Update docs

[platform/upstream/gst-plugins-good.git] / ext / speex / gstspeexenc.c

diff --git a/ext/speex/gstspeexenc.c b/ext/speex/gstspeexenc.c
index f475cab..4266af6 100644 (file)
--- a/ext/speex/gstspeexenc.c
+++ b/ext/speex/gstspeexenc.c
@@ -13,8 +13,8 @@
  *
  * You should have received a copy of the GNU Library General Public
  * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
  */
 
 /**
@@ -29,7 +29,7 @@
  * <refsect2>
  * <title>Example pipelines</title>
  * |[
- * gst-launch audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
+ * gst-launch-1.0 audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
  * ]| Encode an Ogg/Speex file.
  * </refsect2>
  */
@@ -211,10 +211,8 @@ gst_speex_enc_class_init (GstSpeexEncClass * klass)
           "The last status message", NULL,
           G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
 
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&src_factory));
-  gst_element_class_add_pad_template (gstelement_class,
-      gst_static_pad_template_get (&sink_factory));
+  gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
+  gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
   gst_element_class_set_static_metadata (gstelement_class,
       "Speex audio encoder", "Codec/Encoder/Audio",
       "Encodes audio in Speex format", "Wim Taymans <wim@fluendo.com>");
@@ -242,6 +240,7 @@ gst_speex_enc_init (GstSpeexEnc * enc)
   /* arrange granulepos marking (and required perfect ts) */
   gst_audio_encoder_set_mark_granule (benc, TRUE);
   gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
+  GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
 }
 
 static gboolean
@@ -253,6 +252,7 @@ gst_speex_enc_start (GstAudioEncoder * benc)
   speex_bits_init (&enc->bits);
   enc->tags = gst_tag_list_new_empty ();
   enc->header_sent = FALSE;
+  enc->encoded_samples = 0;
 
   return TRUE;
 }
@@ -269,7 +269,8 @@ gst_speex_enc_stop (GstAudioEncoder * benc)
     enc->state = NULL;
   }
   speex_bits_destroy (&enc->bits);
-  gst_tag_list_free (enc->tags);
+  speex_bits_set_bit_buffer (&enc->bits, NULL, 0);
+  gst_tag_list_unref (enc->tags);
   enc->tags = NULL;
 
   gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
@@ -350,7 +351,7 @@ gst_speex_enc_create_metadata_buffer (GstSpeexEnc * enc)
   GST_DEBUG_OBJECT (enc, "merged   tags = %" GST_PTR_FORMAT, merged_tags);
   comments = gst_tag_list_to_vorbiscomment_buffer (merged_tags, NULL,
       0, "Encoded with GStreamer Speexenc");
-  gst_tag_list_free (merged_tags);
+  gst_tag_list_unref (merged_tags);
 
   GST_BUFFER_OFFSET (comments) = 0;
   GST_BUFFER_OFFSET_END (comments) = 0;
@@ -510,6 +511,9 @@ gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
       }
       break;
     }
+    case GST_EVENT_SEGMENT:
+      enc->encoded_samples = 0;
+      break;
     default:
       break;
   }
@@ -529,6 +533,8 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
   gsize bsize, size;
   GstBuffer *outbuf;
   GstFlowReturn ret = GST_FLOW_OK;
+  GstSegment *segment;
+  GstClockTime duration;
 
   if (G_LIKELY (buf)) {
     gst_buffer_map (buf, &map, GST_MAP_READ);
@@ -538,6 +544,28 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
     if (G_UNLIKELY (bsize % bytes)) {
       GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
 
+      /* If encoding part of a frame, and we have no set stop time on
+       * the output segment, we update the segment stop time to reflect
+       * the last sample. This will let oggmux set the last page's
+       * granpos to tell a decoder the dummy samples should be clipped.
+       */
+      segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (enc);
+      GST_DEBUG_OBJECT (enc, "existing output segment %" GST_SEGMENT_FORMAT,
+          segment);
+      if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
+        int input_samples = bsize / (enc->channels * 2);
+        GST_DEBUG_OBJECT (enc,
+            "No stop time and partial frame, updating segment");
+        duration =
+            gst_util_uint64_scale (enc->encoded_samples + input_samples,
+            GST_SECOND, enc->rate);
+        segment->stop = segment->start + duration;
+        GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
+            segment);
+        gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
+            gst_event_new_segment (segment));
+      }
+
       size = ((bsize / bytes) + 1) * bytes;
       data0 = data = g_malloc0 (size);
       memcpy (data, bdata, bsize);
@@ -602,6 +630,7 @@ gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
 
   ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
       outbuf, samples);
+  enc->encoded_samples += frame_size;
 
 done:
   g_free (data0);
@@ -611,16 +640,17 @@ done:
 /*
  * (really really) FIXME: move into core (dixit tpm)
  */
-/**
+/*
  * _gst_caps_set_buffer_array:
- * @caps: a #GstCaps
+ * @caps: (transfer full): a #GstCaps
  * @field: field in caps to set
  * @buf: header buffers
  *
  * Adds given buffers to an array of buffers set as the given @field
  * on the given @caps.  List of buffer arguments must be NULL-terminated.
  *
- * Returns: input caps with a streamheader field added, or NULL if some error
+ * Returns: (transfer full): input caps with a streamheader field added, or NULL
+ *     if some error occurred
  */
 static GstCaps *
 _gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
@@ -658,6 +688,7 @@ _gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
 
     buf = va_arg (va, GstBuffer *);
   }
+  va_end (va);
 
   gst_structure_set_value (structure, field, &array);
   g_value_unset (&array);