ext/flac/gstflacenc.*: Handle non-zero start timestamps correctly, mark header packets as

Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_write_callback),
(gst_flac_enc_check_discont), (gst_flac_enc_chain),
(gst_flac_enc_change_state):
* ext/flac/gstflacenc.h:
Handle non-zero start timestamps correctly, mark header packets as
IN_CAPS and print a warning and suggest using audiorate if stream
discontinuities are detected. When FLAC supports flushing the encoder
somehow this should be done for discontinuities instead.
Remove some unused variables from the instance struct.

1 files changed, 62 insertions, 6 deletions

diff --git a/ext/flac/gstflacenc.c b/ext/flac/gstflacenc.c
index 4f290c4a..4440fcca 100644
--- a/ext/flac/gstflacenc.c
+++ b/ext/flac/gstflacenc.c
@@ -17,10 +17,9 @@
  * Boston, MA 02111-1307, USA.
  */
 
-/* FIXME:
- *  - we assume timestamps start from 0 and that we get a perfect stream; we
- *    don't handle non-zero starts and mid-stream discontinuities, esp. not if
- *    we're muxing into ogg
+/* TODO: - We currently don't handle discontinuities in the stream in a useful
+ *         way and instead rely on the developer plugging in audiorate if
+ *         the stream contains discontinuities.
  */
 
 #ifdef HAVE_CONFIG_H
@@ -966,13 +965,14 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
   if (samples > 0 && flacenc->samples_written != (guint64) - 1) {
     guint64 granulepos;
 
-    GST_BUFFER_TIMESTAMP (outbuf) =
+    GST_BUFFER_TIMESTAMP (outbuf) = flacenc->start_ts +
         GST_FRAMES_TO_CLOCK_TIME (flacenc->samples_written,
         flacenc->sample_rate);
     GST_BUFFER_DURATION (outbuf) =
         GST_FRAMES_TO_CLOCK_TIME (samples, flacenc->sample_rate);
     /* offset_end = granulepos for ogg muxer */
-    granulepos = flacenc->samples_written + samples;
+    granulepos =
+        flacenc->granulepos_offset + flacenc->samples_written + samples;
     GST_BUFFER_OFFSET_END (outbuf) = granulepos;
     /* offset = timestamp corresponding to granulepos for ogg muxer
      * (see vorbisenc for a much more elaborate version of this) */
@@ -981,6 +981,10 @@ gst_flac_enc_write_callback (const FLAC__StreamEncoder * encoder,
   } else {
     GST_BUFFER_TIMESTAMP (outbuf) = GST_CLOCK_TIME_NONE;
     GST_BUFFER_DURATION (outbuf) = GST_CLOCK_TIME_NONE;
+    GST_BUFFER_OFFSET (outbuf) =
+        flacenc->samples_written * flacenc->width * flacenc->channels;
+    GST_BUFFER_OFFSET_END (outbuf) = 0;
+    GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_IN_CAPS);
   }
 
   /* we assume libflac passes us stuff neatly framed */
@@ -1110,6 +1114,27 @@ gst_flac_enc_sink_event (GstPad * pad, GstEvent * event)
   return ret;
 }
 
+static gboolean
+gst_flac_enc_check_discont (GstFlacEnc * flacenc, GstClockTime expected,
+    GstClockTime timestamp)
+{
+  guint allowed_diff = GST_SECOND / flacenc->sample_rate / 2;
+
+  if ((timestamp + allowed_diff < expected)
+      || (timestamp > expected + allowed_diff)) {
+    GST_ELEMENT_WARNING (flacenc, STREAM, FORMAT, (NULL),
+        ("Stream discontinuity detected (wanted %" GST_TIME_FORMAT " got %"
+            GST_TIME_FORMAT "). The output will have wrong timestamps,"
+            " consider using audiorate to handle discontinuities"));
+    return TRUE;
+  }
+
+  /* TODO: Do something to handle discontinuities in the stream. The FLAC encoder
+   * unfortunately doesn't have any way to flush it's internal buffers */
+
+  return FALSE;
+}
+
 static GstFlowReturn
 gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer)
 {
@@ -1128,6 +1153,34 @@ gst_flac_enc_chain (GstPad * pad, GstBuffer * buffer)
 
   width = flacenc->width;
 
+  /* Save the timestamp of the first buffer. This will be later
+   * used as offset for all following buffers */
+  if (flacenc->start_ts == GST_CLOCK_TIME_NONE) {
+    if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
+      flacenc->start_ts = GST_BUFFER_TIMESTAMP (buffer);
+      flacenc->granulepos_offset = gst_util_uint64_scale
+          (GST_BUFFER_TIMESTAMP (buffer), flacenc->sample_rate, GST_SECOND);
+    } else {
+      flacenc->start_ts = 0;
+      flacenc->granulepos_offset = 0;
+    }
+  }
+
+  /* Check if we have a continous stream, if not drop some samples or the buffer or
+   * insert some silence samples */
+  if (flacenc->next_ts != GST_CLOCK_TIME_NONE
+      && GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
+    gst_flac_enc_check_discont (flacenc, flacenc->next_ts,
+        GST_BUFFER_TIMESTAMP (buffer));
+  }
+
+  if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)
+      && GST_BUFFER_DURATION_IS_VALID (buffer))
+    flacenc->next_ts =
+        GST_BUFFER_TIMESTAMP (buffer) + GST_BUFFER_DURATION (buffer);
+  else
+    flacenc->next_ts = GST_CLOCK_TIME_NONE;
+
   insize = GST_BUFFER_SIZE (buffer);
   samples = insize / (width >> 3);
 
@@ -1449,6 +1502,9 @@ gst_flac_enc_change_state (GstElement * element, GstStateChange transition)
     case GST_STATE_CHANGE_NULL_TO_READY:
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       flacenc->stopped = FALSE;
+      flacenc->start_ts = GST_CLOCK_TIME_NONE;
+      flacenc->next_ts = GST_CLOCK_TIME_NONE;
+      flacenc->granulepos_offset = 0;
       break;
     case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
     default: