Merge branch 'master' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good

9 files changed, 350 insertions, 46 deletions

diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c
index 017a7f58..adea05fd 100644
--- a/gst/avi/gstavidemux.c
+++ b/gst/avi/gstavidemux.c
@@ -619,9 +619,13 @@ gst_avi_demux_handle_src_event (GstPad * pad, GstEvent * event)
 
   switch (GST_EVENT_TYPE (event)) {
     case GST_EVENT_SEEK:
-      /* handle seeking */
-      res = gst_avi_demux_handle_seek (avi, pad, event);
-      gst_event_unref (event);
+      /* handle seeking only in pull mode */
+      if (!avi->streaming) {
+        res = gst_avi_demux_handle_seek (avi, pad, event);
+        gst_event_unref (event);
+      } else {
+        res = gst_pad_event_default (pad, event);
+      }
       break;
     case GST_EVENT_QOS:
     case GST_EVENT_NAVIGATION:
@@ -4246,6 +4250,7 @@ gst_avi_demux_sink_activate_pull (GstPad * sinkpad, gboolean active)
 
   if (active) {
     avi->segment_running = TRUE;
+    avi->streaming = FALSE;
     return gst_pad_start_task (sinkpad, (GstTaskFunction) gst_avi_demux_loop,
         sinkpad);
   } else {
@@ -4257,9 +4262,11 @@ gst_avi_demux_sink_activate_pull (GstPad * sinkpad, gboolean active)
 static gboolean
 gst_avi_demux_activate_push (GstPad * pad, gboolean active)
 {
+  GstAviDemux *avi = GST_AVI_DEMUX (GST_OBJECT_PARENT (pad));
 
   if (active) {
     GST_DEBUG ("avi: activating push/chain function");
+    avi->streaming = TRUE;
   } else {
     GST_DEBUG ("avi: deactivating push/chain function");
   }
diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c
index 118886fc..50011cff 100644
--- a/gst/matroska/matroska-demux.c
+++ b/gst/matroska/matroska-demux.c
@@ -5135,9 +5135,6 @@ gst_matroska_demux_video_caps (GstMatroskaTrackVideoContext *
       guint rformat;
       guint subformat;
 
-      gst_util_dump_mem (data, size);
-      gst_util_dump_mem (data + 0x1a, size - 0x1a);
-
       subformat = GST_READ_UINT32_BE (data + 0x1a);
       rformat = GST_READ_UINT32_BE (data + 0x1e);
 
@@ -5503,7 +5500,6 @@ gst_matroska_demux_audio_caps (GstMatroskaTrackAudioContext *
       guint extra_data_size;
 
       GST_ERROR ("real audio raversion:%d", raversion);
-      gst_util_dump_mem (data, size);
       if (raversion == 8) {
         /* COOK */
         flavor = GST_READ_UINT16_BE (data + 22);
diff --git a/gst/replaygain/gstrgvolume.c b/gst/replaygain/gstrgvolume.c
index 41fe441d..d569b4bb 100644
--- a/gst/replaygain/gstrgvolume.c
+++ b/gst/replaygain/gstrgvolume.c
@@ -92,7 +92,7 @@ enum
 #define PEAK_FORMAT ".06f"
 
 #define VALID_GAIN(x) ((x) > -60.00 && (x) < 60.00)
-#define VALID_PEAK(x) ((x) > 0.)
+#define VALID_PEAK(x) ((x) > 0. && (x) < 1.)
 
 /* Same template caps as GstVolume, for I don't like having just ANY caps. */
 
diff --git a/gst/rtp/gstrtpvrawdepay.c b/gst/rtp/gstrtpvrawdepay.c
index 35e68a3e..3f599f5d 100644
--- a/gst/rtp/gstrtpvrawdepay.c
+++ b/gst/rtp/gstrtpvrawdepay.c
@@ -139,6 +139,11 @@ gst_rtp_vraw_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
     goto no_height;
   height = atoi (str);
 
+  /* optional interlace value but we don't handle interlaced
+   * formats yet */
+  if ((str = gst_structure_get_string (structure, "interlace")))
+    goto interlaced;
+
   if (!(str = gst_structure_get_string (structure, "sampling")))
     goto no_sampling;
 
@@ -248,6 +253,11 @@ no_height:
     GST_ERROR_OBJECT (depayload, "no height specified");
     return FALSE;
   }
+interlaced:
+  {
+    GST_ERROR_OBJECT (depayload, "interlaced formats not supported yet");
+    return FALSE;
+  }
 no_sampling:
   {
     GST_ERROR_OBJECT (depayload, "no sampling specified");
diff --git a/gst/rtp/gstrtpvrawpay.c b/gst/rtp/gstrtpvrawpay.c
index 19c14c2b..0f0ef7e8 100644
--- a/gst/rtp/gstrtpvrawpay.c
+++ b/gst/rtp/gstrtpvrawpay.c
@@ -211,6 +211,7 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   GstVideoFormat sampling;
   const gchar *depthstr, *samplingstr, *colorimetrystr;
   gchar *wstr, *hstr;
+  gboolean interlaced;
 
   rtpvrawpay = GST_RTP_VRAW_PAY (payload);
 
@@ -229,6 +230,13 @@ gst_rtp_vraw_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
   if (!res)
     goto missing_dimension;
 
+  /* fail on interlaced video for now */
+  if (!gst_structure_get_boolean (s, "interlaced", &interlaced))
+    interlaced = FALSE;
+
+  if (interlaced)
+    goto interlaced;
+
   yp = up = vp = 0;
   xinc = yinc = 1;
 
@@ -358,6 +366,11 @@ unknown_fourcc:
     GST_ERROR_OBJECT (payload, "invalid or missing fourcc");
     return FALSE;
   }
+interlaced:
+  {
+    GST_ERROR_OBJECT (payload, "interlaced video not supported yet");
+    return FALSE;
+  }
 missing_dimension:
   {
     GST_ERROR_OBJECT (payload, "missing width or height property");
@@ -409,7 +422,7 @@ gst_rtp_vraw_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
     GstBuffer *out;
     guint8 *outdata, *headers;
     gboolean next_line;
-    guint length, cont, pixels;
+    guint length, cont, pixels, fieldid;
 
     /* get the max allowed payload length size, we try to fill the complete MTU */
     left = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
@@ -422,6 +435,24 @@ gst_rtp_vraw_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
     GST_LOG_OBJECT (rtpvrawpay, "created buffer of size %u for MTU %u", left,
         mtu);
 
+    /*
+     *   0                   1                   2                   3
+     *   0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+     *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     *  |   Extended Sequence Number    |            Length             |
+     *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     *  |F|          Line No            |C|           Offset            |
+     *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     *  |            Length             |F|          Line No            |
+     *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
+     *  |C|           Offset            |                               .
+     *  +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+                               .
+     *  .                                                               .
+     *  .                 Two (partial) lines of video data             .
+     *  .                                                               .
+     *  +---------------------------------------------------------------+
+     */
+
     /* need 2 bytes for the extended sequence number */
     *outdata++ = 0;
     *outdata++ = 0;
@@ -456,8 +487,12 @@ gst_rtp_vraw_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
       /* write length */
       *outdata++ = (length >> 8) & 0xff;
       *outdata++ = length & 0xff;
+
+      /* always 0 for now */
+      fieldid = 0x00;
+
       /* write line no */
-      *outdata++ = (line >> 8) & 0x7f;
+      *outdata++ = ((line >> 8) & 0x7f) | fieldid;
       *outdata++ = line & 0xff;
 
       if (next_line) {
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
index b3e1df47..7c5ef8a6 100644
--- a/gst/rtsp/gstrtspsrc.c
+++ b/gst/rtsp/gstrtspsrc.c
@@ -146,6 +146,7 @@ enum
 #define DEFAULT_LATENCY_MS       3000
 #define DEFAULT_CONNECTION_SPEED 0
 #define DEFAULT_NAT_METHOD       GST_RTSP_NAT_DUMMY
+#define DEFAULT_DO_RTCP          TRUE
 
 enum
 {
@@ -159,6 +160,7 @@ enum
   PROP_LATENCY,
   PROP_CONNECTION_SPEED,
   PROP_NAT_METHOD,
+  PROP_DO_RTCP,
   PROP_LAST
 };
 
@@ -335,6 +337,19 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
           GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
           G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
 
+  /**
+   * GstRTSPSrc::do-rtcp
+   *
+   * Enable RTCP support. Some old server don't like RTCP and then this property
+   * needs to be set to FALSE.
+   *
+   * Since: 0.10.15
+   */
+  g_object_class_install_property (gobject_class, PROP_DO_RTCP,
+      g_param_spec_boolean ("do-rtcp", "Do RTCP",
+          "Don't send RTCP packets",
+          DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
+
   gstelement_class->change_state = gst_rtspsrc_change_state;
 
   gstbin_class->handle_message = gst_rtspsrc_handle_message;
@@ -454,6 +469,9 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
     case PROP_NAT_METHOD:
       rtspsrc->nat_method = g_value_get_enum (value);
       break;
+    case PROP_DO_RTCP:
+      rtspsrc->do_rtcp = g_value_get_boolean (value);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -502,6 +520,9 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
     case PROP_NAT_METHOD:
       g_value_set_enum (value, rtspsrc->nat_method);
       break;
+    case PROP_DO_RTCP:
+      g_value_set_boolean (value, rtspsrc->do_rtcp);
+      break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
@@ -1942,8 +1963,8 @@ gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
     }
     gst_object_unref (template);
   }
-  /* setup RTCP transport back to the server */
-  if (src->session) {
+  /* setup RTCP transport back to the server if we have to. */
+  if (src->session && src->do_rtcp) {
     GstPad *pad;
 
     template = gst_static_pad_template_get (&anysinktemplate);
@@ -2162,7 +2183,7 @@ gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
   }
   /* it's possible that the server does not want us to send RTCP in which case
    * the port is -1 */
-  if (rtcp_port != -1 && src->session != NULL) {
+  if (rtcp_port != -1 && src->session != NULL && src->do_rtcp) {
     GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
         rtcp_port);
 
@@ -3944,7 +3965,7 @@ gst_rtspsrc_setup_streams (GstRTSPSrc * src)
     GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
         stream->setup_url);
 
-retry:
+  retry:
     /* create a string with all the transports */
     res = gst_rtspsrc_create_transports_string (src, protocols, &transports);
     if (res < 0)
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
index 95dd9869..40a368c6 100644
--- a/gst/rtsp/gstrtspsrc.h
+++ b/gst/rtsp/gstrtspsrc.h
@@ -186,6 +186,7 @@ struct _GstRTSPSrc {
   guint             latency;
   guint             connection_speed;
   GstRTSPNatMethod  nat_method;
+  gboolean          do_rtcp;
 
   /* state */
   GstRTSPState       state;
diff --git a/gst/udp/gstudpnetutils.c b/gst/udp/gstudpnetutils.c
index 07268509..5fa7593a 100644
--- a/gst/udp/gstudpnetutils.c
+++ b/gst/udp/gstudpnetutils.c
@@ -115,12 +115,17 @@ beach:
 int
 gst_udp_set_loop_ttl (int sockfd, gboolean loop, int ttl)
 {
+  socklen_t socklen;
+  struct sockaddr_storage addr;
   int ret = -1;
-
-#if 0
   int l = (loop == FALSE) ? 0 : 1;
 
-  switch (addr->ss_family) {
+  socklen = sizeof (addr);
+  if ((ret = getsockname (sockfd, (struct sockaddr *) &addr, &socklen)) < 0) {
+    return ret;
+  }
+
+  switch (addr.ss_family) {
     case AF_INET:
     {
       if ((ret =
@@ -155,7 +160,6 @@ gst_udp_set_loop_ttl (int sockfd, gboolean loop, int ttl)
       errno = EAFNOSUPPORT;
 #endif
   }
-#endif
   return ret;
 }
 
diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c
index 266c430f..c8e7c756 100644
--- a/gst/wavparse/gstwavparse.c
+++ b/gst/wavparse/gstwavparse.c
@@ -77,6 +77,7 @@ static gboolean gst_wavparse_pad_convert (GstPad * pad,
     gint64 src_value, GstFormat * dest_format, gint64 * dest_value);
 
 static GstFlowReturn gst_wavparse_chain (GstPad * pad, GstBuffer * buf);
+static gboolean gst_wavparse_sink_event (GstPad * pad, GstEvent * event);
 static void gst_wavparse_loop (GstPad * pad);
 static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event);
 
@@ -193,6 +194,8 @@ gst_wavparse_init (GstWavParse * wavparse, GstWavParseClass * g_class)
       GST_DEBUG_FUNCPTR (gst_wavparse_sink_activate_pull));
   gst_pad_set_chain_function (wavparse->sinkpad,
       GST_DEBUG_FUNCPTR (gst_wavparse_chain));
+  gst_pad_set_event_function (wavparse->sinkpad,
+      GST_DEBUG_FUNCPTR (gst_wavparse_sink_event));
   gst_element_add_pad (GST_ELEMENT_CAST (wavparse), wavparse->sinkpad);
 
   /* src, will be created later */
@@ -725,8 +728,35 @@ gst_wavparse_stream_init (GstWavParse * wav)
   return GST_FLOW_OK;
 }
 
-/* This function is used to perform seeks on the element in
- * pull mode.
+static gboolean
+gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos)
+{
+  /* -1 always maps to -1 */
+  if (ts == -1) {
+    *bytepos = -1;
+    return TRUE;
+  }
+
+  /* 0 always maps to 0 */
+  if (ts == 0) {
+    *bytepos = 0;
+    return TRUE;
+  }
+
+  if (wav->bps > 0) {
+    *bytepos = uint64_ceiling_scale (ts, (guint64) wav->bps, GST_SECOND);
+    return TRUE;
+  } else if (wav->fact) {
+    guint64 bps =
+        gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
+    *bytepos = uint64_ceiling_scale (ts, bps, GST_SECOND);
+    return TRUE;
+  }
+
+  return FALSE;
+}
+
+/* This function is used to perform seeks on the element.
  *
  * It also works when event is NULL, in which case it will just
  * start from the last configured segment. This technique is
@@ -783,6 +813,48 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
     stop_type = GST_SEEK_TYPE_SET;
   }
 
+  /* in push mode, we must delegate to upstream */
+  if (wav->streaming) {
+    gboolean res = FALSE;
+
+    /* if streaming not yet started; only prepare initial newsegment */
+    if (!event || wav->state != GST_WAVPARSE_DATA) {
+      if (wav->start_segment)
+        gst_event_unref (wav->start_segment);
+      wav->start_segment =
+          gst_event_new_new_segment (FALSE, wav->segment.rate,
+          wav->segment.format, wav->segment.last_stop, wav->segment.duration,
+          wav->segment.last_stop);
+      res = TRUE;
+    } else {
+      /* convert seek positions to byte positions in data sections */
+      if (format == GST_FORMAT_TIME) {
+        /* should not fail */
+        if (!gst_wavparse_time_to_bytepos (wav, cur, &cur))
+          goto no_position;
+        if (!gst_wavparse_time_to_bytepos (wav, stop, &stop))
+          goto no_position;
+      }
+      /* mind sample boundary and header */
+      if (cur >= 0) {
+        cur -= (cur % wav->bytes_per_sample);
+        cur += wav->datastart;
+      }
+      if (stop >= 0) {
+        stop -= (stop % wav->bytes_per_sample);
+        stop += wav->datastart;
+      }
+      GST_DEBUG_OBJECT (wav, "Pushing BYTE seek rate %g, "
+          "start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT, rate, cur,
+          stop);
+      /* BYTE seek event */
+      event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, cur,
+          stop_type, stop);
+      res = gst_pad_push_event (wav->sinkpad, event);
+    }
+    return res;
+  }
+
   /* get flush flag */
   flush = flags & GST_SEEK_FLAG_FLUSH;
 
@@ -832,16 +904,8 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
     /* bring offset to bytes, if the bps is 0, we have the segment in BYTES and
      * we can just copy the last_stop. If not, we use the bps to convert TIME to
      * bytes. */
-    if (wav->bps > 0)
-      wav->offset =
-          uint64_ceiling_scale (seeksegment.last_stop, (guint64) wav->bps,
-          GST_SECOND);
-    else if (wav->fact) {
-      guint64 bps =
-          gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
-      wav->offset =
-          uint64_ceiling_scale (seeksegment.last_stop, bps, GST_SECOND);
-    } else
+    if (!gst_wavparse_time_to_bytepos (wav, seeksegment.last_stop,
+            (gint64 *) & wav->offset))
       wav->offset = seeksegment.last_stop;
     GST_LOG_OBJECT (wav, "offset=%" G_GUINT64_FORMAT, wav->offset);
     wav->offset -= (wav->offset % wav->bytes_per_sample);
@@ -854,14 +918,7 @@ gst_wavparse_perform_seek (GstWavParse * wav, GstEvent * event)
   }
 
   if (stop_type != GST_SEEK_TYPE_NONE) {
-    if (wav->bps > 0)
-      wav->end_offset =
-          uint64_ceiling_scale (stop, (guint64) wav->bps, GST_SECOND);
-    else if (wav->fact) {
-      guint64 bps =
-          gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
-      wav->end_offset = uint64_ceiling_scale (stop, bps, GST_SECOND);
-    } else
+    if (!gst_wavparse_time_to_bytepos (wav, stop, (gint64 *) & wav->end_offset))
       wav->end_offset = stop;
     GST_LOG_OBJECT (wav, "end_offset=%" G_GUINT64_FORMAT, wav->end_offset);
     wav->end_offset -= (wav->end_offset % wav->bytes_per_sample);
@@ -962,6 +1019,12 @@ no_format:
     GST_DEBUG_OBJECT (wav, "unsupported format given, seek aborted.");
     return FALSE;
   }
+no_position:
+  {
+    GST_DEBUG_OBJECT (wav,
+        "Could not determine byte position for desired time");
+    return FALSE;
+  }
 }
 
 /*
@@ -1678,6 +1741,32 @@ iterate_adapter:
 
   if (wav->streaming) {
     guint avail = gst_adapter_available (wav->adapter);
+    guint extra;
+
+    /* flush some bytes if evil upstream sends segment that starts
+     * before data or does is not send sample aligned segment */
+    if (G_LIKELY (wav->offset >= wav->datastart)) {
+      extra = (wav->offset - wav->datastart) % wav->bytes_per_sample;
+    } else {
+      extra = wav->datastart - wav->offset;
+    }
+
+    if (G_UNLIKELY (extra)) {
+      extra = wav->bytes_per_sample - extra;
+      if (extra <= avail) {
+        GST_DEBUG_OBJECT (wav, "flushing %d bytes to sample boundary", extra);
+        gst_adapter_flush (wav->adapter, extra);
+        wav->offset += extra;
+        wav->dataleft -= extra;
+        goto iterate_adapter;
+      } else {
+        GST_DEBUG_OBJECT (wav, "flushing %d bytes", avail);
+        gst_adapter_clear (wav->adapter);
+        wav->offset += avail;
+        wav->dataleft -= avail;
+        return GST_FLOW_OK;
+      }
+    }
 
     if (avail < desired) {
       GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail);
@@ -1927,6 +2016,8 @@ gst_wavparse_chain (GstPad * pad, GstBuffer * buf)
 
       /* fall-through */
     case GST_WAVPARSE_DATA:
+      if (buf && GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT))
+        wav->discont = TRUE;
       if ((ret = gst_wavparse_stream_data (wav)) != GST_FLOW_OK)
         goto done;
       break;
@@ -1937,6 +2028,132 @@ done:
   return ret;
 }
 
+static GstFlowReturn
+gst_wavparse_flush_data (GstWavParse * wav)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+  guint av;
+
+  if ((av = gst_adapter_available (wav->adapter)) > 0) {
+    wav->dataleft = av;
+    wav->end_offset = wav->offset + av;
+    ret = gst_wavparse_stream_data (wav);
+  }
+
+  return ret;
+}
+
+static gboolean
+gst_wavparse_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstWavParse *wav = GST_WAVPARSE (GST_PAD_PARENT (pad));
+  gboolean ret = TRUE;
+
+  GST_LOG_OBJECT (wav, "handling %s event", GST_EVENT_TYPE_NAME (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_NEWSEGMENT:
+    {
+      GstFormat format;
+      gdouble rate, arate;
+      gint64 start, stop, time, offset = 0, end_offset = -1;
+      gboolean update;
+      GstSegment segment;
+
+      /* some debug output */
+      gst_segment_init (&segment, GST_FORMAT_UNDEFINED);
+      gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
+          &start, &stop, &time);
+      gst_segment_set_newsegment_full (&segment, update, rate, arate, format,
+          start, stop, time);
+      GST_DEBUG_OBJECT (wav,
+          "received format %d newsegment %" GST_SEGMENT_FORMAT, format,
+          &segment);
+
+      if (wav->state != GST_WAVPARSE_DATA) {
+        GST_DEBUG_OBJECT (wav, "still starting, eating event");
+        goto exit;
+      }
+
+      /* now we are either committed to TIME or BYTE format,
+       * and we only expect a BYTE segment, e.g. following a seek */
+      if (format == GST_FORMAT_BYTES) {
+        if (start > 0) {
+          offset = start;
+          start -= wav->datastart;
+          start = MAX (start, 0);
+        }
+        if (stop > 0) {
+          end_offset = stop;
+          stop -= wav->datastart;
+          stop = MAX (stop, 0);
+        }
+        if (wav->segment.format == GST_FORMAT_TIME) {
+          guint64 bps = wav->bps;
+
+          /* operating in format TIME, so we can convert */
+          if (!bps && wav->fact)
+            bps =
+                gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact);
+          if (bps) {
+            if (start >= 0)
+              start =
+                  uint64_ceiling_scale (start, GST_SECOND, (guint64) wav->bps);
+            if (stop >= 0)
+              stop =
+                  uint64_ceiling_scale (stop, GST_SECOND, (guint64) wav->bps);
+          }
+        }
+      } else {
+        GST_DEBUG_OBJECT (wav, "unsupported segment format, ignoring");
+        goto exit;
+      }
+
+      /* accept upstream's notion of segment and distribute along */
+      gst_segment_set_newsegment_full (&wav->segment, update, rate, arate,
+          wav->segment.format, start, stop, start);
+      /* also store the newsegment event for the streaming thread */
+      if (wav->start_segment)
+        gst_event_unref (wav->start_segment);
+      wav->start_segment =
+          gst_event_new_new_segment_full (update, rate, arate,
+          wav->segment.format, start, stop, start);
+      GST_DEBUG_OBJECT (wav, "Pushing newseg update %d, rate %g, "
+          "applied rate %g, format %d, start %" G_GINT64_FORMAT ", "
+          "stop %" G_GINT64_FORMAT, update, rate, arate, wav->segment.format,
+          start, stop);
+
+      /* stream leftover data in current segment */
+      gst_wavparse_flush_data (wav);
+      /* and set up streaming thread for next one */
+      wav->offset = offset;
+      wav->end_offset = end_offset;
+      if (wav->end_offset > 0) {
+        wav->dataleft = wav->end_offset - wav->offset;
+      } else {
+        /* infinity; upstream will EOS when done */
+        wav->dataleft = G_MAXUINT64;
+      }
+    exit:
+      gst_event_unref (event);
+      break;
+    }
+    case GST_EVENT_EOS:
+      /* stream leftover data in current segment */
+      gst_wavparse_flush_data (wav);
+      /* fall-through */
+    case GST_EVENT_FLUSH_STOP:
+      gst_adapter_clear (wav->adapter);
+      wav->discont = TRUE;
+      /* fall-through */
+    default:
+      ret = gst_pad_event_default (wav->sinkpad, event);
+      break;
+  }
+
+  return ret;
+}
+
 #if 0
 /* convert and query stuff */
 static const GstFormat *
@@ -2089,6 +2306,8 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
     return FALSE;
   }
 
+  GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
+
   switch (GST_QUERY_TYPE (query)) {
     case GST_QUERY_POSITION:
     {
@@ -2152,20 +2371,31 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query)
     }
     case GST_QUERY_SEEKING:{
       GstFormat fmt;
+      gboolean seekable = FALSE;
 
       gst_query_parse_seeking (query, &fmt, NULL, NULL, NULL);
-      if (fmt == GST_FORMAT_TIME) {
-        gboolean seekable = TRUE;
+      if (fmt == wav->segment.format) {
+        res = TRUE;
+        if (wav->streaming) {
+          GstQuery *q;
 
-        if ((wav->bps == 0) && !wav->fact) {
-          seekable = FALSE;
-        } else if (!gst_wavparse_calculate_duration (wav)) {
-          seekable = FALSE;
+          q = gst_query_new_seeking (GST_FORMAT_BYTES);
+          if ((res = gst_pad_peer_query (wav->sinkpad, q))) {
+            gst_query_parse_seeking (q, &fmt, &seekable, NULL, NULL);
+            GST_LOG_OBJECT (wav, "upstream BYTE seekable %d", seekable);
+          }
+          gst_query_unref (q);
+        } else {
+          GST_LOG_OBJECT (wav, "looping => seekable");
+          seekable = TRUE;
+          res = TRUE;
         }
-        gst_query_set_seeking (query, GST_FORMAT_TIME, seekable,
-            0, wav->duration);
+      } else if (fmt == GST_FORMAT_TIME) {
         res = TRUE;
       }
+      if (res) {
+        gst_query_set_seeking (query, fmt, seekable, 0, wav->segment.duration);
+      }
       break;
     }
     default: