFFER_SPLIT))
+#define GST_AUDIO_BUFFER_SPLIT_CLASS(klass)    (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_BUFFER_SPLIT,GstAudioBufferSplitClass))
+#define GST_IS_AUDIO_BUFFER_SPLIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_BUFFER_SPLIT))
+#define GST_AUDIO_BUFFER_SPLIT_GET_CLASS(obj)  (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_BUFFER_SPLIT,GstAudioBufferSplitClass))
+
+typedef struct _GstAudioBufferSplit      GstAudioBufferSplit;
+typedef struct _GstAudioBufferSplitClass GstAudioBufferSplitClass;
+
+struct _GstAudioBufferSplit {
+  GstElement parent;
+
+  GstPad *srcpad, *sinkpad;
+
+  /* Properties */
+  gint output_buffer_duration_n;
+  gint output_buffer_duration_d;
+  GstClockTime alignment_threshold;
+  GstClockTime discont_wait;
+
+  /* State */
+  GstSegment segment;
+  GstAudioInfo info;
+
+  GstAdapter *adapter;
+
+  GstClockTime discont_time; /* timestamp of last discont */
+  guint64 next_offset; /* expected next input sample offset */
+
+  GstClockTime resync_time; /* timestamp of resync after discont */
+  guint64 current_offset; /* offset from start time in samples */
+
+  guint samples_per_buffer;
+  guint error_per_buffer;
+  guint accumulated_error;
+};
+
+struct _GstAudioBufferSplitClass {
+  GstElementClass parent_class;
+};
+
+GType gst_audio_buffer_split_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_BUFFER_SPLIT_H__ */

gst/audiobuffersplit/meson.build

@@ -0,0 +1,12 @@
+audiobuffersplit_sources = [
+  'gstaudiobuffersplit.c',
+]
+
+gstaudiobuffersplit = library('gstaudiobuffersplit',
+  audiobuffersplit_sources,
+  c_args : gst_plugins_bad_args,
+  include_directories : [configinc],
+  dependencies : [gstbase_dep, gstaudio_dep],
+  install : true,
+  install_dir : plugins_install_dir,
+)

gst/meson.build

@@ -5,6 +5,7 @@ subdir('aiff')
 subdir('asfmux')
 # not ported to 1.0
 #subdir('audiobuffer')
+subdir('audiobuffersplit')
 subdir('audiofxbad')
 subdir('audiomixer')
 subdir('audiovisualizers')