Port to gstreamer 1.0 API

Includes fixes:
 - use a callback, not a signal for appsink's new_buffer

3 files changed, 88 insertions, 46 deletions

diff --git a/configure.ac b/configure.ac
index de8a9ff..b65d62e 100644
--- a/configure.ac
+++ b/configure.ac
@@ -38,7 +38,7 @@ PKG_CHECK_MODULES(PROTOBUF, [libprotobuf-c],[], [
 ])
 PKG_CHECK_MODULES(GLIB, [glib-2.0 >= 2.28])
 PKG_CHECK_MODULES(GIO,  [gio-2.0])
-PKG_CHECK_MODULES(GSTREAMER, [gstreamer-0.10 gstreamer-app-0.10])
+PKG_CHECK_MODULES(GSTREAMER, [gstreamer-1.0 gstreamer-app-1.0 gstreamer-tag-1.0])
 PKG_CHECK_MODULES(CELT, [celt])
 
 GST_ELEMENTS="appsrc appsink celtdec celtenc capsfilter
@@ -46,8 +46,8 @@ GST_ELEMENTS="appsrc appsink celtdec celtenc capsfilter
 
 for element in $GST_ELEMENTS
 do
-	AM_GST_ELEMENT_CHECK([$element], [],
-			     AC_MSG_ERROR([gstreamer element $element not found]))
+	GST_ELEMENT_CHECK([$element], 1.0, [],
+			  AC_MSG_ERROR([gstreamer element $element not found]))
 done
 
 if test "x$GCC" = "xyes"; then
diff --git a/src/audio.c b/src/audio.c
index 146fbf3..b1d46b3 100644
--- a/src/audio.c
+++ b/src/audio.c
@@ -3,8 +3,15 @@
 #include "cmumble.h"
 #include <string.h>
 
+#include <gst/tag/tag.h>
+
 #define SAMPLERATE 48000
+#define FRAMESIZE 480 /* SAMPLERATE/100 */
 #define CHANNELS 1
+#define CELT_CAPS "audio/x-celt,channels=" G_STRINGIFY(CHANNELS) "," \
+	"rate=" G_STRINGIFY(SAMPLERATE) ",frame-size=" G_STRINGIFY(FRAMESIZE)
+#define AUDIO_CAPS "audio/x-raw,format=S16LE,channels=" \
+	G_STRINGIFY(CHANNELS) ",rate=" G_STRINGIFY(SAMPLERATE)
 
 void
 cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
@@ -32,14 +39,15 @@ cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
 	 *  - enqueue as now?
 	 */
 
+	gstbuf = gst_buffer_new_wrapped(g_memdup(data, size), size);
+
 	/* Asume packets are in order, since we're using tcp tunnel only atm.
 	 * FIXME: This assumption is probably wrong, since the packets may have
 	 *        been received out of order at the server? */
 
-	if (sequence < 0) {
-		time = 0;
-	} else if (user->last_sequence < 0 || sequence == 0 ||
+	if (user->last_sequence < 0 || sequence == 0 ||
 		   sequence < (user->last_sequence + 1)) {
+		GST_BUFFER_FLAG_SET(gstbuf, GST_BUFFER_FLAG_RESYNC);
 		time = now - base;
 		if (cm->verbose)
 			g_print("%s: set time to now\n", __func__);
@@ -59,15 +67,15 @@ cmumble_audio_push(struct cmumble *cm, struct cmumble_user *user,
 	}
 
 	if (time < (now - base)) {
+		GST_BUFFER_FLAG_SET(gstbuf, GST_BUFFER_FLAG_RESYNC);
 		time = now - base;
 		if (cm->verbose)
 			g_print("%s: time is in the past, setting to now\n",
 				__func__);
 	}
 
-	gstbuf = gst_app_buffer_new(g_memdup(data, size), size, g_free, NULL);
-
-	GST_BUFFER_TIMESTAMP(gstbuf) = time;
+	GST_BUFFER_DTS(gstbuf) = now - base;
+	GST_BUFFER_PTS(gstbuf) = time;
 	GST_BUFFER_DURATION(gstbuf) = gst_util_uint64_scale_int (1, GST_SECOND, 100);
 
 	user->last_time_end = time + GST_BUFFER_DURATION(gstbuf);
@@ -80,11 +88,11 @@ static GstFlowReturn
 pull_buffer(GstAppSink *sink, gpointer user_data)
 {
 	struct cmumble *cm = user_data;
+	GstSample *sample;
 	GstBuffer *buf;
 	uint8_t data[1024];
 	uint32_t write = 0, pos = 0;
 	mumble_udptunnel_t tunnel;
-	static int seq = 0;
 
 	/* FIXME: Make this more generic/disable pulling
 	 * the pipeline completely if not connected?
@@ -92,14 +100,20 @@ pull_buffer(GstAppSink *sink, gpointer user_data)
 	if (cm->con.conn == NULL)
 		return GST_FLOW_OK;
 
-	buf = gst_app_sink_pull_buffer(cm->audio.sink);
+	sample = gst_app_sink_pull_sample(cm->audio.sink);
+	if (sample == NULL)
+		return GST_FLOW_ERROR;
+
+	buf = gst_sample_get_buffer(sample);
 
-	if (++seq <= 2) {
-		gst_buffer_unref(buf);
+	if (GST_BUFFER_FLAG_IS_SET(buf, GST_BUFFER_FLAG_HEADER)) {
+		gst_sample_unref(sample);
 		return GST_FLOW_OK;
 	}
-	if (GST_BUFFER_SIZE(buf) > 127) {
-		g_printerr("GOT TOO BIG BUFFER\n");
+
+	if (gst_buffer_get_size(buf) > 127) {
+		g_printerr("error: unexpected buffer size\n");
+		gst_sample_unref(sample);
 		return GST_FLOW_ERROR;
 	}
 
@@ -108,11 +122,11 @@ pull_buffer(GstAppSink *sink, gpointer user_data)
 	encode_varint(&data[pos], &write, ++cm->sequence, sizeof(data)-pos);
 	pos += write;
 
-	data[pos++] = 0x00 /*: 0x80 */ | (GST_BUFFER_SIZE(buf) & 0x7F);
-	memcpy(&data[pos], GST_BUFFER_DATA(buf), GST_BUFFER_SIZE(buf));
-	pos += GST_BUFFER_SIZE(buf);
+	data[pos++] = 0x00 /*: 0x80 */ | (gst_buffer_get_size(buf) & 0x7F);
+	gst_buffer_extract(buf, 0, &data[pos], gst_buffer_get_size(buf));
+	pos += gst_buffer_get_size(buf);
 
-	gst_buffer_unref(buf);
+	gst_sample_unref(sample);
 
 	cmumble_init_udptunnel(&tunnel);
 	tunnel.packet.data = data;
@@ -135,7 +149,7 @@ idle(gpointer user_data)
 }
 
 static GstFlowReturn
-new_buffer(GstAppSink *sink, gpointer user_data)
+new_sample(GstAppSink *sink, gpointer user_data)
 {
 	struct cmumble *cm = user_data;
 
@@ -145,16 +159,18 @@ new_buffer(GstAppSink *sink, gpointer user_data)
 	return GST_FLOW_OK;
 }
 
+GstAppSinkCallbacks sink_cbs = {
+	.new_sample = new_sample
+};
+
 static int
 setup_recording_gst_pipeline(struct cmumble *cm)
 {
 	GstElement *pipeline, *cutter, *sink;
 	GError *error = NULL;
-	GstCaps *caps;
-
-	char *desc = "autoaudiosrc ! cutter name=cutter ! audioresample ! audioconvert ! "
-		"audio/x-raw-int,channels=1,depth=16,rate=48000,signed=TRUE,width=16 ! "
-		"celtenc ! appsink name=sink";
+	char *desc = "autoaudiosrc name=src ! cutter name=cutter ! "
+		     "audioresample ! audioconvert ! "AUDIO_CAPS" ! "
+		     "celtenc ! appsink name=sink caps="CELT_CAPS;
 
 	pipeline = gst_parse_launch(desc, &error);
 	if (error) {
@@ -170,17 +186,8 @@ setup_recording_gst_pipeline(struct cmumble *cm)
 	g_object_set(G_OBJECT(cutter),
 		     "threshold_dB", -45.0, "leaky", TRUE, NULL);
 
-	gst_app_sink_set_emit_signals(cm->audio.sink, TRUE);
+	gst_app_sink_set_callbacks(cm->audio.sink, &sink_cbs, cm, NULL);
 	gst_app_sink_set_drop(cm->audio.sink, FALSE);;
-	g_signal_connect(sink, "new-buffer", G_CALLBACK(new_buffer), cm);
-
-	caps = gst_caps_new_simple("audio/x-celt",
-				   "rate", G_TYPE_INT, SAMPLERATE,
-				   "channels", G_TYPE_INT, 1,
-				   "frame-size", G_TYPE_INT, SAMPLERATE/100,
-				   NULL);
-	gst_app_sink_set_caps(cm->audio.sink, caps);
-	gst_caps_unref(caps);
 
 	gst_element_set_state(pipeline, GST_STATE_PLAYING);
 
@@ -190,9 +197,9 @@ setup_recording_gst_pipeline(struct cmumble *cm)
 }
 
 static void
-set_pulse_states(gpointer data, gpointer user_data)
+set_pulse_states(const GValue *item, gpointer user_data)
 {
-	GstElement *elm = data;
+	GstElement *elm = g_value_get_object(item);
 	struct cmumble_user *user = user_data;
 	GstStructure *props;
 	gchar *name;
@@ -224,13 +231,54 @@ out:
 	g_object_unref(G_OBJECT(elm));
 }
 
+static void
+add_celt_streamheader(struct cmumble *cm, GstAppSrc *src)
+{
+	GValue streamheader = { 0, }, val = { 0, };
+	GstTagList *tags;
+	GstBuffer *buf[2];
+	GstStructure *s;
+	GstCaps *caps;
+	gint i;
+
+	buf[0] = gst_buffer_new_allocate(NULL, sizeof(CELTHeader), NULL);
+	gst_buffer_fill(buf[0], 0, cm->audio.celt_header_packet,
+			sizeof(CELTHeader));
+
+	tags = gst_tag_list_new_empty();
+	buf[1] = gst_tag_list_to_vorbiscomment_buffer(tags, NULL, 0, "mumble");
+	gst_tag_list_unref(tags);
+
+	g_value_init(&streamheader, GST_TYPE_ARRAY);
+	for (i = 0; i < G_N_ELEMENTS(buf); ++i) {
+		GST_BUFFER_FLAG_SET(buf[i], GST_BUFFER_FLAG_HEADER);
+		GST_BUFFER_OFFSET(buf[i]) = 0;
+		GST_BUFFER_OFFSET_END(buf[i]) = 0;
+		g_value_init(&val, GST_TYPE_BUFFER);
+		gst_value_take_buffer(&val, buf[i]);
+		gst_value_array_append_value(&streamheader, &val);
+		g_value_unset(&val);
+	}
+
+	caps = gst_app_src_get_caps(src);
+	caps = gst_caps_make_writable(caps);
+	s = gst_caps_get_structure(caps, 0);
+	gst_structure_set_value(s, "streamheader", &streamheader);
+	g_value_unset(&streamheader);
+
+	gst_app_src_set_caps(src, caps);
+	gst_caps_unref(caps);
+}
+
 int
 cmumble_audio_create_playback_pipeline(struct cmumble *cm,
 				       struct cmumble_user *user)
 {
 	GstElement *pipeline, *sink_bin;
 	GError *error = NULL;
-	char *desc = "appsrc name=src ! celtdec ! audioconvert ! autoaudiosink name=sink";
+	char *desc = "appsrc name=src format=GST_FORMAT_TIME caps="CELT_CAPS" "
+		     "! celtdec name=dec "
+		     "! audioresample ! audioconvert ! autoaudiosink name=sink";
 	GstIterator *it;
 
 	pipeline = gst_parse_launch(desc, &error);
@@ -241,7 +289,7 @@ cmumble_audio_create_playback_pipeline(struct cmumble *cm,
 
 	user->pipeline = pipeline;
 	user->src = GST_APP_SRC(gst_bin_get_by_name(GST_BIN(pipeline), "src"));
-	gst_base_src_set_format(GST_BASE_SRC(user->src), GST_FORMAT_TIME);
+	add_celt_streamheader(cm, user->src);
 
 	gst_element_set_state(pipeline, GST_STATE_PLAYING);
 
@@ -252,12 +300,7 @@ cmumble_audio_create_playback_pipeline(struct cmumble *cm,
 	gst_iterator_foreach(it, set_pulse_states, user);
 	gst_iterator_free(it);
 
-	user->last_sequence = -2;
-	/* Setup Celt Decoder */
-	cmumble_audio_push(cm, user,
-			   cm->audio.celt_header_packet, sizeof(CELTHeader), -2);
-	/* fake vorbiscomment buffer */
-	cmumble_audio_push(cm, user, NULL, 0, -2);
+	user->last_sequence = -1;
 
 	return 0;
 }
diff --git a/src/audio.h b/src/audio.h
index 6f91012..8e47ca9 100644
--- a/src/audio.h
+++ b/src/audio.h
@@ -6,7 +6,6 @@
 #include <gst/gst.h>
 #include <gst/app/gstappsrc.h>
 #include <gst/app/gstappsink.h>
-#include <gst/app/gstappbuffer.h>
 
 #include <celt/celt.h>
 #include <celt/celt_header.h>