gstreamer 2.2.5-x64-mingw32 → 3.0.0-x64-mingw32

data/vendor/local/include/FLAC/stream_encoder.h

@@ -1,6 +1,6 @@
 /* libFLAC - Free Lossless Audio Codec library
  * Copyright (C) 2000-2009  Josh Coalson
- * Copyright (C) 2011-2013  Xiph.Org Foundation
+ * Copyright (C) 2011-2014  Xiph.Org Foundation
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions
@@ -830,28 +830,28 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_sample_rate(FLAC__StreamEncoder *en
  * The actual values set for each level are:
  * <table>
  * <tr>
- *  <td><b>level</b><td>
- *  <td>do mid-side stereo<td>
- *  <td>loose mid-side stereo<td>
- *  <td>apodization<td>
- *  <td>max lpc order<td>
- *  <td>qlp coeff precision<td>
- *  <td>qlp coeff prec search<td>
- *  <td>escape coding<td>
- *  <td>exhaustive model search<td>
- *  <td>min residual partition order<td>
- *  <td>max residual partition order<td>
- *  <td>rice parameter search dist<td>
+ *  <td><b>level</b></td>
+ *  <td>do mid-side stereo</td>
+ *  <td>loose mid-side stereo</td>
+ *  <td>apodization</td>
+ *  <td>max lpc order</td>
+ *  <td>qlp coeff precision</td>
+ *  <td>qlp coeff prec search</td>
+ *  <td>escape coding</td>
+ *  <td>exhaustive model search</td>
+ *  <td>min residual partition order</td>
+ *  <td>max residual partition order</td>
+ *  <td>rice parameter search dist</td>
  * </tr>
- * <tr>  <td><b>0</b><td>  <td>false<td>  <td>false<td>  <td>tukey(0.5)<td>  <td>0<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>3<td>  <td>0<td>  </tr>
- * <tr>  <td><b>1</b><td>  <td>true<td>   <td>true<td>   <td>tukey(0.5)<td>  <td>0<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>3<td>  <td>0<td>  </tr>
- * <tr>  <td><b>2</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>0<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>3<td>  <td>0<td>  </tr>
- * <tr>  <td><b>3</b><td>  <td>false<td>  <td>false<td>  <td>tukey(0.5)<td>  <td>6<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>4<td>  <td>0<td>  </tr>
- * <tr>  <td><b>4</b><td>  <td>true<td>   <td>true<td>   <td>tukey(0.5)<td>  <td>8<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>4<td>  <td>0<td>  </tr>
- * <tr>  <td><b>5</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>8<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>5<td>  <td>0<td>  </tr>
- * <tr>  <td><b>6</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>8<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>false<td>  <td>0<td>  <td>6<td>  <td>0<td>  </tr>
- * <tr>  <td><b>7</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>8<td>   <td>0<td>  <td>false<td>  <td>false<td>  <td>true<td>   <td>0<td>  <td>6<td>  <td>0<td>  </tr>
- * <tr>  <td><b>8</b><td>  <td>true<td>   <td>false<td>  <td>tukey(0.5)<td>  <td>12<td>  <td>0<td>  <td>false<td>  <td>false<td>  <td>true<td>   <td>0<td>  <td>6<td>  <td>0<td>  </tr>
+ * <tr>  <td><b>0</b></td> <td>false</td> <td>false</td> <td>tukey(0.5)<td>                                     <td>0</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>3</td> <td>0</td> </tr>
+ * <tr>  <td><b>1</b></td> <td>true</td>  <td>true</td>  <td>tukey(0.5)<td>                                     <td>0</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>3</td> <td>0</td> </tr>
+ * <tr>  <td><b>2</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5)<td>                                     <td>0</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>3</td> <td>0</td> </tr>
+ * <tr>  <td><b>3</b></td> <td>false</td> <td>false</td> <td>tukey(0.5)<td>                                     <td>6</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>4</td> <td>0</td> </tr>
+ * <tr>  <td><b>4</b></td> <td>true</td>  <td>true</td>  <td>tukey(0.5)<td>                                     <td>8</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>4</td> <td>0</td> </tr>
+ * <tr>  <td><b>5</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5)<td>                                     <td>8</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>5</td> <td>0</td> </tr>
+ * <tr>  <td><b>6</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5);partial_tukey(2)<td>                    <td>8</td>  <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>6</td> <td>0</td> </tr>
+ * <tr>  <td><b>7</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5);partial_tukey(2)<td>                    <td>12</td> <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>6</td> <td>0</td> </tr>
+ * <tr>  <td><b>8</b></td> <td>true</td>  <td>false</td> <td>tukey(0.5);partial_tukey(2);punchout_tukey(3)</td> <td>12</td> <td>0</td> <td>false</td> <td>false</td> <td>false</td> <td>0</td> <td>6</td> <td>0</td> </tr>
  * </table>
  *
  * \default \c 5
@@ -920,7 +920,8 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_loose_mid_side_stereo(FLAC__StreamE
  * The available functions are \c bartlett, \c bartlett_hann,
  * \c blackman, \c blackman_harris_4term_92db, \c connes, \c flattop,
  * \c gauss(STDDEV), \c hamming, \c hann, \c kaiser_bessel, \c nuttall,
- * \c rectangle, \c triangle, \c tukey(P), \c welch.
+ * \c rectangle, \c triangle, \c tukey(P), \c partial_tukey(n[/ov[/P]]),
+ * \c punchout_tukey(n[/ov[/P]]), \c welch.
  *
  * For \c gauss(STDDEV), STDDEV specifies the standard deviation
  * (0<STDDEV<=0.5).
@@ -929,6 +930,24 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_loose_mid_side_stereo(FLAC__StreamE
  * tapered (0<=P<=1).  P=0 corresponds to \c rectangle and P=1
  * corresponds to \c hann.
  *
+ * Specifying \c partial_tukey or \c punchout_tukey works a little
+ * different. These do not specify a single apodization function, but
+ * a series of them with some overlap. partial_tukey specifies a series
+ * of small windows (all treated separately) while punchout_tukey
+ * specifies a series of windows that have a hole in them. In this way,
+ * the predictor is constructed with only a part of the block, which
+ * helps in case a block consists of dissimilar parts.
+ *
+ * The three parameters that can be specified for the functions are
+ * n, ov and P. n is the number of functions to add, ov is the overlap
+ * of the windows in case of partial_tukey and the overlap in the gaps
+ * in case of punchout_tukey. P is the fraction of the window that is
+ * tapered, like with a regular tukey window. The function can be
+ * specified with only a number, a number and an overlap, or a number
+ * an overlap and a P, for example, partial_tukey(3), partial_tukey(3/0.3)
+ * and partial_tukey(3/0.3/0.5) are all valid. ov should be smaller than 1
+ * and can be negative.
+ *
  * Example specifications are \c "blackman" or
  * \c "hann;triangle;tukey(0.5);tukey(0.25);tukey(0.125)"
  *
@@ -941,7 +960,9 @@ FLAC_API FLAC__bool FLAC__stream_encoder_set_loose_mid_side_stereo(FLAC__StreamE
  * results in the smallest compressed subframe.
  *
  * Note that each function specified causes the encoder to occupy a
- * floating point array in which to store the window.
+ * floating point array in which to store the window. Also note that the
+ * values of P, STDDEV and ov are locale-specific, so if the comma
+ * separator specified by the locale is a comma, a comma should be used.
  *
  * \default \c "tukey(0.5)"
  * \param  encoder        An encoder instance to set.

data/vendor/local/include/gstreamer-1.0/gst/allocators/gstfdmemory.h

@@ -0,0 +1,87 @@
+/* GStreamer fd memory
+ * Copyright (C) 2013 Linaro SA
+ * Author: Benjamin Gaignard <benjamin.gaignard@linaro.org> for Linaro.
+ *
+ * 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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_FD_ALLOCATOR_H__
+#define __GST_FD_ALLOCATOR_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstFdAllocator GstFdAllocator;
+typedef struct _GstFdAllocatorClass GstFdAllocatorClass;
+
+#define GST_ALLOCATOR_FD "fd"
+
+#define GST_TYPE_FD_ALLOCATOR              (gst_fd_allocator_get_type())
+#define GST_IS_FD_ALLOCATOR(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_FD_ALLOCATOR))
+#define GST_IS_FD_ALLOCATOR_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_FD_ALLOCATOR))
+#define GST_FD_ALLOCATOR_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_FD_ALLOCATOR, GstFdAllocatorClass))
+#define GST_FD_ALLOCATOR(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_FD_ALLOCATOR, GstFdAllocator))
+#define GST_FD_ALLOCATOR_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_FD_ALLOCATOR, GstFdAllocatorClass))
+#define GST_FD_ALLOCATOR_CAST(obj)         ((GstFdAllocator *)(obj))
+
+/**
+ * GstFdMemoryFlags:
+ * @GST_FD_MEMORY_FLAG_NONE: no flag
+ * @GST_FD_MEMORY_FLAG_KEEP_MAPPED: once the memory is mapped,
+ *        keep it mapped until the memory is destroyed.
+ * @GST_FD_MEMORY_FLAG_MAP_PRIVATE: do a private mapping instead of
+ *        the default shared mapping.
+ *
+ * Various flags to control the operation of the fd backed memory.
+ *
+ * Since: 1.6
+ */
+typedef enum {
+  GST_FD_MEMORY_FLAG_NONE = 0,
+  GST_FD_MEMORY_FLAG_KEEP_MAPPED = (1 << 0),
+  GST_FD_MEMORY_FLAG_MAP_PRIVATE = (1 << 1),
+} GstFdMemoryFlags;
+
+/**
+ * GstFdAllocator:
+ *
+ * Base class for allocators with fd-backed memory
+ *
+ * Since: 1.6
+ */
+struct _GstFdAllocator
+{
+  GstAllocator parent;
+};
+
+struct _GstFdAllocatorClass
+{
+  GstAllocatorClass parent_class;
+};
+
+GType gst_fd_allocator_get_type (void);
+
+GstAllocator *  gst_fd_allocator_new    (void);
+GstMemory *     gst_fd_allocator_alloc  (GstAllocator * allocator, gint fd,
+                                         gsize size, GstFdMemoryFlags flags);
+
+gboolean        gst_is_fd_memory        (GstMemory *mem);
+gint            gst_fd_memory_get_fd    (GstMemory *mem);
+
+G_END_DECLS
+
+#endif /* __GST_FD_ALLOCATOR_H__ */

data/vendor/local/include/gstreamer-1.0/gst/app/gstappsink.h

@@ -42,6 +42,8 @@ typedef struct _GstAppSink GstAppSink;
 typedef struct _GstAppSinkClass GstAppSinkClass;
 typedef struct _GstAppSinkPrivate GstAppSinkPrivate;
 
+/* FIXME 2.0: Make the instance/class struct private */
+
 /**
  * GstAppSinkCallbacks: (skip)
  * @eos: Called when the end-of-stream has been reached. This callback

data/vendor/local/include/gstreamer-1.0/gst/app/gstappsrc.h

@@ -42,6 +42,8 @@ typedef struct _GstAppSrc GstAppSrc;
 typedef struct _GstAppSrcClass GstAppSrcClass;
 typedef struct _GstAppSrcPrivate GstAppSrcPrivate;
 
+/* FIXME 2.0: Make the instance/class struct private */
+
 /**
  * GstAppSrcCallbacks: (skip)
  * @need_data: Called when the appsrc needs more data. A buffer or EOS should be
@@ -107,9 +109,10 @@ struct _GstAppSrcClass
   /* actions */
   GstFlowReturn (*push_buffer)     (GstAppSrc *appsrc, GstBuffer *buffer);
   GstFlowReturn (*end_of_stream)   (GstAppSrc *appsrc);
+  GstFlowReturn (*push_sample)     (GstAppSrc *appsrc, GstSample *sample);
 
   /*< private >*/
-  gpointer     _gst_reserved[GST_PADDING];
+  gpointer     _gst_reserved[GST_PADDING-1];
 };
 
 GType gst_app_src_get_type(void);
@@ -140,6 +143,7 @@ gboolean         gst_app_src_get_emit_signals        (GstAppSrc *appsrc);
 
 GstFlowReturn    gst_app_src_push_buffer             (GstAppSrc *appsrc, GstBuffer *buffer);
 GstFlowReturn    gst_app_src_end_of_stream           (GstAppSrc *appsrc);
+GstFlowReturn    gst_app_src_push_sample             (GstAppSrc *appsrc, GstSample *sample);
 
 void             gst_app_src_set_callbacks           (GstAppSrc * appsrc,
                                                       GstAppSrcCallbacks *callbacks,

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudiobasesink.h

@@ -59,6 +59,7 @@ G_BEGIN_DECLS
 
 #define GST_TYPE_AUDIO_BASE_SINK                (gst_audio_base_sink_get_type())
 #define GST_AUDIO_BASE_SINK(obj)                (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSink))
+#define GST_AUDIO_BASE_SINK_CAST(obj)           ((GstAudioBaseSink*)obj)
 #define GST_AUDIO_BASE_SINK_CLASS(klass)        (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SINK,GstAudioBaseSinkClass))
 #define GST_AUDIO_BASE_SINK_GET_CLASS(obj)      (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SINK, GstAudioBaseSinkClass))
 #define GST_IS_AUDIO_BASE_SINK(obj)             (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SINK))
@@ -85,6 +86,7 @@ G_BEGIN_DECLS
  * @GST_AUDIO_BASE_SINK_SLAVE_SKEW: Adjust playout pointer when master clock
  * drifts too much.
  * @GST_AUDIO_BASE_SINK_SLAVE_NONE: No adjustment is done.
+ * @GST_AUDIO_BASE_SINK_SLAVE_CUSTOM: Use custom clock slaving algorithm (Since: 1.6)
  *
  * Different possible clock slaving algorithms used when the internal audio
  * clock is not selected as the pipeline master clock.
@@ -93,7 +95,8 @@ typedef enum
 {
   GST_AUDIO_BASE_SINK_SLAVE_RESAMPLE,
   GST_AUDIO_BASE_SINK_SLAVE_SKEW,
-  GST_AUDIO_BASE_SINK_SLAVE_NONE
+  GST_AUDIO_BASE_SINK_SLAVE_NONE,
+  GST_AUDIO_BASE_SINK_SLAVE_CUSTOM
 } GstAudioBaseSinkSlaveMethod;
 
 #define GST_TYPE_AUDIO_BASE_SINK_SLAVE_METHOD (gst_audio_base_sink_slave_method_get_type ())
@@ -102,6 +105,65 @@ typedef struct _GstAudioBaseSink GstAudioBaseSink;
 typedef struct _GstAudioBaseSinkClass GstAudioBaseSinkClass;
 typedef struct _GstAudioBaseSinkPrivate GstAudioBaseSinkPrivate;
 
+/**
+ * GstAudioBaseSinkDiscontReason:
+ * GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT: No discontinuity occurred
+ * GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS: New caps are set, causing renegotiotion
+ * GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH: Samples have been flushed
+ * GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY: Sink was synchronized to the estimated latency (occurs during initialization)
+ * GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT: Aligning buffers failed because the timestamps are too discontinuous
+ * GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE: Audio output device experienced and recovered from an error but introduced latency in the process (see also @gst_audio_base_sink_report_device_failure)
+ *
+ * Different possible reasons for discontinuities. This enum is useful for the custom
+ * slave method.
+ *
+ * Since: 1.6
+ */
+typedef enum
+{
+  GST_AUDIO_BASE_SINK_DISCONT_REASON_NO_DISCONT,
+  GST_AUDIO_BASE_SINK_DISCONT_REASON_NEW_CAPS,
+  GST_AUDIO_BASE_SINK_DISCONT_REASON_FLUSH,
+  GST_AUDIO_BASE_SINK_DISCONT_REASON_SYNC_LATENCY,
+  GST_AUDIO_BASE_SINK_DISCONT_REASON_ALIGNMENT,
+  GST_AUDIO_BASE_SINK_DISCONT_REASON_DEVICE_FAILURE
+} GstAudioBaseSinkDiscontReason;
+
+/**
+ * GstAudioBaseSinkCustomSlavingCallback:
+ * @sink: a #GstAudioBaseSink
+ * @etime: external clock time
+ * @itime: internal clock time
+ * @requested_skew: skew amount requested by the callback
+ * @discont_reason: TRUE if there was a discontinuity in the average skew
+ * @user_data: user data
+ *
+ * This function is set with gst_audio_base_sink_set_custom_slaving_callback()
+ * and is called during playback. It receives the current time of external and
+ * internal clocks, which the callback can then use to apply any custom
+ * slaving/synchronization schemes. The external clock is the sink's element clock,
+ * the internal one is the internal audio clock. The internal audio clock's
+ * calibration is applied to the timestamps before they are passed to the
+ * callback. The difference between etime and itime is the skew; how much
+ * internal and external clock lie apart from each other. A skew of 0 means
+ * both clocks are perfectly in sync. itime > etime means the external clock
+ * is going slower, while itime < etime means it is going faster than the
+ * internal clock. etime and itime are always valid timestamps, except for when
+ * discont is set to TRUE.
+ * requested_skew is an output value the callback can write to. It informs the sink
+ * of whether or not it should move the playout pointer, and if so, by how much.
+ * This pointer is only NULL if discont is true; otherwise, it is safe to write
+ * to *requested_skew. The default skew is 0.
+ *
+ * The sink may experience discontinuities. If one happens, discont is TRUE,
+ * itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL.
+ * This makes it possible to reset custom clock slaving algorithms when a
+ * discontinuity happens.
+ *
+ * Since: 1.6
+ */
+typedef void (*GstAudioBaseSinkCustomSlavingCallback) (GstAudioBaseSink *sink, GstClockTime etime, GstClockTime itime, GstClockTimeDiff *requested_skew, GstAudioBaseSinkDiscontReason discont_reason, gpointer user_data);
+
 /**
  * GstAudioBaseSink:
  *
@@ -186,6 +248,14 @@ void       gst_audio_base_sink_set_discont_wait        (GstAudioBaseSink * sink,
 GstClockTime
            gst_audio_base_sink_get_discont_wait        (GstAudioBaseSink * sink);
 
+void
+gst_audio_base_sink_set_custom_slaving_callback        (GstAudioBaseSink * sink,
+                                                        GstAudioBaseSinkCustomSlavingCallback callback,
+                                                        gpointer user_data,
+                                                        GDestroyNotify notify);
+
+void gst_audio_base_sink_report_device_failure         (GstAudioBaseSink * sink);
+
 G_END_DECLS
 
 #endif /* __GST_AUDIO_BASE_SINK_H__ */

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudiobasesrc.h

@@ -37,6 +37,7 @@ G_BEGIN_DECLS
 
 #define GST_TYPE_AUDIO_BASE_SRC                 (gst_audio_base_src_get_type())
 #define GST_AUDIO_BASE_SRC(obj)                 (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrc))
+#define GST_AUDIO_BASE_SRC_CAST(obj)            ((GstAudioBaseSrc*)obj)
 #define GST_AUDIO_BASE_SRC_CLASS(klass)         (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIO_BASE_SRC,GstAudioBaseSrcClass))
 #define GST_AUDIO_BASE_SRC_GET_CLASS(obj)       (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIO_BASE_SRC, GstAudioBaseSrcClass))
 #define GST_IS_AUDIO_BASE_SRC(obj)              (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_BASE_SRC))

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudiodecoder.h

@@ -231,6 +231,21 @@ struct _GstAudioDecoder
  *                      Propose buffer allocation parameters for upstream elements.
  *                      Subclasses should chain up to the parent implementation to
  *                      invoke the default handler.
+ * @sink_query:     Optional.
+ *                  Query handler on the sink pad. This function should
+ *                  return TRUE if the query could be performed. Subclasses
+ *                  should chain up to the parent implementation to invoke the
+ *                  default handler. Since 1.6
+ * @src_query:      Optional.
+ *                  Query handler on the source pad. This function should
+ *                  return TRUE if the query could be performed. Subclasses
+ *                  should chain up to the parent implementation to invoke the
+ *                  default handler. Since 1.6
+ * @getcaps:        Optional.
+ *                  Allows for a custom sink getcaps implementation.
+ *                  If not implemented,
+ *                  default returns gst_audio_decoder_proxy_getcaps
+ *                  applied to sink template caps.
  *
  * Subclasses can override any of the available virtual methods or not, as
  * needed. At minimum @handle_frame (and likely @set_format) needs to be
@@ -278,8 +293,15 @@ struct _GstAudioDecoderClass
   gboolean      (*propose_allocation) (GstAudioDecoder *dec,
                                        GstQuery * query);
 
+  gboolean      (*sink_query)         (GstAudioDecoder *dec, GstQuery *query);
+
+  gboolean      (*src_query)          (GstAudioDecoder *dec, GstQuery *query);
+
+  GstCaps *     (*getcaps)            (GstAudioDecoder * dec,
+                                       GstCaps * filter);
+
   /*< private >*/
-  gpointer       _gst_reserved[GST_PADDING_LARGE];
+  gpointer       _gst_reserved[GST_PADDING_LARGE - 3];
 };
 
 GType             gst_audio_decoder_get_type (void);
@@ -287,6 +309,9 @@ GType             gst_audio_decoder_get_type (void);
 gboolean          gst_audio_decoder_set_output_format  (GstAudioDecoder    * dec,
                                                         const GstAudioInfo * info);
 
+GstCaps *         gst_audio_decoder_proxy_getcaps (GstAudioDecoder * decoder,
+                                                   GstCaps         * caps,
+                                                   GstCaps         * filter);
 gboolean          gst_audio_decoder_negotiate (GstAudioDecoder * dec);
 
 GstFlowReturn     gst_audio_decoder_finish_frame (GstAudioDecoder * dec,

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudioringbuffer.h

@@ -220,7 +220,7 @@ struct _GstAudioRingBuffer {
  * @pause: pause processing of samples
  * @resume: resume processing of samples after pause
  * @stop: stop processing of samples
- * @delay: get number of samples queued in device
+ * @delay: get number of frames queued in device
  * @activate: activate the thread that starts pulling and monitoring the
  * consumed segments in the device.
  * @commit: write samples into the ringbuffer

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudiosink.h

@@ -66,7 +66,7 @@ struct _GstAudioSink {
  * @unprepare: Undo operations done in prepare.
  * @close: Close the device.
  * @write: Write data to the device.
- * @delay: Return how many samples are still in the device. This is used to
+ * @delay: Return how many frames are still in the device. This is used to
  *         drive the synchronisation.
  * @reset: Returns as quickly as possible from a write and flush any pending
  *         samples from the device.
@@ -88,7 +88,7 @@ struct _GstAudioSinkClass {
   gboolean (*close)     (GstAudioSink *sink);
   /* write samples to the device */
   gint     (*write)     (GstAudioSink *sink, gpointer data, guint length);
-  /* get number of samples queued in the device */
+  /* get number of frames queued in the device */
   guint    (*delay)     (GstAudioSink *sink);
   /* reset the audio device, unblock from a write */
   void     (*reset)     (GstAudioSink *sink);

data/vendor/local/include/gstreamer-1.0/gst/audio/gstaudiosrc.h

@@ -65,7 +65,7 @@ struct _GstAudioSrc {
  * @unprepare: undo the configuration
  * @close: close the device
  * @read: read samples from the audio device
- * @delay: the number of samples queued in the device
+ * @delay: the number of frames queued in the device
  * @reset: unblock a read to the device and reset.
  *
  * #GstAudioSrc class. Override the vmethod to implement
@@ -87,7 +87,7 @@ struct _GstAudioSrcClass {
   /* read samples from the device */
   guint    (*read)      (GstAudioSrc *src, gpointer data, guint length,
       GstClockTime *timestamp);
-  /* get number of samples queued in the device */
+  /* get number of frames queued in the device */
   guint    (*delay)     (GstAudioSrc *src);
   /* reset the audio device, unblock from a write */
   void     (*reset)     (GstAudioSrc *src);

data/vendor/local/include/gstreamer-1.0/gst/base/gstadapter.h

@@ -63,6 +63,7 @@ gpointer                gst_adapter_take                (GstAdapter *adapter, gs
 GstBuffer*              gst_adapter_take_buffer         (GstAdapter *adapter, gsize nbytes);
 GList*                  gst_adapter_take_list           (GstAdapter *adapter, gsize nbytes);
 GstBuffer *             gst_adapter_take_buffer_fast    (GstAdapter *adapter, gsize nbytes);
+GstBufferList *         gst_adapter_take_buffer_list    (GstAdapter *adapter, gsize nbytes);
 gsize                   gst_adapter_available           (GstAdapter *adapter);
 gsize                   gst_adapter_available_fast      (GstAdapter *adapter);
 

data/vendor/local/include/gstreamer-1.0/gst/base/gstbaseparse.h

@@ -156,6 +156,7 @@ typedef struct _GstBaseParsePrivate GstBaseParsePrivate;
  * The opaque #GstBaseParse data structure.
  */
 struct _GstBaseParse {
+  /*< public >*/
   GstElement     element;
 
   /*< protected >*/
@@ -182,8 +183,10 @@ struct _GstBaseParse {
  * @stop:           Optional.
  *                  Called when the element stops processing.
  *                  Allows closing external resources.
- * @set_sink_caps:  allows the subclass to be notified of the actual caps set.
- * @get_sink_caps:  allows the subclass to do its own sink get caps if needed.
+ * @set_sink_caps:  Optional.
+ *                  Allows the subclass to be notified of the actual caps set.
+ * @get_sink_caps:  Optional.
+ *                  Allows the subclass to do its own sink get caps if needed.
  * @handle_frame:   Parses the input data into valid frames as defined by subclass
  *                  which should be passed to gst_base_parse_finish_frame().
  *                  The frame's input buffer is guaranteed writable,
@@ -222,8 +225,7 @@ struct _GstBaseParse {
  *                   parent to let the default handler run (Since 1.2)
  *
  * Subclasses can override any of the available virtual methods or not, as
- * needed. At minimum @check_valid_frame and @parse_frame needs to be
- * overridden.
+ * needed. At minimum @handle_frame needs to be overridden.
  */
 struct _GstBaseParseClass {
   GstElementClass parent_class;

data/vendor/local/include/gstreamer-1.0/gst/base/gstbasesrc.h

@@ -127,7 +127,8 @@ struct _GstBaseSrc {
  * @get_times: Given a buffer, return the start and stop time when it
  *    should be pushed out. The base class will sync on the clock using
  *    these times.
- * @get_size: Return the total size of the resource, in the configured format.
+ * @get_size: Return the total size of the resource, in the format set by
+ *     gst_base_src_set_format().
  * @is_seekable: Check if the source can seek
  * @prepare_seek_segment: Prepare the #GstSegment that will be passed to the
  *   #GstBaseSrcClass.do_seek() vmethod for executing a seek
@@ -190,7 +191,8 @@ struct _GstBaseSrcClass {
   void          (*get_times)    (GstBaseSrc *src, GstBuffer *buffer,
                                  GstClockTime *start, GstClockTime *end);
 
-  /* get the total size of the resource in bytes */
+  /* get the total size of the resource in the format set by
+   * gst_base_src_set_format() */
   gboolean      (*get_size)     (GstBaseSrc *src, guint64 *size);
 
   /* check if the resource is seekable */

data/vendor/local/include/gstreamer-1.0/gst/base/gstbasetransform.h

@@ -90,11 +90,14 @@ struct _GstBaseTransform {
   /* MT-protected (with STREAM_LOCK) */
   gboolean       have_segment;
   GstSegment     segment;
+  /* Default submit_input_buffer places the buffer here,
+   * for consumption by the generate_output method: */
+  GstBuffer      *queued_buf;
 
   /*< private >*/
   GstBaseTransformPrivate *priv;
 
-  gpointer       _gst_reserved[GST_PADDING_LARGE];
+  gpointer       _gst_reserved[GST_PADDING_LARGE-1];
 };
 
 /**
@@ -189,7 +192,23 @@ struct _GstBaseTransform {
  *                  of the outgoing buffer.
  * @transform_ip:   Required if the element operates in-place.
  *                  Transform the incoming buffer in-place.
- *
+ * @submit_input_buffer: Function which accepts a new input buffer and pre-processes it.
+ *                  The default implementation performs caps (re)negotiation, then
+ *                  QoS if needed, and places the input buffer into the @queued_buf
+ *                  member variable. If the buffer is dropped due to QoS, it returns
+ *                  GST_BASE_TRANSFORM_FLOW_DROPPED. If this input buffer is not
+ *                  contiguous with any previous input buffer, then @is_discont
+ *                  is set to #TRUE. (Since 1.6)
+ * @generate_output: Called after each new input buffer is submitted repeatedly
+ *                   until it either generates an error or fails to generate an output
+ *                   buffer. The default implementation takes the contents of the
+ *                   @queued_buf variable, generates an output buffer if needed
+ *                   by calling the class @prepare_output_buffer, and then
+ *                   calls either @transform or @transform_ip. Elements that don't
+ *                   do 1-to-1 transformations on input to output buffers can either
+ *                   return GST_BASE_TRANSFORM_FLOW_DROPPED or simply not generate
+ *                   an output buffer until they are ready to do so. (Since 1.6)
+ *                   
  * Subclasses can override any of the available virtual methods or not, as
  * needed. At minimum either @transform or @transform_ip need to be overridden.
  * If the element can overwrite the input data with the results (data is of the
@@ -258,8 +277,11 @@ struct _GstBaseTransformClass {
                                  GstBuffer *outbuf);
   GstFlowReturn (*transform_ip) (GstBaseTransform *trans, GstBuffer *buf);
 
+  GstFlowReturn (*submit_input_buffer) (GstBaseTransform *trans, gboolean is_discont, GstBuffer *input);
+  GstFlowReturn (*generate_output) (GstBaseTransform *trans, GstBuffer **outbuf);
+
   /*< private >*/
-  gpointer       _gst_reserved[GST_PADDING_LARGE];
+  gpointer       _gst_reserved[GST_PADDING_LARGE - 2];
 };
 
 GType           gst_base_transform_get_type         (void);
@@ -293,6 +315,8 @@ void            gst_base_transform_get_allocator    (GstBaseTransform *trans,
 
 void		gst_base_transform_reconfigure_sink (GstBaseTransform *trans);
 void		gst_base_transform_reconfigure_src  (GstBaseTransform *trans);
+gboolean gst_base_transform_update_src_caps (GstBaseTransform *trans,
+                                             GstCaps *updated_caps);
 G_END_DECLS
 
 #endif /* __GST_BASE_TRANSFORM_H__ */