Merge branch 'master' into 0.11

Conflicts:
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/pbutils/gstdiscoverer.c

gst-libs/gst/audio/gstaudiodecoder.c

@@ -177,6 +177,8 @@ enum
 #define DEFAULT_LATENCY    0
 #define DEFAULT_TOLERANCE  0
 #define DEFAULT_PLC        FALSE
+#define DEFAULT_DRAINABLE  TRUE
+#define DEFAULT_NEEDS_FORMAT  FALSE
 
 typedef struct _GstAudioDecoderContext
 {
@@ -258,6 +260,8 @@ struct _GstAudioDecoderPrivate
   GstClockTime latency;
   GstClockTime tolerance;
   gboolean plc;
+  gboolean drainable;
+  gboolean needs_format;
 
   /* pending serialized sink events, will be sent from finish_frame() */
   GList *pending_events;
@@ -420,6 +424,8 @@ gst_audio_decoder_init (GstAudioDecoder * dec, GstAudioDecoderClass * klass)
   dec->priv->latency = DEFAULT_LATENCY;
   dec->priv->tolerance = DEFAULT_TOLERANCE;
   dec->priv->plc = DEFAULT_PLC;
+  dec->priv->drainable = DEFAULT_DRAINABLE;
+  dec->priv->needs_format = DEFAULT_NEEDS_FORMAT;
 
   /* init state */
   gst_audio_decoder_reset (dec, TRUE);
@@ -1049,6 +1055,7 @@ gst_audio_decoder_push_buffers (GstAudioDecoder * dec, gboolean force)
           break;
         } else if (ret == GST_FLOW_OK) {
           GST_LOG_OBJECT (dec, "frame at offset %d of length %d", offset, len);
+          g_assert (len);
           g_assert (offset + len <= av);
           priv->sync_flush = 0;
         } else {
@@ -1072,6 +1079,10 @@ gst_audio_decoder_push_buffers (GstAudioDecoder * dec, gboolean force)
     } else {
       if (!force)
         break;
+      if (!priv->drainable) {
+        priv->drained = TRUE;
+        break;
+      }
       buffer = NULL;
     }
 
@@ -1392,6 +1403,9 @@ gst_audio_decoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
 
   dec = GST_AUDIO_DECODER (parent);
 
+  if (G_UNLIKELY (!gst_pad_has_current_caps (pad) && dec->priv->needs_format))
+    goto not_negotiated;
+
   GST_LOG_OBJECT (dec,
       "received buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
       ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buffer),
@@ -1429,6 +1443,15 @@ gst_audio_decoder_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
   GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
 
   return ret;
+
+  /* ERRORS */
+not_negotiated:
+  {
+    GST_ELEMENT_ERROR (dec, CORE, NEGOTIATION, (NULL),
+        ("decoder not initialized"));
+    gst_buffer_unref (buffer);
+    return GST_FLOW_NOT_NEGOTIATED;
+  }
 }
 
 /* perform upstream byte <-> time conversion (duration, seeking)
@@ -2476,3 +2499,105 @@ gst_audio_decoder_get_tolerance (GstAudioDecoder * dec)
 
   return result;
 }
+
+/**
+ * gst_audio_decoder_set_drainable:
+ * @enc: a #GstAudioDecoder
+ * @enabled: new state
+ *
+ * Configures decoder drain handling.  If drainable, subclass might
+ * be handed a NULL buffer to have it return any leftover decoded data.
+ * Otherwise, it is not considered so capable and will only ever be passed
+ * real data.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+void
+gst_audio_decoder_set_drainable (GstAudioDecoder * dec, gboolean enabled)
+{
+  g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
+
+  GST_OBJECT_LOCK (dec);
+  dec->priv->drainable = enabled;
+  GST_OBJECT_UNLOCK (dec);
+}
+
+/**
+ * gst_audio_decoder_get_drainable:
+ * @enc: a #GstAudioDecoder
+ *
+ * Queries decoder drain handling.
+ *
+ * Returns: TRUE if drainable handling is enabled.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+gboolean
+gst_audio_decoder_get_drainable (GstAudioDecoder * dec)
+{
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
+
+  GST_OBJECT_LOCK (dec);
+  result = dec->priv->drainable;
+  GST_OBJECT_UNLOCK (dec);
+
+  return result;
+}
+
+/**
+ * gst_audio_decoder_set_needs_format:
+ * @enc: a #GstAudioDecoder
+ * @enabled: new state
+ *
+ * Configures decoder format needs.  If enabled, subclass needs to be
+ * negotiated with format caps before it can process any data.  It will then
+ * never be handed any data before it has been configured.
+ * Otherwise, it might be handed data without having been configured and
+ * is then expected being able to do so either by default
+ * or based on the input data.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+void
+gst_audio_decoder_set_needs_format (GstAudioDecoder * dec, gboolean enabled)
+{
+  g_return_if_fail (GST_IS_AUDIO_DECODER (dec));
+
+  GST_OBJECT_LOCK (dec);
+  dec->priv->needs_format = enabled;
+  GST_OBJECT_UNLOCK (dec);
+}
+
+/**
+ * gst_audio_decoder_get_needs_format:
+ * @enc: a #GstAudioDecoder
+ *
+ * Queries decoder required format handling.
+ *
+ * Returns: TRUE if required format handling is enabled.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+gboolean
+gst_audio_decoder_get_needs_format (GstAudioDecoder * dec)
+{
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_AUDIO_DECODER (dec), 0);
+
+  GST_OBJECT_LOCK (dec);
+  result = dec->priv->needs_format;
+  GST_OBJECT_UNLOCK (dec);
+
+  return result;
+}

gst-libs/gst/audio/gstaudiodecoder.h

@@ -296,6 +296,16 @@ void              gst_audio_decoder_set_tolerance   (GstAudioDecoder * dec,
 
 gint64            gst_audio_decoder_get_tolerance   (GstAudioDecoder * dec);
 
+void              gst_audio_decoder_set_drainable (GstAudioDecoder * dec,
+                                                   gboolean enabled);
+
+gboolean          gst_audio_decoder_get_drainable (GstAudioDecoder * dec);
+
+void              gst_audio_decoder_set_needs_format (GstAudioDecoder * dec,
+                                                      gboolean enabled);
+
+gboolean          gst_audio_decoder_get_needs_format (GstAudioDecoder * dec);
+
 G_END_DECLS
 
 #endif /* _GST_AUDIO_DECODER_H_ */

gst-libs/gst/audio/gstaudioencoder.c

@@ -180,6 +180,8 @@ enum
 #define DEFAULT_GRANULE      FALSE
 #define DEFAULT_HARD_RESYNC  FALSE
 #define DEFAULT_TOLERANCE    40000000
+#define DEFAULT_HARD_MIN     FALSE
+#define DEFAULT_DRAINABLE    TRUE
 
 typedef struct _GstAudioEncoderContext
 {
@@ -239,6 +241,8 @@ struct _GstAudioEncoderPrivate
   gboolean perfect_ts;
   gboolean hard_resync;
   gboolean granule;
+  gboolean hard_min;
+  gboolean drainable;
 
   /* pending tags */
   GstTagList *tags;
@@ -396,6 +400,8 @@ gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
   enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
   enc->priv->hard_resync = DEFAULT_HARD_RESYNC;
   enc->priv->tolerance = DEFAULT_TOLERANCE;
+  enc->priv->hard_min = DEFAULT_HARD_MIN;
+  enc->priv->drainable = DEFAULT_DRAINABLE;
 
   /* init state */
   gst_audio_encoder_reset (enc, TRUE);
@@ -769,13 +775,17 @@ gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
       }
     }
 
-    if (need) {
+    priv->got_data = FALSE;
+    if (G_LIKELY (need)) {
       const guint8 *data;
 
       data = gst_adapter_map (priv->adapter, priv->offset + need);
       buf =
           gst_buffer_new_wrapped_full ((gpointer) data, NULL, priv->offset,
           need);
+    } else if (!priv->drainable) {
+      GST_DEBUG_OBJECT (enc, "non-drainable and no more data");
+      goto finish;
     }
 
     GST_LOG_OBJECT (enc, "providing subclass with %d bytes at offset %d",
@@ -786,14 +796,21 @@ gst_audio_encoder_push_buffers (GstAudioEncoder * enc, gboolean force)
     priv->offset += need;
     priv->samples_in += need / ctx->info.bpf;
 
-    priv->got_data = FALSE;
+    /* subclass might not want to be bothered with leftover data,
-    ret = klass->handle_frame (enc, buf);
+     * so take care of that here if so, otherwise pass along */
+    if (G_UNLIKELY (priv->force && priv->hard_min && buf)) {
+      GST_DEBUG_OBJECT (enc, "bypassing subclass with leftover");
+      ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
+    } else {
+      ret = klass->handle_frame (enc, buf);
+    }
 
     if (G_LIKELY (buf)) {
       gst_buffer_unref (buf);
       gst_adapter_unmap (priv->adapter);
     }
 
+  finish:
     /* no data to feed, no leftover provided, then bail out */
     if (G_UNLIKELY (!buf && !priv->got_data)) {
       priv->drained = TRUE;
@@ -2135,6 +2152,106 @@ gst_audio_encoder_get_tolerance (GstAudioEncoder * enc)
   return result;
 }
 
+/**
+ * gst_audio_encoder_set_hard_min:
+ * @enc: a #GstAudioEncoder
+ * @enabled: new state
+ *
+ * Configures encoder hard minimum handling.  If enabled, subclass
+ * will never be handed less samples than it configured, which otherwise
+ * might occur near end-of-data handling.  Instead, the leftover samples
+ * will simply be discarded.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+void
+gst_audio_encoder_set_hard_min (GstAudioEncoder * enc, gboolean enabled)
+{
+  g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
+
+  GST_OBJECT_LOCK (enc);
+  enc->priv->hard_min = enabled;
+  GST_OBJECT_UNLOCK (enc);
+}
+
+/**
+ * gst_audio_encoder_get_hard_min:
+ * @enc: a #GstAudioEncoder
+ *
+ * Queries encoder hard minimum handling.
+ *
+ * Returns: TRUE if hard minimum handling is enabled.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+gboolean
+gst_audio_encoder_get_hard_min (GstAudioEncoder * enc)
+{
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
+
+  GST_OBJECT_LOCK (enc);
+  result = enc->priv->hard_min;
+  GST_OBJECT_UNLOCK (enc);
+
+  return result;
+}
+
+/**
+ * gst_audio_encoder_set_drainable:
+ * @enc: a #GstAudioEncoder
+ * @enabled: new state
+ *
+ * Configures encoder drain handling.  If drainable, subclass might
+ * be handed a NULL buffer to have it return any leftover encoded data.
+ * Otherwise, it is not considered so capable and will only ever be passed
+ * real data.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+void
+gst_audio_encoder_set_drainable (GstAudioEncoder * enc, gboolean enabled)
+{
+  g_return_if_fail (GST_IS_AUDIO_ENCODER (enc));
+
+  GST_OBJECT_LOCK (enc);
+  enc->priv->drainable = enabled;
+  GST_OBJECT_UNLOCK (enc);
+}
+
+/**
+ * gst_audio_encoder_get_drainable:
+ * @enc: a #GstAudioEncoder
+ *
+ * Queries encoder drain handling.
+ *
+ * Returns: TRUE if drainable handling is enabled.
+ *
+ * MT safe.
+ *
+ * Since: 0.10.36
+ */
+gboolean
+gst_audio_encoder_get_drainable (GstAudioEncoder * enc)
+{
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_AUDIO_ENCODER (enc), 0);
+
+  GST_OBJECT_LOCK (enc);
+  result = enc->priv->drainable;
+  GST_OBJECT_UNLOCK (enc);
+
+  return result;
+}
+
 /**
  * gst_audio_encoder_merge_tags:
  * @enc: a #GstAudioEncoder

gst-libs/gst/audio/gstaudioencoder.h

@@ -250,6 +250,16 @@ void            gst_audio_encoder_set_tolerance (GstAudioEncoder * enc,
 
 gint64          gst_audio_encoder_get_tolerance (GstAudioEncoder * enc);
 
+void            gst_audio_encoder_set_hard_min (GstAudioEncoder * enc,
+                                                gboolean enabled);
+
+gboolean        gst_audio_encoder_get_hard_min (GstAudioEncoder * enc);
+
+void            gst_audio_encoder_set_drainable (GstAudioEncoder * enc,
+                                                 gboolean enabled);
+
+gboolean        gst_audio_encoder_get_drainable (GstAudioEncoder * enc);
+
 void            gst_audio_encoder_merge_tags (GstAudioEncoder * enc,
                                               const GstTagList * tags, GstTagMergeMode mode);
 

gst-libs/gst/pbutils/gstdiscoverer.c

@@ -1024,6 +1024,31 @@ discoverer_collect (GstDiscoverer * dc)
       if (gst_element_query_duration (pipeline, GST_FORMAT_TIME, &dur)) {
         GST_DEBUG ("Got duration %" GST_TIME_FORMAT, GST_TIME_ARGS (dur));
         dc->priv->current_info->duration = (guint64) dur;
+      } else {
+        GstStateChangeReturn sret;
+
+        /* Some parsers may not even return a rough estimate right away, e.g.
+         * because they've only processed a single frame so far, so if we
+         * didn't get a duration the first time, spin a bit and try again.
+         * Ugly, but still better than making parsers or other elements return
+         * completely bogus values. We need some API extensions to solve this
+         * better. */
+        GST_INFO ("No duration yet, try a bit harder..");
+        sret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
+        if (sret != GST_STATE_CHANGE_FAILURE) {
+          int i;
+
+          for (i = 0; i < 2; ++i) {
+            g_usleep (G_USEC_PER_SEC / 20);
+            if (gst_element_query_duration (pipeline, GST_FORMAT_TIME, &dur)
+                && dur > 0) {
+              GST_DEBUG ("Got duration %" GST_TIME_FORMAT, GST_TIME_ARGS (dur));
+              dc->priv->current_info->duration = (guint64) dur;
+              break;
+            }
+          }
+          gst_element_set_state (pipeline, GST_STATE_PAUSED);
+        }
       }
 
       if (dc->priv->seeking_query) {

gst/playback/gstplaybin2.c

@@ -175,13 +175,15 @@
  * <refsect2>
  * <title>Specifying which CD/DVD device to use</title>
  * The device to use for CDs/DVDs needs to be set on the source element
- * playbin creates before it is opened. The only way to do this at the moment
+ * playbin creates before it is opened. The most generic way of doing this
- * is to connect to playbin's "notify::source" signal, which will be emitted
+ * is to connect to playbin's "source-setup" (or "notify::source") signal,
- * by playbin when it has created the source element for a particular URI.
+ * which will be emitted by playbin2 when it has created the source element
- * In the signal callback you can check if the source element has a "device"
+ * for a particular URI. In the signal callback you can check if the source
- * property and set it appropriately. In future ways might be added to specify
+ * element has a "device" property and set it appropriately. In some cases
- * the device as part of the URI, but at the time of writing this is not
+ * the device can also be set as part of the URI, but it depends on the
- * possible yet.
+ * elements involved if this will work or not. For example, for DVD menu
+ * playback, the following syntax might work (if the resindvd plugin is used):
+ * dvd://[/path/to/device]
  * </refsect2>
  * <refsect2>
  * <title>Handling redirects</title>
@@ -196,19 +198,19 @@
  * <refsect2>
  * <title>Examples</title>
  * |[
- * gst-launch -v playbin uri=file:///path/to/somefile.avi
+ * gst-launch -v playbin2 uri=file:///path/to/somefile.avi
  * ]| This will play back the given AVI video file, given that the video and
  * audio decoders required to decode the content are installed. Since no
  * special audio sink or video sink is supplied (not possible via gst-launch),
  * playbin will try to find a suitable audio and video sink automatically
  * using the autoaudiosink and autovideosink elements.
  * |[
- * gst-launch -v playbin uri=cdda://4
+ * gst-launch -v playbin2 uri=cdda://4
  * ]| This will play back track 4 on an audio CD in your disc drive (assuming
  * the drive is detected automatically by the plugin).
  * |[
- * gst-launch -v playbin uri=dvd://1
+ * gst-launch -v playbin2 uri=dvd://
- * ]| This will play back title 1 of a DVD in your disc drive (assuming
+ * ]| This will play back the DVD in your disc drive (assuming
  * the drive is detected automatically by the plugin).
  * </refsect2>
  */