Add GStreamer 1.0 supportgst-1.0

This commit adds GStreamer 1.0 support. As the changes compared to
GStreamer 0.10 are minor (only 2 functions to modify), I've kept support
for both versions. GStreamer 1.0 support can be enabled using
--with-audio=gstreamer1

3 files changed, 58 insertions, 6 deletions

diff --git a/configure.ac b/configure.ac
index f47ee20..91633e4 100644
--- a/configure.ac
+++ b/configure.ac
@@ -288,7 +288,7 @@ AS_IF([test "x$have_phodav" = "xyes"],
        AC_DEFINE(USE_PHODAV, [1], [Define if supporting phodav]))
 
 AC_ARG_WITH([audio],
-  AS_HELP_STRING([--with-audio=@<:@gstreamer/pulse/auto/no@:>@], [Select audio backend @<:@default=auto@:>@]),
+  AS_HELP_STRING([--with-audio=@<:@gstreamer/gstreamer1/pulse/auto/no@:>@], [Select audio backend @<:@default=auto@:>@]),
   [],
   [with_audio="auto"])
 
@@ -297,7 +297,7 @@ AS_IF([test "x$with_audio" = "xauto"], [
 ])
 
 case "$with_audio" in
-  gstreamer|pulse|no*)
+  gstreamer|gstreamer1|pulse|no*)
     ;;
   *) AC_MSG_ERROR(Unsupported audio backend)
 esac
@@ -326,7 +326,18 @@ AS_IF([test "x$have_gst" = "xyes"],
              [AC_MSG_ERROR([GStreamer requested but not found])
       ])
 ])
-AM_CONDITIONAL([WITH_GSTAUDIO], [test "x$have_gst" = "xyes"])
+
+AS_IF([test "x$with_audio" = "xgstreamer1"],
+      [PKG_CHECK_MODULES(GST, gstreamer-1.0 gstreamer-base-1.0 gstreamer-app-1.0 gstreamer-audio-1.0, [have_gst1=yes], [have_gst1=no])],
+      [have_gst1=no])
+
+AS_IF([test "x$have_gst1" = "xyes"],
+      [AC_DEFINE([WITH_GST1AUDIO], 1, [Have GStreamer 1.0?])],
+      [AS_IF([test "x$with_audio" = "xgstreamer1"],
+             [AC_MSG_ERROR([GStreamer 1.0 requested but not found])
+      ])
+])
+AM_CONDITIONAL([WITH_GSTAUDIO], [test "x$have_gst" = "xyes" -o "x$have_gst1" = "xyes"])
 AC_SUBST(GST_CFLAGS)
 AC_SUBST(GST_LIBS)
 
diff --git a/gtk/spice-audio.c b/gtk/spice-audio.c
index dbd3a8b..34c1b69 100644
--- a/gtk/spice-audio.c
+++ b/gtk/spice-audio.c
@@ -47,7 +47,7 @@
 #ifdef WITH_PULSE
 #include "spice-pulse.h"
 #endif
-#ifdef WITH_GSTAUDIO
+#if defined(WITH_GSTAUDIO) || defined(WITH_GST1AUDIO)
 #include "spice-gstaudio.h"
 #endif
 
@@ -219,7 +219,7 @@ SpiceAudio *spice_audio_new(SpiceSession *session, GMainContext *context,
 #ifdef WITH_PULSE
     self = SPICE_AUDIO(spice_pulse_new(session, context, name));
 #endif
-#ifdef WITH_GSTAUDIO
+#if defined(WITH_GSTAUDIO) || defined(WITH_GST1AUDIO)
     self = SPICE_AUDIO(spice_gstaudio_new(session, context, name));
 #endif
     if (!self)
diff --git a/gtk/spice-gstaudio.c b/gtk/spice-gstaudio.c
index faa0c74..fb76dfb 100644
--- a/gtk/spice-gstaudio.c
+++ b/gtk/spice-gstaudio.c
@@ -21,9 +21,13 @@
 
 #include <gst/gst.h>
 #include <gst/app/gstappsrc.h>
-#include <gst/app/gstappbuffer.h>
 #include <gst/app/gstappsink.h>
+#ifdef WITH_GST1AUDIO
+#include <gst/audio/streamvolume.h>
+#else
+#include <gst/app/gstappbuffer.h>
 #include <gst/interfaces/streamvolume.h>
+#endif
 
 #include "spice-gstaudio.h"
 #include "spice-common.h"
@@ -155,6 +159,38 @@ static gboolean record_bus_cb(GstBus *bus, GstMessage *msg, gpointer data)
     g_return_val_if_fail(p != NULL, FALSE);
 
     switch (GST_MESSAGE_TYPE(msg)) {
+#ifdef WITH_GST1AUDIO
+    case GST_MESSAGE_APPLICATION: {
+        GstSample *s;
+        GstBuffer *buffer;
+        GstMapInfo mapping;
+
+        s = gst_app_sink_pull_sample(GST_APP_SINK(p->record.sink));
+        if (!s) {
+            if (!gst_app_sink_is_eos(GST_APP_SINK(p->record.sink)))
+                g_warning("eos not reached, but can't pull new sample");
+            return TRUE;
+        }
+
+        buffer = gst_sample_get_buffer(s);
+        if (!buffer) {
+            if (!gst_app_sink_is_eos(GST_APP_SINK(p->record.sink)))
+                g_warning("eos not reached, but can't pull new buffer");
+            return TRUE;
+        }
+        if (!gst_buffer_map(buffer, &mapping, GST_MAP_READ)) {
+            return TRUE;
+        }
+
+        spice_record_send_data(SPICE_RECORD_CHANNEL(p->rchannel),
+                               /* FIXME: server side doesn't care about ts?
+                                  what is the unit? ms apparently */
+                               mapping.data, mapping.size, 0);
+        gst_buffer_unmap(buffer, &mapping);
+        gst_sample_unref(s);
+        break;
+    }
+#else
     case GST_MESSAGE_APPLICATION: {
         GstBuffer *b;
 
@@ -171,6 +207,7 @@ static gboolean record_bus_cb(GstBus *bus, GstMessage *msg, gpointer data)
                                GST_BUFFER_DATA(b), GST_BUFFER_SIZE(b), 0);
         break;
     }
+#endif
     default:
         break;
     }
@@ -356,7 +393,11 @@ static void playback_data(SpicePlaybackChannel *channel,
     g_return_if_fail(p != NULL);
 
     audio = g_memdup(audio, size); /* TODO: try to avoid memory copy */
+#ifdef WITH_GST1AUDIO
+    buf = gst_buffer_new_wrapped(audio, size);
+#else
     buf = gst_app_buffer_new(audio, size, g_free, audio);
+#endif
     gst_app_src_push_buffer(GST_APP_SRC(p->playback.src), buf);
 }