Merge branch 'fdo' into lv2

6 files changed, 53 insertions, 16 deletions

diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c
index 1aa1baf8..60e41381 100644
--- a/ext/jack/gstjackaudioclient.c
+++ b/ext/jack/gstjackaudioclient.c
@@ -42,6 +42,7 @@ typedef struct
 {
   gint refcount;
   GMutex *lock;
+  GCond *flush_cond;
 
   /* id/server pair and the connection */
   gchar *id;
@@ -61,6 +62,7 @@ struct _GstJackAudioClient
 
   GstJackClientType type;
   gboolean active;
+  gboolean deactivate;
 
   void (*shutdown) (void *arg);
   JackProcessCallback process;
@@ -92,15 +94,25 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
     GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
 
     /* only call active clients */
-    if (client->active && client->process)
+    if ((client->active || client->deactivate) && client->process) {
       res = client->process (nframes, client->user_data);
+      if (client->deactivate) {
+        client->deactivate = FALSE;
+        g_cond_signal (conn->flush_cond);
+      }
+    }
   }
   for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
     GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
 
     /* only call active clients */
-    if (client->active && client->process)
+    if ((client->active || client->deactivate) && client->process) {
       res = client->process (nframes, client->user_data);
+      if (client->deactivate) {
+        client->deactivate = FALSE;
+        g_cond_signal (conn->flush_cond);
+      }
+    }
   }
   g_mutex_unlock (conn->lock);
 
@@ -203,6 +215,7 @@ gst_jack_audio_make_connection (const gchar * id, const gchar * server,
   conn = g_new (GstJackAudioConnection, 1);
   conn->refcount = 1;
   conn->lock = g_mutex_new ();
+  conn->flush_cond = g_cond_new ();
   conn->id = g_strdup (id);
   conn->server = g_strdup (server);
   conn->client = jclient;
@@ -325,6 +338,7 @@ gst_jack_audio_unref_connection (GstJackAudioConnection * conn)
 
     /* free resources */
     g_mutex_free (conn->lock);
+    g_cond_free (conn->flush_cond);
     g_free (conn->id);
     g_free (conn->server);
     g_free (conn);
@@ -409,9 +423,11 @@ gst_jack_audio_client_new (const gchar * id, const gchar * server,
   if (conn == NULL)
     goto no_connection;
 
+  GST_INFO ("new client %s", id);
+
   /* make new client using the connection */
   client = g_new (GstJackAudioClient, 1);
-  client->active = FALSE;
+  client->active = client->deactivate = FALSE;
   client->conn = conn;
   client->type = type;
   client->shutdown = shutdown;
@@ -446,6 +462,8 @@ gst_jack_audio_client_free (GstJackAudioClient * client)
 
   g_return_if_fail (client != NULL);
 
+  GST_INFO ("free client");
+
   conn = client->conn;
 
   /* remove from connection first so that it's not scheduled anymore after this
@@ -492,6 +510,14 @@ gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active)
 
   /* make sure that we are not dispatching the client */
   g_mutex_lock (client->conn->lock);
+  if (client->active && !active) {
+    /* we need to process once more to flush the port */
+    client->deactivate = TRUE;
+
+    /* need to wait for process_cb run once more */
+    while (client->deactivate)
+      g_cond_wait (client->conn->flush_cond, client->conn->lock);
+  }
   client->active = active;
   g_mutex_unlock (client->conn->lock);
 
diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c
index 01383cbb..58a699d0 100644
--- a/ext/jack/gstjackaudiosink.c
+++ b/ext/jack/gstjackaudiosink.c
@@ -213,8 +213,8 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
     if (nframes * sizeof (sample_t) != flen)
       goto wrong_size;
 
-    GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, readptr,
-        flen, channels);
+    GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
+        nframes, readptr, flen, channels);
     data = (sample_t *) readptr;
 
     /* the samples in the ringbuffer have the channels interleaved, we need to
@@ -231,6 +231,7 @@ jack_process_cb (jack_nframes_t nframes, void *arg)
     /* we wrote one segment */
     gst_ring_buffer_advance (buf, 1);
   } else {
+    GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
     /* We are not allowed to read from the ringbuffer, write silence to all
      * jack output buffers */
     for (i = 0; i < channels; i++) {
@@ -607,7 +608,7 @@ gst_jack_ring_buffer_delay (GstRingBuffer * buf)
       res = latency;
   }
 
-  GST_DEBUG_OBJECT (sink, "delay %u", res);
+  GST_LOG_OBJECT (sink, "delay %u", res);
 
   return res;
 }
diff --git a/ext/resindvd/gstmpegdemux.c b/ext/resindvd/gstmpegdemux.c
index 5ab26107..d9aefe53 100644
--- a/ext/resindvd/gstmpegdemux.c
+++ b/ext/resindvd/gstmpegdemux.c
@@ -356,7 +356,7 @@ gst_flups_demux_get_stream (GstFluPSDemux * demux, gint id, gint type)
 {
   GstFluPSStream *stream = demux->streams[id];
 
-  if (stream == NULL) {
+  if (stream == NULL && !demux->disable_stream_creation) {
     if (!(stream = gst_flups_demux_create_stream (demux, id, type)))
       goto unknown_stream;
 
@@ -612,6 +612,8 @@ gst_flups_demux_handle_dvd_event (GstFluPSDemux * demux, GstEvent * event)
 
     GST_DEBUG_OBJECT (demux, "Handling language codes event");
 
+    demux->disable_stream_creation = FALSE;
+
     /* Create a video pad to ensure it exists before emitting no more pads */
     temp = gst_flups_demux_get_stream (demux, 0xe0, ST_VIDEO_MPEG2);
     /* Send a video format event downstream */
@@ -719,6 +721,8 @@ gst_flups_demux_handle_dvd_event (GstFluPSDemux * demux, GstEvent * event)
           ST_PS_DVD_SUBPICTURE);
     }
 
+    demux->disable_stream_creation = TRUE;
+
     GST_DEBUG_OBJECT (demux, "Created all pads from Language Codes event, "
         "signalling no-more-pads");
 
@@ -767,32 +771,33 @@ gst_flups_demux_handle_dvd_event (GstFluPSDemux * demux, GstEvent * event)
           stream_id += 0x80;
           temp = demux->streams[stream_id];
           break;
-#if 0                           /* FIXME: Ignore non AC-3 requests until the decoder bin can handle them */
         case 0x2:
         case 0x3:
           /* MPEG audio without and with extension stream are 
            * treated the same */
-          stream_id = 0xC0 + i;
+          stream_id += 0xC0;
           temp = demux->streams[stream_id];
           break;
         case 0x4:
           /* LPCM */
-          stream_id = 0xA0 + i;
+          stream_id += 0xA0;
           temp = demux->streams[stream_id];
           break;
         case 0x6:
           /* DTS */
-          stream_id = 0x88 + i;
+          stream_id += 0x88;
           temp = demux->streams[stream_id];
           break;
         case 0x7:
           /* FIXME: What range is SDDS? */
+          temp = NULL;
           break;
-#endif
         default:
           temp = NULL;
           break;
       }
+      GST_INFO_OBJECT (demux, "Have DVD audio stream select event: "
+          "stream 0x%02x", stream_id);
       if (temp != NULL && temp->pad != NULL) {
         /* Send event to the selector to activate the desired pad */
         GstStructure *s = gst_structure_new ("application/x-gst-dvd",
diff --git a/ext/resindvd/gstmpegdemux.h b/ext/resindvd/gstmpegdemux.h
index 683b7cde..f6291500 100644
--- a/ext/resindvd/gstmpegdemux.h
+++ b/ext/resindvd/gstmpegdemux.h
@@ -106,6 +106,7 @@ struct _GstFluPSDemux {
 
   /* Indicates an MPEG-2 stream */
   gboolean is_mpeg2_pack;
+  gboolean disable_stream_creation;
 
   /* Language codes event is stored when a dvd-lang-codes
    * custom event arrives from upstream */
diff --git a/ext/resindvd/resindvdsrc.c b/ext/resindvd/resindvdsrc.c
index f24e45aa..122f2b67 100644
--- a/ext/resindvd/resindvdsrc.c
+++ b/ext/resindvd/resindvdsrc.c
@@ -1882,7 +1882,7 @@ rsn_dvdsrc_prepare_streamsinfo_event (resinDvdSrc * src)
     GST_DEBUG_OBJECT (src, "mapped logical audio %d to MPEG substream %d",
         i, phys_id);
 
-#if 1
+#if 0
     /* FIXME: Only output A52 streams for now, until the decoder switching
      * is ready */
     if (a->audio_format != 0) {
@@ -1891,7 +1891,8 @@ rsn_dvdsrc_prepare_streamsinfo_event (resinDvdSrc * src)
       continue;
     }
 #endif
-    have_audio = TRUE;
+    if (a->audio_format == 0)
+      have_audio = TRUE;
 
     GST_DEBUG_OBJECT (src, "Audio stream %d is format %d, substream %d", i,
         (int) a->audio_format, phys_id);
@@ -1917,7 +1918,7 @@ rsn_dvdsrc_prepare_streamsinfo_event (resinDvdSrc * src)
   }
 
   if (have_audio == FALSE) {
-    /* Always create at least one audio stream */
+    /* Always create at least one audio stream of the required type */
     gst_structure_set (s, "audio-0-format", G_TYPE_INT, (int) 0,
         "audio-0-stream", G_TYPE_INT, (int) 0, NULL);
   }
diff --git a/ext/resindvd/rsnaudiomunge.c b/ext/resindvd/rsnaudiomunge.c
index 50d33be7..fda2a96b 100644
--- a/ext/resindvd/rsnaudiomunge.c
+++ b/ext/resindvd/rsnaudiomunge.c
@@ -200,10 +200,13 @@ rsn_audiomunge_make_audio (RsnAudioMunge * munge,
   guint buf_size;
 
   /* Just generate a 48khz stereo buffer for now */
+  /* FIXME: Adapt to the allowed formats, according to the currently
+   * plugged decoder, or at least add a source pad that accepts the
+   * caps we're outputting if the upstream decoder does not */
 #if 0
   caps =
       gst_caps_from_string
-      ("audio/x-raw-int,rate=48000,channels=2,width=16,depth=16,signed=(boolean)true,endianness=1234");
+      ("audio/x-raw-int,rate=48000,channels=2,width=16,depth=16,signed=(boolean)true,endianness=4321");
   buf_size = 4 * (48000 * fill_time / GST_SECOND);
 #else
   caps = gst_caps_from_string ("audio/x-raw-float, endianness=(int)1234,"