[gstreamer-bugs] [Bug 586570] New: Add GAP Flag support to audioresample

[GStreamer (bugzilla.gnome.org)]

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  GStreamer | gst-plugins-base | Ver: git
           Summary: Add GAP Flag support to audioresample
           Product: GStreamer
           Version: git
          Platform: Other
        OS/Version: Linux
            Status: UNCONFIRMED
          Severity: normal
          Priority: Normal
         Component: gst-plugins-base
        AssignedTo: gstreamer-bugs at lists.sourceforge.net
        ReportedBy: channa at caltech.edu
         QAContact: gstreamer-bugs at lists.sourceforge.net
                CC: kcannon at ligo.caltech.edu
     GNOME version: Unspecified
   GNOME milestone: Unspecified


The machinery for audioresample to support the GAP flag is basically there
because it subclasses the GSTBaseTransform.  However this one is a bit more
tricky than others.  Care must be taken to make sure that the correct buffer is
pushed when skipping the actual audioresamper filter during a GAP flag.  I
think this patch does the trick (I have tested it quite a bit).  

diff --git a/gst/audioresample/gstaudioresample.c
b/gst/audioresample/gstaudioresample.c
index 0cae09e..1d5c265 100644
--- a/gst/audioresample/gstaudioresample.c
+++ b/gst/audioresample/gstaudioresample.c
@@ -224,6 +224,7 @@ gst_audio_resample_init (GstAudioResample * resample,

   resample->need_discont = FALSE;

+  gst_base_transform_set_gap_aware (trans, TRUE);
   gst_pad_set_query_function (trans->srcpad, gst_audio_resample_query);
   gst_pad_set_query_type_function (trans->srcpad,
       gst_audio_resample_query_type);
@@ -929,6 +930,15 @@ gst_audio_resample_process (GstAudioResample * resample,
GstBuffer * inbuf,
   gint err = RESAMPLER_ERR_SUCCESS;
   guint8 *in_tmp = NULL, *out_tmp = NULL;
   gboolean need_convert = (resample->funcs->width != resample->width);
+  guint32 in_latency, latency;
+  gboolean skipGAP;
+
+  /* Used to calculate proper behavior when there is a gap 
+   * +1 for downsampling, 0 otherwise */
+  in_latency =
+      resample->funcs->get_input_latency (resample->state) +
+      (resample->inrate >= resample->outrate);
+  latency = in_latency * resample->outrate / resample->inrate;

   in_len = GST_BUFFER_SIZE (inbuf) / resample->channels;
   out_len = GST_BUFFER_SIZE (outbuf) / resample->channels;
@@ -936,6 +946,11 @@ gst_audio_resample_process (GstAudioResample * resample,
GstBuffer * inbuf,
   in_len /= (resample->width / 8);
   out_len /= (resample->width / 8);

+  /* Check if we should skip this gap, if it is shorter than the latency times
+   * it will break so we don't skip that GAP; also check the buffer is long
enough */
+  skipGAP = in_len > in_latency
+      && GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP);
+
   in_processed = in_len;
   out_processed = out_len;

@@ -963,13 +978,20 @@ gst_audio_resample_process (GstAudioResample * resample,
GstBuffer * inbuf,
     }
   }

-  if (need_convert) {
-    err = resample->funcs->process (resample->state,
-        in_tmp, &in_processed, out_tmp, &out_processed);
+  /* don't bother with the resample filter if it is a gap */
+  if (skipGAP) {
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
+    memset (GST_BUFFER_DATA (outbuf), 0, GST_BUFFER_SIZE (outbuf));
+    err = 0;
   } else {
-    err = resample->funcs->process (resample->state,
-        (const guint8 *) GST_BUFFER_DATA (inbuf), &in_processed,
-        (guint8 *) GST_BUFFER_DATA (outbuf), &out_processed);
+    if (need_convert) {
+      err = resample->funcs->process (resample->state,
+          in_tmp, &in_processed, out_tmp, &out_processed);
+    } else {
+      err = resample->funcs->process (resample->state,
+          (const guint8 *) GST_BUFFER_DATA (inbuf), &in_processed,
+          (guint8 *) GST_BUFFER_DATA (outbuf), &out_processed);
+    }
   }

   if (G_UNLIKELY (in_len != in_processed))
@@ -998,6 +1020,12 @@ gst_audio_resample_process (GstAudioResample * resample,
GstBuffer * inbuf,
       gst_audio_resample_convert_buffer (resample, out_tmp,
           GST_BUFFER_DATA (outbuf), out_processed, TRUE);

+    if (skipGAP) {
+      out_processed -= latency;
+      gst_audio_resample_push_drain (resample);
+      gst_audio_resample_reset_state (resample);
+    }
+
     GST_BUFFER_DURATION (outbuf) =
         GST_FRAMES_TO_CLOCK_TIME (out_processed, resample->outrate);
     GST_BUFFER_SIZE (outbuf) =


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