gst/audiobuffersplit/gstaudiobuffersplit.c - imx-gst-plugins-bad - Git at Google

 /*
  * 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,
   PROP_STRICT_BUFFER_SIZE,
   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 DEFAULT_STRICT_BUFFER_SIZE (FALSE)

 #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));

   g_object_class_install_property (gobject_class, PROP_STRICT_BUFFER_SIZE,
       g_param_spec_boolean ("strict-buffer-size", "Strict buffer size",
           "Discard the last samples at EOS or discont if they are too "
           "small to fill a buffer", DEFAULT_STRICT_BUFFER_SIZE,
           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->srcpad,
       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->strict_buffer_size = DEFAULT_STRICT_BUFFER_SIZE;

   self->adapter = gst_adapter_new ();

   self->stream_align =
       gst_audio_stream_align_new (48000, DEFAULT_ALIGNMENT_THRESHOLD,
       DEFAULT_DISCONT_WAIT);
 }

 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;
   }

   if (self->stream_align) {
     gst_audio_stream_align_free (self->stream_align);
     self->stream_align = NULL;
   }

   G_OBJECT_CLASS (parent_class)->finalize (object);
 }

 static gboolean
 gst_audio_buffer_split_update_samples_per_buffer (GstAudioBufferSplit * self)
 {
   gboolean ret = TRUE;

   GST_OBJECT_LOCK (self);

   /* For a later time */
   if (!self->info.finfo
       || GST_AUDIO_INFO_FORMAT (&self->info) == GST_AUDIO_FORMAT_UNKNOWN) {
     self->samples_per_buffer = 0;
     goto out;
   }

   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;
     goto out;
   }

   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, "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);
 out:
   GST_OBJECT_UNLOCK (self);

   return ret;
 }

 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);
       gst_audio_buffer_split_update_samples_per_buffer (self);
       break;
     case PROP_ALIGNMENT_THRESHOLD:
       GST_OBJECT_LOCK (self);
       gst_audio_stream_align_set_alignment_threshold (self->stream_align,
           g_value_get_uint64 (value));
       GST_OBJECT_UNLOCK (self);
       break;
     case PROP_DISCONT_WAIT:
       GST_OBJECT_LOCK (self);
       gst_audio_stream_align_set_discont_wait (self->stream_align,
           g_value_get_uint64 (value));
       GST_OBJECT_UNLOCK (self);
       break;
     case PROP_STRICT_BUFFER_SIZE:
       self->strict_buffer_size = g_value_get_boolean (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:
       GST_OBJECT_LOCK (self);
       g_value_set_uint64 (value,
           gst_audio_stream_align_get_alignment_threshold (self->stream_align));
       GST_OBJECT_UNLOCK (self);
       break;
     case PROP_DISCONT_WAIT:
       GST_OBJECT_LOCK (self);
       g_value_set_uint64 (value,
           gst_audio_stream_align_get_discont_wait (self->stream_align));
       GST_OBJECT_UNLOCK (self);
       break;
     case PROP_STRICT_BUFFER_SIZE:
       g_value_set_boolean (value, self->strict_buffer_size);
       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);
       GST_OBJECT_LOCK (self);
       gst_audio_stream_align_mark_discont (self->stream_align);
       GST_OBJECT_UNLOCK (self);
       self->current_offset = -1;
       self->accumulated_error = 0;
       self->samples_per_buffer = 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);
       GST_OBJECT_LOCK (self);
       gst_audio_stream_align_mark_discont (self->stream_align);
       GST_OBJECT_UNLOCK (self);
       break;
     default:
       break;
   }

   return state_ret;
 }

 static GstFlowReturn
 gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force,
     gint rate, gint bpf, guint samples_per_buffer)
 {
   gint size, avail;
   GstFlowReturn ret = GST_FLOW_OK;
   GstClockTime resync_time;

   resync_time = self->resync_time;
   size = 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);

     /* After a reset we have to set the discont flag */
     if (self->current_offset == 0)
       GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);

     resync_time_diff =
         gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
     if (self->segment.rate < 0.0) {
       if (resync_time > resync_time_diff)
         GST_BUFFER_TIMESTAMP (buffer) = 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) = 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) - 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;

     /* Update the size based on the accumulated error we have now after
      * taking out a buffer. Same code as above */
     size = samples_per_buffer * bpf;
     if (self->error_per_buffer + self->accumulated_error >=
         self->output_buffer_duration_d)
       size += bpf;
   }

   return ret;
 }

 static GstFlowReturn
 gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
     GstBuffer * buffer, gint rate, gint bpf, guint samples_per_buffer)
 {
   gboolean discont;
   GstFlowReturn ret = GST_FLOW_OK;

   GST_OBJECT_LOCK (self);
   discont =
       gst_audio_stream_align_process (self->stream_align,
       self->segment.rate < 0 ? FALSE : GST_BUFFER_IS_DISCONT (buffer)
       || GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_RESYNC),
       GST_BUFFER_PTS (buffer), gst_buffer_get_size (buffer) / bpf, NULL, NULL,
       NULL);
   GST_OBJECT_UNLOCK (self);

   if (discont) {
     if (self->strict_buffer_size) {
       gst_adapter_clear (self->adapter);
       ret = GST_FLOW_OK;
     } else {
       ret =
           gst_audio_buffer_split_output (self, TRUE, rate, bpf,
           samples_per_buffer);
     }

     self->current_offset = 0;
     self->accumulated_error = 0;
     self->resync_time = GST_BUFFER_PTS (buffer);
   }

   return ret;
 }

 static GstBuffer *
 gst_audio_buffer_split_clip_buffer (GstAudioBufferSplit * self,
     GstBuffer * buffer, const GstSegment * segment, gint rate, gint bpf)
 {
   return gst_audio_buffer_clip (buffer, segment, rate, bpf);
 }

 static GstFlowReturn
 gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
     GstBuffer * buffer)
 {
   GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
   GstFlowReturn ret;
   GstAudioFormat format;
   gint rate, bpf, samples_per_buffer;

   GST_OBJECT_LOCK (self);
   format =
       self->info.
       finfo ? GST_AUDIO_INFO_FORMAT (&self->info) : GST_AUDIO_FORMAT_UNKNOWN;
   rate = GST_AUDIO_INFO_RATE (&self->info);
   bpf = GST_AUDIO_INFO_BPF (&self->info);
   samples_per_buffer = self->samples_per_buffer;
   GST_OBJECT_UNLOCK (self);

   if (format == GST_AUDIO_FORMAT_UNKNOWN || samples_per_buffer == 0) {
     gst_buffer_unref (buffer);
     return GST_FLOW_NOT_NEGOTIATED;
   }

   buffer =
       gst_audio_buffer_split_clip_buffer (self, buffer, &self->segment, rate,
       bpf);
   if (!buffer)
     return GST_FLOW_OK;

   ret =
       gst_audio_buffer_split_handle_discont (self, buffer, rate, bpf,
       samples_per_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, rate, bpf,
       samples_per_buffer);
 }

 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;
       GstAudioInfo info;

       gst_event_parse_caps (event, &caps);

       ret = gst_audio_info_from_caps (&info, caps);
       if (ret) {
         GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps);

         if (!gst_audio_info_is_equal (&info, &self->info)) {
           if (self->strict_buffer_size) {
             gst_adapter_clear (self->adapter);
           } else {
             GstAudioFormat format;
             gint rate, bpf, samples_per_buffer;

             GST_OBJECT_LOCK (self);
             format =
                 self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
                 GST_AUDIO_FORMAT_UNKNOWN;
             rate = GST_AUDIO_INFO_RATE (&self->info);
             bpf = GST_AUDIO_INFO_BPF (&self->info);
             samples_per_buffer = self->samples_per_buffer;
             GST_OBJECT_UNLOCK (self);

             if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
               gst_audio_buffer_split_output (self, TRUE, rate, bpf,
                   samples_per_buffer);
           }
         }
         self->info = info;
         GST_OBJECT_LOCK (self);
         gst_audio_stream_align_set_rate (self->stream_align, self->info.rate);
         GST_OBJECT_UNLOCK (self);
         ret = gst_audio_buffer_split_update_samples_per_buffer (self);
       } 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);
       GST_OBJECT_LOCK (self);
       gst_audio_stream_align_mark_discont (self->stream_align);
       GST_OBJECT_UNLOCK (self);
       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:
       if (self->strict_buffer_size) {
         gst_adapter_clear (self->adapter);
       } else {
         GstAudioFormat format;
         gint rate, bpf, samples_per_buffer;

         GST_OBJECT_LOCK (self);
         format =
             self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
             GST_AUDIO_FORMAT_UNKNOWN;
         rate = GST_AUDIO_INFO_RATE (&self->info);
         bpf = GST_AUDIO_INFO_BPF (&self->info);
         samples_per_buffer = self->samples_per_buffer;
         GST_OBJECT_UNLOCK (self);

         if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
           gst_audio_buffer_split_output (self, TRUE, rate, bpf,
               samples_per_buffer);
       }
       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)
	/*
	* 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,
	PROP_STRICT_BUFFER_SIZE,
	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 DEFAULT_STRICT_BUFFER_SIZE (FALSE)

	#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));

	g_object_class_install_property (gobject_class, PROP_STRICT_BUFFER_SIZE,
	g_param_spec_boolean ("strict-buffer-size", "Strict buffer size",
	"Discard the last samples at EOS or discont if they are too "
	"small to fill a buffer", DEFAULT_STRICT_BUFFER_SIZE,
	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->srcpad,
	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->strict_buffer_size = DEFAULT_STRICT_BUFFER_SIZE;

	self->adapter = gst_adapter_new ();

	self->stream_align =
	gst_audio_stream_align_new (48000, DEFAULT_ALIGNMENT_THRESHOLD,
	DEFAULT_DISCONT_WAIT);
	}

	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;
	}

	if (self->stream_align) {
	gst_audio_stream_align_free (self->stream_align);
	self->stream_align = NULL;
	}

	G_OBJECT_CLASS (parent_class)->finalize (object);
	}

	static gboolean
	gst_audio_buffer_split_update_samples_per_buffer (GstAudioBufferSplit * self)
	{
	gboolean ret = TRUE;

	GST_OBJECT_LOCK (self);

	/* For a later time */
	if (!self->info.finfo
	\|\| GST_AUDIO_INFO_FORMAT (&self->info) == GST_AUDIO_FORMAT_UNKNOWN) {
	self->samples_per_buffer = 0;
	goto out;
	}

	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;
	goto out;
	}

	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, "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);
	out:
	GST_OBJECT_UNLOCK (self);

	return ret;
	}

	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);
	gst_audio_buffer_split_update_samples_per_buffer (self);
	break;
	case PROP_ALIGNMENT_THRESHOLD:
	GST_OBJECT_LOCK (self);
	gst_audio_stream_align_set_alignment_threshold (self->stream_align,
	g_value_get_uint64 (value));
	GST_OBJECT_UNLOCK (self);
	break;
	case PROP_DISCONT_WAIT:
	GST_OBJECT_LOCK (self);
	gst_audio_stream_align_set_discont_wait (self->stream_align,
	g_value_get_uint64 (value));
	GST_OBJECT_UNLOCK (self);
	break;
	case PROP_STRICT_BUFFER_SIZE:
	self->strict_buffer_size = g_value_get_boolean (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:
	GST_OBJECT_LOCK (self);
	g_value_set_uint64 (value,
	gst_audio_stream_align_get_alignment_threshold (self->stream_align));
	GST_OBJECT_UNLOCK (self);
	break;
	case PROP_DISCONT_WAIT:
	GST_OBJECT_LOCK (self);
	g_value_set_uint64 (value,
	gst_audio_stream_align_get_discont_wait (self->stream_align));
	GST_OBJECT_UNLOCK (self);
	break;
	case PROP_STRICT_BUFFER_SIZE:
	g_value_set_boolean (value, self->strict_buffer_size);
	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);
	GST_OBJECT_LOCK (self);
	gst_audio_stream_align_mark_discont (self->stream_align);
	GST_OBJECT_UNLOCK (self);
	self->current_offset = -1;
	self->accumulated_error = 0;
	self->samples_per_buffer = 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);
	GST_OBJECT_LOCK (self);
	gst_audio_stream_align_mark_discont (self->stream_align);
	GST_OBJECT_UNLOCK (self);
	break;
	default:
	break;
	}

	return state_ret;
	}

	static GstFlowReturn
	gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force,
	gint rate, gint bpf, guint samples_per_buffer)
	{
	gint size, avail;
	GstFlowReturn ret = GST_FLOW_OK;
	GstClockTime resync_time;

	resync_time = self->resync_time;
	size = 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);

	/* After a reset we have to set the discont flag */
	if (self->current_offset == 0)
	GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DISCONT);

	resync_time_diff =
	gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
	if (self->segment.rate < 0.0) {
	if (resync_time > resync_time_diff)
	GST_BUFFER_TIMESTAMP (buffer) = 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) = 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) - 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;

	/* Update the size based on the accumulated error we have now after
	* taking out a buffer. Same code as above */
	size = samples_per_buffer * bpf;
	if (self->error_per_buffer + self->accumulated_error >=
	self->output_buffer_duration_d)
	size += bpf;
	}

	return ret;
	}

	static GstFlowReturn
	gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
	GstBuffer * buffer, gint rate, gint bpf, guint samples_per_buffer)
	{
	gboolean discont;
	GstFlowReturn ret = GST_FLOW_OK;

	GST_OBJECT_LOCK (self);
	discont =
	gst_audio_stream_align_process (self->stream_align,
	self->segment.rate < 0 ? FALSE : GST_BUFFER_IS_DISCONT (buffer)
	\|\| GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_RESYNC),
	GST_BUFFER_PTS (buffer), gst_buffer_get_size (buffer) / bpf, NULL, NULL,
	NULL);
	GST_OBJECT_UNLOCK (self);

	if (discont) {
	if (self->strict_buffer_size) {
	gst_adapter_clear (self->adapter);
	ret = GST_FLOW_OK;
	} else {
	ret =
	gst_audio_buffer_split_output (self, TRUE, rate, bpf,
	samples_per_buffer);
	}

	self->current_offset = 0;
	self->accumulated_error = 0;
	self->resync_time = GST_BUFFER_PTS (buffer);
	}

	return ret;
	}

	static GstBuffer *
	gst_audio_buffer_split_clip_buffer (GstAudioBufferSplit * self,
	GstBuffer * buffer, const GstSegment * segment, gint rate, gint bpf)
	{
	return gst_audio_buffer_clip (buffer, segment, rate, bpf);
	}

	static GstFlowReturn
	gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
	GstBuffer * buffer)
	{
	GstAudioBufferSplit *self = GST_AUDIO_BUFFER_SPLIT (parent);
	GstFlowReturn ret;
	GstAudioFormat format;
	gint rate, bpf, samples_per_buffer;

	GST_OBJECT_LOCK (self);
	format =
	self->info.
	finfo ? GST_AUDIO_INFO_FORMAT (&self->info) : GST_AUDIO_FORMAT_UNKNOWN;
	rate = GST_AUDIO_INFO_RATE (&self->info);
	bpf = GST_AUDIO_INFO_BPF (&self->info);
	samples_per_buffer = self->samples_per_buffer;
	GST_OBJECT_UNLOCK (self);

	if (format == GST_AUDIO_FORMAT_UNKNOWN \|\| samples_per_buffer == 0) {
	gst_buffer_unref (buffer);
	return GST_FLOW_NOT_NEGOTIATED;
	}

	buffer =
	gst_audio_buffer_split_clip_buffer (self, buffer, &self->segment, rate,
	bpf);
	if (!buffer)
	return GST_FLOW_OK;

	ret =
	gst_audio_buffer_split_handle_discont (self, buffer, rate, bpf,
	samples_per_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, rate, bpf,
	samples_per_buffer);
	}

	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;
	GstAudioInfo info;

	gst_event_parse_caps (event, &caps);

	ret = gst_audio_info_from_caps (&info, caps);
	if (ret) {
	GST_DEBUG_OBJECT (self, "Got caps %" GST_PTR_FORMAT, caps);

	if (!gst_audio_info_is_equal (&info, &self->info)) {
	if (self->strict_buffer_size) {
	gst_adapter_clear (self->adapter);
	} else {
	GstAudioFormat format;
	gint rate, bpf, samples_per_buffer;

	GST_OBJECT_LOCK (self);
	format =
	self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
	GST_AUDIO_FORMAT_UNKNOWN;
	rate = GST_AUDIO_INFO_RATE (&self->info);
	bpf = GST_AUDIO_INFO_BPF (&self->info);
	samples_per_buffer = self->samples_per_buffer;
	GST_OBJECT_UNLOCK (self);

	if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
	gst_audio_buffer_split_output (self, TRUE, rate, bpf,
	samples_per_buffer);
	}
	}
	self->info = info;
	GST_OBJECT_LOCK (self);
	gst_audio_stream_align_set_rate (self->stream_align, self->info.rate);
	GST_OBJECT_UNLOCK (self);
	ret = gst_audio_buffer_split_update_samples_per_buffer (self);
	} 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);
	GST_OBJECT_LOCK (self);
	gst_audio_stream_align_mark_discont (self->stream_align);
	GST_OBJECT_UNLOCK (self);
	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:
	if (self->strict_buffer_size) {
	gst_adapter_clear (self->adapter);
	} else {
	GstAudioFormat format;
	gint rate, bpf, samples_per_buffer;

	GST_OBJECT_LOCK (self);
	format =
	self->info.finfo ? GST_AUDIO_INFO_FORMAT (&self->info) :
	GST_AUDIO_FORMAT_UNKNOWN;
	rate = GST_AUDIO_INFO_RATE (&self->info);
	bpf = GST_AUDIO_INFO_BPF (&self->info);
	samples_per_buffer = self->samples_per_buffer;
	GST_OBJECT_UNLOCK (self);

	if (format != GST_AUDIO_FORMAT_UNKNOWN && samples_per_buffer != 0)
	gst_audio_buffer_split_output (self, TRUE, rate, bpf,
	samples_per_buffer);
	}
	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)