Merge branch 'master' into 0.11

Conflicts:
	gst-libs/gst/audio/gstaudiodecoder.c

diff --cc ext/vorbis/gstvorbisenc.c
Simple merge

diff --cc gst-libs/gst/audio/gstaudiodecoder.c
index c9a1ab6cd09fba552f6aed17b4115afb6c88501c,a45e4e53bb2b0e1b34a46fa3386d1001b859e222..6fb3ebed51bf448c1f87c0976bb3142d8c750374
--- 1/gst-libs/gst/audio/gstaudiodecoder.c
--- 2/gst-libs/gst/audio/gstaudiodecoder.c
+++ b/gst-libs/gst/audio/gstaudiodecoder.c
@@@ -600,32 -570,85 +597,83 @@@ gst_audio_decoder_push_forward (GstAudi
    priv = dec->priv;
    ctx = &dec->priv->ctx;
  
 -      ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
+   g_return_val_if_fail (ctx->info.bpf != 0, GST_FLOW_ERROR);
+ 
+   if (G_UNLIKELY (!buf)) {
+     g_assert_not_reached ();
+     return GST_FLOW_OK;
+   }
+ 
+   GST_LOG_OBJECT (dec, "clipping buffer of size %d with ts %" GST_TIME_FORMAT
 -  gst_buffer_set_caps (buf, GST_PAD_CAPS (dec->srcpad));
 -
++      ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
+       GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+       GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+ 
+   /* clip buffer */
+   buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->info.rate,
+       ctx->info.bpf);
+   if (G_UNLIKELY (!buf)) {
+     GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
+     goto exit;
+   }
+ 
+   /* decorate */
 -    dec->segment.last_stop =
+   if (G_UNLIKELY (priv->discont)) {
+     GST_LOG_OBJECT (dec, "marking discont");
+     GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
+     priv->discont = FALSE;
+   }
+ 
+   /* track where we are */
+   if (G_LIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (buf))) {
+     /* duration should always be valid for raw audio */
+     g_assert (GST_BUFFER_DURATION_IS_VALID (buf));
 -      ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
++    dec->segment.position =
+         GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
+   }
+ 
+   if (klass->pre_push) {
+     /* last chance for subclass to do some dirty stuff */
+     ret = klass->pre_push (dec, &buf);
+     if (ret != GST_FLOW_OK || !buf) {
+       GST_DEBUG_OBJECT (dec, "subclass returned %s, buf %p",
+           gst_flow_get_name (ret), buf);
+       if (buf)
+         gst_buffer_unref (buf);
+       goto exit;
+     }
+   }
+ 
+   GST_LOG_OBJECT (dec, "pushing buffer of size %d with ts %" GST_TIME_FORMAT
++      ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
+       GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
+       GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
+ 
+   ret = gst_pad_push (dec->srcpad, buf);
+ 
+ exit:
+   return ret;
+ }
+ 
+ /* mini aggregator combining output buffers into fewer larger ones,
+  * if so allowed/configured */
+ static GstFlowReturn
+ gst_audio_decoder_output (GstAudioDecoder * dec, GstBuffer * buf)
+ {
+   GstAudioDecoderPrivate *priv;
+   GstFlowReturn ret = GST_FLOW_OK;
+   GstBuffer *inbuf = NULL;
+ 
+   priv = dec->priv;
+ 
    if (G_UNLIKELY (priv->agg < 0))
      gst_audio_decoder_setup (dec);
  
    if (G_LIKELY (buf)) {
-     g_return_val_if_fail (ctx->info.bpf != 0, GST_FLOW_ERROR);
- 
-     GST_LOG_OBJECT (dec,
-         "output buffer of size %" G_GSIZE_FORMAT " with ts %" GST_TIME_FORMAT
+     GST_LOG_OBJECT (dec, "output buffer of size %d with ts %" GST_TIME_FORMAT
 -        ", duration %" GST_TIME_FORMAT, GST_BUFFER_SIZE (buf),
 +        ", duration %" GST_TIME_FORMAT, gst_buffer_get_size (buf),
          GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
          GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
- 
-     /* clip buffer */
-     buf = gst_audio_buffer_clip (buf, &dec->segment, ctx->info.rate,
-         ctx->info.bpf);
-     if (G_UNLIKELY (!buf)) {
-       GST_DEBUG_OBJECT (dec, "no data after clipping to segment");
-     } else {
-       GST_LOG_OBJECT (dec,
-           "buffer after segment clipping has size %" G_GSIZE_FORMAT " with ts %"
-           GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT,
-           gst_buffer_get_size (buf), GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
-           GST_TIME_ARGS (GST_BUFFER_DURATION (buf)));
-     }
-   } else {
-     GST_DEBUG_OBJECT (dec, "no output buffer");
    }
  
  again:

diff --cc gst-libs/gst/audio/gstaudioencoder.c
index 485087228f2567e4cc3b28bc6c339814312c6eff,14064f03bd74838d38ff127b435dd15c51c2c5a3..81f61b5f1b2a1c2a4fd7e850c8c55e6627d813f4
--- 1/gst-libs/gst/audio/gstaudioencoder.c
--- 2/gst-libs/gst/audio/gstaudioencoder.c
+++ b/gst-libs/gst/audio/gstaudioencoder.c
@@@ -1082,11 -1071,15 +1082,18 @@@ gst_audio_encoder_sink_setcaps (GstAudi
      GST_OBJECT_UNLOCK (enc);
  
      if (klass->set_format)
 -      res = klass->set_format (enc, state);
 +      res = klass->set_format (enc, &state);
 +
 +    if (res)
 +      ctx->info = state;
  
 -      gst_audio_info_clear (state);
+     /* invalidate state to ensure no casual carrying on */
+     if (!res) {
+       GST_DEBUG_OBJECT (enc, "subclass did not accept format");
++      gst_audio_info_init (&state);
+       goto exit;
+     }
+ 
      /* notify if new latency */
      GST_OBJECT_LOCK (enc);
      if ((ctx->min_latency > 0 && ctx->min_latency != old_min_latency) ||

diff --cc gst-libs/gst/rtp/gstrtcpbuffer.h
index 2a9d41ae3d766750917ada9ea6f4f1c5b8790bd4,aec9232f33ee0a54923bdf7ab72f280ce908d9af..da8dce26d11ba9bc7b3e0682de1d3ff8e17b4862
--- 1/gst-libs/gst/rtp/gstrtcpbuffer.h
--- 2/gst-libs/gst/rtp/gstrtcpbuffer.h
+++ b/gst-libs/gst/rtp/gstrtcpbuffer.h
@@@ -82,10 -92,15 +92,15 @@@ typedef enu
    GST_RTCP_PSFB_TYPE_PLI      = 1,
    GST_RTCP_PSFB_TYPE_SLI      = 2,
    GST_RTCP_PSFB_TYPE_RPSI     = 3,
-   GST_RTCP_PSFB_TYPE_AFB      = 15
+   GST_RTCP_PSFB_TYPE_AFB      = 15,
+   /* PSFB types assigned in RFC 5104 */
+   GST_RTCP_PSFB_TYPE_FIR      = 4,
+   GST_RTCP_PSFB_TYPE_TSTR     = 5,
+   GST_RTCP_PSFB_TYPE_TSTN     = 6,
+   GST_RTCP_PSFB_TYPE_VBCN     = 7,
  } GstRTCPFBType;
  
 -/** 
 +/**
   * GstRTCPSDESType:
   * @GST_RTCP_SDES_INVALID: Invalid SDES entry
   * @GST_RTCP_SDES_END: End of SDES list