Merge branch 'master' into 0.11

Conflicts:
	ext/celt/gstceltdec.c
	ext/opus/gstopusdec.c
	ext/opus/gstopusdec.h
	ext/opus/gstopusenc.c
	ext/opus/gstopusenc.h
	ext/opus/gstopusparse.c

ext/opus/gstopusdec.c

@@ -67,14 +67,14 @@ GST_STATIC_PAD_TEMPLATE ("sink",
 
 G_DEFINE_TYPE (GstOpusDec, gst_opus_dec, GST_TYPE_AUDIO_DECODER);
 
+static GstFlowReturn gst_opus_dec_parse_header (GstOpusDec * dec,
+    GstBuffer * buf);
 static gboolean gst_opus_dec_start (GstAudioDecoder * dec);
 static gboolean gst_opus_dec_stop (GstAudioDecoder * dec);
 static GstFlowReturn gst_opus_dec_handle_frame (GstAudioDecoder * dec,
     GstBuffer * buffer);
 static gboolean gst_opus_dec_set_format (GstAudioDecoder * bdec,
     GstCaps * caps);
-static GstFlowReturn opus_dec_chain_parse_data (GstOpusDec * dec,
-    GstBuffer * buf, GstClockTime timestamp, GstClockTime duration);
 
 static void
 gst_opus_dec_class_init (GstOpusDecClass * klass)
@@ -112,8 +112,6 @@ gst_opus_dec_reset (GstOpusDec * dec)
     dec->state = NULL;
   }
 
-  dec->next_ts = 0;
-
   gst_buffer_replace (&dec->streamheader, NULL);
   gst_buffer_replace (&dec->vorbiscomment, NULL);
 }
@@ -167,17 +165,19 @@ gst_opus_dec_setup_from_peer_caps (GstOpusDec * dec)
 {
   GstPad *srcpad, *peer;
   GstStructure *s;
-  GstCaps *caps, *template_caps, *peer_caps;
+  GstCaps *caps;
+  const GstCaps *template_caps;
+  const GstCaps *peer_caps;
 
   srcpad = GST_AUDIO_DECODER_SRC_PAD (dec);
   peer = gst_pad_get_peer (srcpad);
 
   if (peer) {
     template_caps = gst_pad_get_pad_template_caps (srcpad);
-    peer_caps = gst_pad_get_caps (peer, NULL);
+    peer_caps = gst_pad_get_caps (peer);
     GST_DEBUG_OBJECT (dec, "Peer caps: %" GST_PTR_FORMAT, peer_caps);
     caps = gst_caps_intersect (template_caps, peer_caps);
-    gst_caps_fixate (caps);
+    gst_pad_fixate_caps (peer, caps);
     GST_DEBUG_OBJECT (dec, "Fixated caps: %" GST_PTR_FORMAT, caps);
 
     s = gst_caps_get_structure (caps, 0);
@@ -203,8 +203,7 @@ gst_opus_dec_setup_from_peer_caps (GstOpusDec * dec)
 }
 
 static GstFlowReturn
-opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
+opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf)
-    GstClockTime timestamp, GstClockTime duration)
 {
   GstFlowReturn res = GST_FLOW_OK;
   gsize size, out_size;
@@ -218,6 +217,8 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
   if (dec->state == NULL) {
     gst_opus_dec_setup_from_peer_caps (dec);
 
+    GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
+        dec->n_channels, dec->sample_rate);
     dec->state = opus_decoder_create (dec->sample_rate, dec->n_channels, &err);
     if (!dec->state || err != OPUS_OK)
       goto creation_failed;
@@ -227,8 +228,6 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
     data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
 
     GST_DEBUG_OBJECT (dec, "received buffer of size %u", size);
-
-    /* copy timestamp */
   } else {
     /* concealment data, pass NULL as the bits parameters */
     GST_DEBUG_OBJECT (dec, "creating concealment data");
@@ -261,20 +260,6 @@ opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buf,
   }
   GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
 
-  if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
-    GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
-  } else {
-    GST_BUFFER_TIMESTAMP (outbuf) = dec->next_ts;
-  }
-
-  GST_BUFFER_DURATION (outbuf) =
-      gst_util_uint64_scale (n, GST_SECOND, dec->sample_rate);
-  dec->next_ts = GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
-
-  GST_LOG_OBJECT (dec, "pushing buffer with ts=%" GST_TIME_FORMAT ", dur=%"
-      GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
-      GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
-
   res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
 
   if (res != GST_FLOW_OK)
@@ -395,8 +380,7 @@ gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
       gst_audio_decoder_finish_frame (adec, NULL, 1);
       res = GST_FLOW_OK;
     } else {
-      res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
+      res = opus_dec_chain_parse_data (dec, buf);
-          GST_BUFFER_DURATION (buf));
     }
   } else {
     /* Otherwise fall back to packet counting and assume that the
@@ -408,8 +392,7 @@ gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
           res = gst_opus_dec_parse_header (dec, buf);
           gst_audio_decoder_finish_frame (adec, NULL, 1);
         } else {
-          res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
+          res = opus_dec_chain_parse_data (dec, buf);
-              GST_BUFFER_DURATION (buf));
         }
         break;
       case 1:
@@ -418,14 +401,12 @@ gst_opus_dec_handle_frame (GstAudioDecoder * adec, GstBuffer * buf)
           res = gst_opus_dec_parse_comments (dec, buf);
           gst_audio_decoder_finish_frame (adec, NULL, 1);
         } else {
-          res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
+          res = opus_dec_chain_parse_data (dec, buf);
-              GST_BUFFER_DURATION (buf));
         }
         break;
       default:
       {
-        res = opus_dec_chain_parse_data (dec, buf, GST_BUFFER_TIMESTAMP (buf),
+        res = opus_dec_chain_parse_data (dec, buf);
-            GST_BUFFER_DURATION (buf));
         break;
       }
     }

ext/opus/gstopusdec.h

@@ -47,7 +47,6 @@ struct _GstOpusDec {
   OpusDecoder          *state;
 
   guint64               packetno;
-  GstClockTime          next_ts;
 
   GstBuffer            *streamheader;
   GstBuffer            *vorbiscomment;

ext/opus/gstopusenc.c

@@ -94,9 +94,7 @@ static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
 static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-opus, "
+    GST_STATIC_CAPS ("audio/x-opus")
-        "rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
-        "channels = (int) [ 1, 2 ], " "frame-size = (int) [ 2, 60 ]")
     );
 
 #define DEFAULT_AUDIO           TRUE
@@ -144,6 +142,9 @@ static GstFlowReturn gst_opus_enc_handle_frame (GstAudioEncoder * benc,
     GstBuffer * buf);
 static GstFlowReturn gst_opus_enc_pre_push (GstAudioEncoder * benc,
     GstBuffer ** buffer);
+static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc);
+
+static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buffer);
 
 static GstFlowReturn gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf);
 static gint64 gst_opus_enc_get_latency (GstOpusEnc * enc);
@@ -157,13 +158,16 @@ static void
 gst_opus_enc_class_init (GstOpusEncClass * klass)
 {
   GObjectClass *gobject_class;
-  GstElementClass *element_class;
+  GstElementClass *gstelement_class;
   GstAudioEncoderClass *base_class;
 
   gobject_class = (GObjectClass *) klass;
-  element_class = (GstElementClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
   base_class = (GstAudioEncoderClass *) klass;
 
+  gobject_class->set_property = gst_opus_enc_set_property;
+  gobject_class->get_property = gst_opus_enc_get_property;
+
   gst_element_class_add_pad_template (element_class,
       gst_static_pad_template_get (&src_factory));
   gst_element_class_add_pad_template (element_class,
@@ -180,9 +184,6 @@ gst_opus_enc_class_init (GstOpusEncClass * klass)
   base_class->pre_push = GST_DEBUG_FUNCPTR (gst_opus_enc_pre_push);
   base_class->event = GST_DEBUG_FUNCPTR (gst_opus_enc_sink_event);
 
-  gobject_class->set_property = gst_opus_enc_set_property;
-  gobject_class->get_property = gst_opus_enc_get_property;
-
   g_object_class_install_property (gobject_class, PROP_AUDIO,
       g_param_spec_boolean ("audio", "Audio or voice",
           "Audio or voice", DEFAULT_AUDIO,
@@ -205,7 +206,7 @@ gst_opus_enc_class_init (GstOpusEncClass * klass)
           "Constant bit rate", DEFAULT_CBR,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
   g_object_class_install_property (gobject_class, PROP_CONSTRAINED_VBR,
-      g_param_spec_boolean ("constrained-cbr", "Constrained VBR",
+      g_param_spec_boolean ("constrained-vbr", "Constrained VBR",
           "Constrained VBR", DEFAULT_CONSTRAINED_VBR,
           G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
   g_object_class_install_property (gobject_class, PROP_COMPLEXITY,
@@ -237,12 +238,11 @@ gst_opus_enc_finalize (GObject * object)
 
   enc = GST_OPUS_ENC (object);
 
-  GST_DEBUG_OBJECT (enc, "finalize");
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
 static void
-gst_opus_enc_init (GstOpusEnc * enc)
+gst_opus_enc_init (GstOpusEnc * enc, GstOpusEncClass * klass)
 {
   GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
 
@@ -273,7 +273,7 @@ gst_opus_enc_start (GstAudioEncoder * benc)
   GstOpusEnc *enc = GST_OPUS_ENC (benc);
 
   GST_DEBUG_OBJECT (enc, "start");
-  enc->tags = gst_tag_list_new_empty ();
+  enc->tags = gst_tag_list_new ();
   enc->header_sent = FALSE;
   return TRUE;
 }
@@ -297,6 +297,15 @@ gst_opus_enc_stop (GstAudioEncoder * benc)
   return TRUE;
 }
 
+static gint64
+gst_opus_enc_get_latency (GstOpusEnc * enc)
+{
+  gint64 latency = gst_util_uint64_scale (enc->frame_samples, GST_SECOND,
+      enc->sample_rate);
+  GST_DEBUG_OBJECT (enc, "Latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
+  return latency;
+}
+
 static gint
 gst_opus_enc_get_frame_samples (GstOpusEnc * enc)
 {
@@ -345,7 +354,6 @@ gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
     opus_encoder_destroy (enc->state);
     enc->state = NULL;
   }
-
   if (!gst_opus_enc_setup (enc))
     return FALSE;
 
@@ -354,7 +362,6 @@ gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
   /* feedback to base class */
   gst_audio_encoder_set_latency (benc,
       gst_opus_enc_get_latency (enc), gst_opus_enc_get_latency (enc));
-
   gst_audio_encoder_set_frame_samples_min (benc,
       enc->frame_samples * enc->n_channels * 2);
   gst_audio_encoder_set_frame_samples_max (benc,
@@ -364,15 +371,6 @@ gst_opus_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
   return TRUE;
 }
 
-static gint64
-gst_opus_enc_get_latency (GstOpusEnc * enc)
-{
-  gint64 latency = gst_util_uint64_scale (enc->frame_samples, GST_SECOND,
-      enc->sample_rate);
-  GST_DEBUG_OBJECT (enc, "Latency: %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
-  return latency;
-}
-
 static GstBuffer *
 gst_opus_enc_create_id_buffer (GstOpusEnc * enc)
 {
@@ -495,8 +493,8 @@ gst_opus_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
 static GstFlowReturn
 gst_opus_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer)
 {
-  GstOpusEnc *enc;
   GstFlowReturn ret = GST_FLOW_OK;
+  GstOpusEnc *enc;
 
   enc = GST_OPUS_ENC (benc);
 
@@ -522,6 +520,39 @@ gst_opus_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer)
   return ret;
 }
 
+static GstFlowReturn
+gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
+{
+  guint8 *bdata, *data, *mdata = NULL;
+  gsize bsize, size;
+  gsize bytes = enc->frame_samples * enc->n_channels * 2;
+  gsize bytes_per_packet =
+      (enc->bitrate * enc->frame_samples / enc->sample_rate + 4) / 8;
+  gint ret = GST_FLOW_OK;
+
+  if (G_LIKELY (buf)) {
+    bdata = GST_BUFFER_DATA (buf);
+    bsize = GST_BUFFER_SIZE (buf);
+    if (G_UNLIKELY (bsize % bytes)) {
+      GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
+
+      size = ((bsize / bytes) + 1) * bytes;
+      mdata = g_malloc0 (size);
+      memcpy (mdata, bdata, bsize);
+      bdata = NULL;
+      data = mdata;
+    } else {
+      data = bdata;
+      size = bsize;
+    }
+  } else {
+    GST_DEBUG_OBJECT (enc, "nothing to drain");
+    goto done;
+  }
+
+  return ret;
+}
+
 static GstFlowReturn
 gst_opus_enc_encode (GstOpusEnc * enc, GstBuffer * buf)
 {
@@ -669,7 +700,6 @@ gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
   GstFlowReturn ret = GST_FLOW_OK;
 
   enc = GST_OPUS_ENC (benc);
-
   GST_DEBUG_OBJECT (enc, "handle_frame");
 
   if (!enc->header_sent) {
@@ -684,17 +714,13 @@ gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
     buf2 = gst_opus_enc_create_metadata_buffer (enc);
 
     /* mark and put on caps */
-    caps =
+    caps = gst_caps_from_string ("audio/x-opus");
-        gst_caps_new_simple ("audio/x-opus", "rate", G_TYPE_INT,
-        enc->sample_rate, "channels", G_TYPE_INT, enc->n_channels, "frame-size",
-        G_TYPE_INT, enc->frame_size, NULL);
     caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL);
 
     /* negotiate with these caps */
     GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
 
     gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
-    gst_caps_unref (caps);
 
     /* push out buffers */
     /* store buffers for later pre_push sending */
@@ -703,12 +729,11 @@ gst_opus_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
     GST_DEBUG_OBJECT (enc, "storing header buffers");
     enc->headers = g_slist_prepend (enc->headers, buf2);
     enc->headers = g_slist_prepend (enc->headers, buf1);
-
     enc->header_sent = TRUE;
   }
 
   GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
-      buf ? gst_buffer_get_size (buf) : 0);
+      buf ? GST_BUFFER_SIZE (buf) : 0);
 
   ret = gst_opus_enc_encode (enc, buf);
 

ext/opus/gstopusenc.h

@@ -48,11 +48,7 @@ typedef struct _GstOpusEnc GstOpusEnc;
 typedef struct _GstOpusEncClass GstOpusEncClass;
 
 struct _GstOpusEnc {
-  GstAudioEncoder        element;
+  GstAudioEncoder       element;
-
-  /* pads */
-  GstPad                *sinkpad;
-  GstPad                *srcpad;
 
   OpusEncoder          *state;
 
@@ -74,7 +70,8 @@ struct _GstOpusEnc {
 
   gboolean              setup;
   gboolean              header_sent;
-  GSList               *headers;
+
+  GSList                *headers;
 
   GstTagList            *tags;
 };