Merge commit '4a58223e4c824fedc024af435337a769e8ce593e' into 0.11

ext/alsa/gstalsasrc.c

@@ -197,7 +197,8 @@ static GstClockTime
 gst_alsasrc_get_timestamp (GstAlsaSrc * src)
 {
   snd_pcm_status_t *status;
-  snd_htimestamp_t tstamp;
+  snd_htimestamp_t htstamp;
+  snd_timestamp_t tstamp;
   GstClockTime timestamp;
   snd_pcm_uframes_t availmax;
   gint64 offset;
@@ -221,8 +222,19 @@ gst_alsasrc_get_timestamp (GstAlsaSrc * src)
   }
 
   /* get high resolution time stamp from driver */
-  snd_pcm_status_get_htstamp (status, &tstamp);
+  snd_pcm_status_get_htstamp (status, &htstamp);
-  timestamp = GST_TIMESPEC_TO_TIME (tstamp);
+  timestamp = GST_TIMESPEC_TO_TIME (htstamp);
+  if (timestamp == 0) {
+    GST_INFO_OBJECT (src,
+        "This alsa source does support high resolution timestamps");
+    snd_pcm_status_get_tstamp (status, &tstamp);
+    timestamp = GST_TIMEVAL_TO_TIME (tstamp);
+    if (timestamp == 0) {
+      GST_INFO_OBJECT (src,
+          "This alsa source does support low resolution timestamps");
+      timestamp = gst_util_get_timestamp ();
+    }
+  }
   GST_DEBUG_OBJECT (src, "Base ts: %" GST_TIME_FORMAT,
       GST_TIME_ARGS (timestamp));
   if (timestamp == 0) {

gst-libs/gst/audio/gstaudiodecoder.c

@@ -744,6 +744,27 @@ again:
   return ret;
 }
 
+/**
+ * gst_audio_decoder_finish_frame:
+ * @dec: a #GstAudioDecoder
+ * @buf: decoded data
+ * @frames: number of decoded frames represented by decoded data
+ *
+ * Collects decoded data and pushes it downstream.
+ *
+ * @buf may be NULL in which case the indicated number of frames
+ * are discarded and considered to have produced no output
+ * (e.g. lead-in or setup frames).
+ * Otherwise, source pad caps must be set when it is called with valid
+ * data in @buf.
+ *
+ * Note that a frame received in gst_audio_decoder_handle_frame() may be
+ * invalidated by a call to this function.
+ *
+ * Returns: a #GstFlowReturn that should be escalated to caller (of caller)
+ *
+ * Since: 0.10.36
+ */
 GstFlowReturn
 gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
     gint frames)
@@ -827,15 +848,15 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
     frames--;
   }
 
+  if (G_UNLIKELY (!buf))
+    goto exit;
+
   /* lock on */
   if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (priv->base_ts))) {
     priv->base_ts = ts;
     GST_DEBUG_OBJECT (dec, "base_ts now %" GST_TIME_FORMAT, GST_TIME_ARGS (ts));
   }
 
-  if (G_UNLIKELY (!buf))
-    goto exit;
-
   /* slightly convoluted approach caters for perfect ts if subclass desires */
   if (GST_CLOCK_TIME_IS_VALID (ts)) {
     if (dec->priv->tolerance > 0) {

gst-libs/gst/audio/gstaudiodecoder.h

@@ -174,7 +174,9 @@ struct _GstAudioDecoder
  *                  frames as defined by audio format.
  * @handle_frame:   Provides input data (or NULL to clear any remaining data)
  *                  to subclass.  Input data ref management is performed by
- *                  base class, subclass should not care or intervene.
+ *                  base class, subclass should not care or intervene,
+ *                  and input data is only valid until next call to base class,
+ *                  most notably a call to gst_audio_decoder_finish_frame().
  * @flush:          Optional.
  *                  Instructs subclass to clear any codec caches and discard
  *                  any pending samples and not yet returned encoded data.

gst-libs/gst/audio/gstaudioencoder.c

@@ -465,6 +465,9 @@ gst_audio_encoder_finalize (GObject * object)
  * are considered discarded, e.g. as a result of discontinuous transmission,
  * and a discontinuity is marked.
  *
+ * Note that samples received in gst_audio_encoder_handle_frame()
+ * may be invalidated by a call to this function.
+ *
  * Returns: a #GstFlowReturn that should be escalated to caller (of caller)
  *
  * Since: 0.10.36

gst-libs/gst/audio/gstaudioencoder.h

@@ -133,7 +133,9 @@ struct _GstAudioEncoder {
  * @handle_frame:   Provides input samples (or NULL to clear any remaining data)
  *                  according to directions as configured by the subclass
  *                  using the API.  Input data ref management is performed
- *                  by base class, subclass should not care or intervene.
+ *                  by base class, subclass should not care or intervene,
+ *                  and input data is only valid until next call to base class,
+ *                  most notably a call to gst_audio_encoder_finish_frame().
  * @flush:          Optional.
  *                  Instructs subclass to clear any codec caches and discard
  *                  any pending samples and not yet returned encoded data.