path: root/media-libs/gst-plugins-base/files/gst-plugins-base.audioresample.patch

--- gst/audioresample/gstaudioresample.c	2006/10/28 16:00:51	1.22
+++ gst/audioresample/gstaudioresample.c	2007/03/15 10:52:21	1.25
@@ -194,6 +194,8 @@
   gst_pad_set_bufferalloc_function (trans->sinkpad, NULL);
 
   audioresample->filter_length = DEFAULT_FILTERLEN;
+
+  audioresample->need_discont = FALSE;
 }
 
 /* vmethods */
@@ -371,7 +373,7 @@
   gboolean use_internal = FALSE;        /* whether we use the internal state */
   gboolean ret = TRUE;
 
-  GST_DEBUG_OBJECT (base, "asked to transform size %d in direction %s",
+  GST_LOG_OBJECT (base, "asked to transform size %d in direction %s",
       size, direction == GST_PAD_SINK ? "SINK" : "SRC");
   if (direction == GST_PAD_SINK) {
     sinkcaps = caps;
@@ -406,7 +408,7 @@
 
   /* we make room for one extra sample, given that the resampling filter
    * can output an extra one for non-integral i_rate/o_rate */
-  GST_DEBUG_OBJECT (base, "transformed size %d to %d", size, *othersize);
+  GST_LOG_OBJECT (base, "transformed size %d to %d", size, *othersize);
 
   if (!use_internal) {
     resample_free (state);
@@ -492,8 +494,7 @@
   r = audioresample->resample;
 
   outsize = resample_get_output_size (r);
-  GST_DEBUG_OBJECT (audioresample, "audioresample can give me %d bytes",
-      outsize);
+  GST_LOG_OBJECT (audioresample, "audioresample can give me %d bytes", outsize);
 
   /* protect against mem corruption */
   if (outsize > GST_BUFFER_SIZE (outbuf)) {
@@ -540,8 +541,8 @@
   /* check for possible mem corruption */
   if (outsize > GST_BUFFER_SIZE (outbuf)) {
     /* this is an error that when it happens, would need fixing in the
-     * resample library; we told
-     * it we wanted only GST_BUFFER_SIZE (outbuf), and it gave us more ! */
+     * resample library; we told it we wanted only GST_BUFFER_SIZE (outbuf),
+     * and it gave us more ! */
     GST_WARNING_OBJECT (audioresample,
         "audioresample, you memory corrupting bastard. "
         "you gave me outsize %d while my buffer was size %d",
@@ -556,9 +557,51 @@
   }
   GST_BUFFER_SIZE (outbuf) = outsize;
 
+  if (G_UNLIKELY (audioresample->need_discont)) {
+    GST_DEBUG_OBJECT (audioresample,
+        "marking this buffer with the DISCONT flag");
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+    audioresample->need_discont = FALSE;
+  }
+
+  GST_LOG_OBJECT (audioresample, "transformed to buffer of %ld bytes, ts %"
+      GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
+      G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
+      outsize, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
+      GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
+      GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
+
+
   return GST_FLOW_OK;
 }
 
+/* llabs() is C99, so we might not have it; just use a simple macro... */
+#define LLABS(x) ((x>0)?x:-x)
+static gboolean
+audioresample_check_discont (GstAudioresample * audioresample,
+    GstClockTime timestamp)
+{
+  if (timestamp != GST_CLOCK_TIME_NONE &&
+      audioresample->prev_ts != GST_CLOCK_TIME_NONE &&
+      audioresample->prev_duration != GST_CLOCK_TIME_NONE &&
+      timestamp != audioresample->prev_ts + audioresample->prev_duration) {
+    /* Potentially a discontinuous buffer. However, it turns out that many
+     * elements generate imperfect streams due to rounding errors, so we permit
+     * a small error (up to one sample) without triggering a filter 
+     * flush/restart (if triggered incorrectly, this will be audible) */
+    GstClockTimeDiff diff = timestamp -
+        (audioresample->prev_ts + audioresample->prev_duration);
+
+    if (LLABS (diff) > GST_SECOND / audioresample->i_rate) {
+      GST_WARNING_OBJECT (audioresample,
+          "encountered timestamp discontinuity of %" G_GINT64_FORMAT, diff);
+      return TRUE;
+    }
+  }
+
+  return FALSE;
+}
+
 static GstFlowReturn
 audioresample_transform (GstBaseTransform * base, GstBuffer * inbuf,
     GstBuffer * outbuf)
@@ -576,7 +619,22 @@
   size = GST_BUFFER_SIZE (inbuf);
   timestamp = GST_BUFFER_TIMESTAMP (inbuf);
 
-  GST_DEBUG_OBJECT (audioresample, "got buffer of %ld bytes", size);
+  GST_LOG_OBJECT (audioresample, "transforming buffer of %ld bytes, ts %"
+      GST_TIME_FORMAT ", duration %" GST_TIME_FORMAT ", offset %"
+      G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
+      size, GST_TIME_ARGS (timestamp),
+      GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)),
+      GST_BUFFER_OFFSET (inbuf), GST_BUFFER_OFFSET_END (inbuf));
+
+  /* check for timestamp discontinuities and flush/reset if needed */
+  if (G_UNLIKELY (audioresample_check_discont (audioresample, timestamp))) {
+    /* Flush internal samples */
+    audioresample_pushthrough (audioresample);
+    /* Inform downstream element about discontinuity */
+    audioresample->need_discont = TRUE;
+    /* We want to recalculate the offset */
+    audioresample->ts_offset = -1;
+  }
 
   if (audioresample->ts_offset == -1) {
     /* if we don't know the initial offset yet, calculate it based on the 
@@ -584,19 +642,21 @@
     if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
       GstClockTime stime;
 
-      /* offset used to calculate the timestamps. We use the sample offset for this
-       * to make it more accurate. We want the first buffer to have the same timestamp
-       * as the incomming timestamp. */
+      /* offset used to calculate the timestamps. We use the sample offset for
+       * this to make it more accurate. We want the first buffer to have the
+       * same timestamp as the incoming timestamp. */
       audioresample->next_ts = timestamp;
       audioresample->ts_offset =
           gst_util_uint64_scale_int (timestamp, r->o_rate, GST_SECOND);
-      /* offset used to set as the buffer offset, this offset is always relative
-       * to the stream time, note that timestamp is not... */
+      /* offset used to set as the buffer offset, this offset is always
+       * relative to the stream time, note that timestamp is not... */
       stime = (timestamp - base->segment.start) + base->segment.time;
       audioresample->offset =
           gst_util_uint64_scale_int (stime, r->o_rate, GST_SECOND);
     }
   }
+  audioresample->prev_ts = timestamp;
+  audioresample->prev_duration = GST_BUFFER_DURATION (inbuf);
 
   /* need to memdup, resample takes ownership. */
   datacopy = g_memdup (data, size);
@@ -618,17 +678,25 @@
   r = audioresample->resample;
 
   outsize = resample_get_output_size (r);
-  if (outsize == 0)
+  if (outsize == 0) {
+    GST_DEBUG_OBJECT (audioresample, "no internal buffers needing flush");
     goto done;
+  }
+
+  trans = GST_BASE_TRANSFORM (audioresample);
 
-  outbuf = gst_buffer_new_and_alloc (outsize);
+  res = gst_pad_alloc_buffer (trans->srcpad, GST_BUFFER_OFFSET_NONE, outsize,
+      GST_PAD_CAPS (trans->srcpad), &outbuf);
+  if (G_UNLIKELY (res != GST_FLOW_OK)) {
+    GST_WARNING_OBJECT (audioresample, "failed allocating buffer of %d bytes",
+        outsize);
+    goto done;
+  }
 
   res = audioresample_do_output (audioresample, outbuf);
-  if (res != GST_FLOW_OK)
+  if (G_UNLIKELY (res != GST_FLOW_OK))
     goto done;
 
-  trans = GST_BASE_TRANSFORM (audioresample);
-
   res = gst_pad_push (trans->srcpad, outbuf);
 
 done:
--- gst/audioresample/gstaudioresample.h	2006/06/01 19:19:50	1.6
+++ gst/audioresample/gstaudioresample.h	2007/03/14 17:16:30	1.7
@@ -53,10 +53,12 @@
   GstCaps *srccaps, *sinkcaps;
 
   gboolean passthru;
+  gboolean need_discont;
 
   guint64 offset;
   guint64 ts_offset;
   GstClockTime next_ts;
+  GstClockTime prev_ts, prev_duration;
   int channels;
 
   int i_rate;