Move audio-related stuff into audio.c

5 files changed, 254 insertions, 205 deletions

diff --git a/src/Makefile.am b/src/Makefile.am
index 7004c5d..3dec946 100644
--- a/src/Makefile.am
+++ b/src/Makefile.am
@@ -1,9 +1,9 @@
 bin_PROGRAMS = cmumble
 
-noinst_HEADERS = cmumble.h message.h varint.h io.h connection.h
+noinst_HEADERS = cmumble.h message.h varint.h io.h connection.h audio.h
 nodist_noinst_HEADERS = mumble.pb-c.h message_list.h
 
-cmumble_SOURCES = cmumble.c message.c varint.c io.c connection.c
+cmumble_SOURCES = cmumble.c message.c varint.c io.c connection.c audio.c
 nodist_cmumble_SOURCES = mumble.pb-c.c
 
 cmumble_LDADD = $(PROTOBUF_LIBS) $(GLIB_LIBS) $(GIO_LIBS) \
diff --git a/src/audio.c b/src/audio.c
new file mode 100644
index 0000000..519dc04
--- /dev/null
+++ b/src/audio.c
@@ -0,0 +1,205 @@
+#include "audio.h"
+#include "varint.h"
+#include "cmumble.h"
+#include <string.h>
+
+#define SAMPLERATE 48000
+#define CHANNELS 1
+
+void
+cmumble_audio_push(struct context *ctx, struct user *user,
+		   const uint8_t *data, gsize size)
+{
+	GstBuffer *gstbuf;
+	
+	gstbuf = gst_app_buffer_new(g_memdup(data, size), size, g_free, NULL);
+	gst_app_src_push_buffer(user->src, gstbuf);
+}
+
+static GstFlowReturn
+pull_buffer(GstAppSink *sink, gpointer user_data)
+{
+	struct context *ctx = user_data;
+	GstBuffer *buf;
+	uint8_t data[1024];
+	uint32_t write = 0, pos = 0;
+	MumbleProto__UDPTunnel tunnel;
+	static int seq = 0;
+
+	buf = gst_app_sink_pull_buffer(ctx->audio.sink);
+
+	if (++seq <= 2) {
+		gst_buffer_unref(buf);
+		return GST_FLOW_OK;
+	}
+	if (GST_BUFFER_SIZE(buf) > 127) {
+		g_printerr("GOT TOO BIG BUFFER\n");
+		return GST_FLOW_ERROR;
+	}
+
+	data[pos++] = (udp_voice_celt_alpha) | (0 << 4);
+
+	encode_varint(&data[pos], &write, ++ctx->sequence, 1024-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);
+
+	gst_buffer_unref(buf);
+
+	mumble_proto__udptunnel__init(&tunnel);
+	tunnel.packet.data = data;
+	tunnel.packet.len = pos;
+	cmumble_send_msg(ctx, &tunnel.base);
+
+	return GST_FLOW_OK;
+}
+
+static int
+setup_recording_gst_pipeline(struct context *ctx)
+{
+	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";
+
+	pipeline = gst_parse_launch(desc, &error);
+	if (error) {
+		g_printerr("Failed to create pipeline: %s\n", error->message);
+		return -1;
+	}
+	sink = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
+	ctx->audio.sink = GST_APP_SINK(sink);
+	ctx->audio.record_pipeline = pipeline;
+
+	cutter = gst_bin_get_by_name(GST_BIN(pipeline), "cutter");
+	g_object_set(G_OBJECT(cutter),
+		     "threshold_dB", -45.0, "leaky", TRUE, NULL);
+	
+	gst_app_sink_set_emit_signals(ctx->audio.sink, TRUE);
+	gst_app_sink_set_drop(ctx->audio.sink, FALSE);;
+	g_signal_connect(sink, "new-buffer", G_CALLBACK(pull_buffer), ctx);
+	
+	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(ctx->audio.sink, caps);
+	gst_caps_unref(caps);
+
+	gst_element_set_state(pipeline, GST_STATE_PLAYING);
+
+	ctx->sequence = 0;
+
+	return 0;
+}
+
+static void
+set_pulse_states(gpointer data, gpointer user_data)
+{
+	GstElement *elm = data;
+	struct user *user = user_data;
+	GstStructure *props;
+	gchar *name;
+
+	if (g_strcmp0(G_OBJECT_TYPE_NAME(elm), "GstPulseSink") != 0 ||
+	    g_object_class_find_property(G_OBJECT_GET_CLASS(elm),
+					 "stream-properties") == NULL)
+		goto out;
+
+	/* configure pulseaudio to use:
+	 * load-module module-device-manager "do_routing=1"
+	 * or new users may join to default output which is not headset?
+	 * Also consider setting device.intended_roles = "phone" for your
+	 * wanted default output (if you dont have a usb headset dev). */
+
+	name = g_strdup_printf("cmumble [%s]", user->name);
+
+	props = gst_structure_new("props",
+				  "application.name", G_TYPE_STRING, name,
+				  "media.role", G_TYPE_STRING, "phone",
+				  NULL);
+					
+	g_object_set(elm, "stream-properties", props, NULL);
+	gst_structure_free(props);
+	g_free(name);
+
+out:
+	g_object_unref(G_OBJECT(elm));
+}
+
+int
+cmumble_audio_create_playback_pipeline(struct context *ctx, struct user *user)
+{
+	GstElement *pipeline, *sink_bin;
+	GError *error = NULL;
+	char *desc = "appsrc name=src ! celtdec ! audioconvert ! autoaudiosink name=sink";
+
+	pipeline = gst_parse_launch(desc, &error);
+	if (error) {
+		g_printerr("Failed to create pipeline: %s\n", error->message);
+		return -1;
+	}
+
+	user->pipeline = pipeline;
+	user->src = GST_APP_SRC(gst_bin_get_by_name(GST_BIN(pipeline), "src"));
+
+	/* Important! */
+	gst_base_src_set_live(GST_BASE_SRC(user->src), TRUE); 
+	gst_base_src_set_do_timestamp(GST_BASE_SRC(user->src), TRUE);
+	gst_base_src_set_format(GST_BASE_SRC(user->src), GST_FORMAT_TIME);
+
+	gst_app_src_set_stream_type(user->src, GST_APP_STREAM_TYPE_STREAM); 
+
+	gst_element_set_state(pipeline, GST_STATE_PLAYING);
+
+	sink_bin = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
+	GstIterator *iter = gst_bin_iterate_sinks(GST_BIN(sink_bin));
+	gst_iterator_foreach(iter, set_pulse_states, user);
+	gst_iterator_free(iter);
+
+	/* Setup Celt Decoder */
+	cmumble_audio_push(ctx, user,
+			   ctx->audio.celt_header_packet, sizeof(CELTHeader));
+	/* fake vorbiscomment buffer */
+	cmumble_audio_push(ctx, user, NULL, 0);
+
+	return 0;
+}
+
+static int
+setup_playback_gst_pipeline(struct context *ctx)
+{
+	ctx->audio.celt_mode = celt_mode_create(SAMPLERATE,
+					  SAMPLERATE / 100, NULL);
+	celt_header_init(&ctx->audio.celt_header, ctx->audio.celt_mode, CHANNELS);
+	celt_header_to_packet(&ctx->audio.celt_header,
+			      ctx->audio.celt_header_packet, sizeof(CELTHeader));
+
+	return 0;
+}
+
+int
+cmumble_audio_init(struct context *ctx)
+{
+	if (setup_playback_gst_pipeline(ctx) < 0)
+		return -1;
+
+	if (setup_recording_gst_pipeline(ctx) < 0)
+		return -1;
+
+	return 0;
+}
+
+int
+cmumble_audio_fini(struct context *ctx)
+{
+
+	return 0;
+}
+
diff --git a/src/audio.h b/src/audio.h
new file mode 100644
index 0000000..b31c77f
--- /dev/null
+++ b/src/audio.h
@@ -0,0 +1,40 @@
+#ifndef _AUDIO_H_
+#define _AUDIO_H_
+
+#include <glib.h>
+
+#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>
+
+struct cmumble_audio {
+	GstElement *record_pipeline;
+	GstAppSink *sink;
+
+	uint8_t celt_header_packet[sizeof(CELTHeader)];
+	CELTHeader celt_header;
+	CELTMode *celt_mode;
+};
+
+struct context;
+struct user;
+
+int
+cmumble_audio_init(struct context *ctx);
+
+int
+cmumble_audio_fini(struct context *ctx);
+
+int
+cmumble_audio_create_playback_pipeline(struct context *ctx,
+				       struct user *user);
+
+void
+cmumble_audio_push(struct context *ctx, struct user *user,
+		   const uint8_t *data, gsize size);
+
+#endif /* _AUDIO_H_ */
diff --git a/src/cmumble.c b/src/cmumble.c
index 3a40e21..3d060ce 100644
--- a/src/cmumble.c
+++ b/src/cmumble.c
@@ -23,55 +23,6 @@ find_user(struct context *ctx, uint32_t session)
 }
 
 static void
-appsrc_push(GstAppSrc *src, const void *mem, size_t size)
-{
-	GstBuffer *gstbuf;
-	
-	gstbuf = gst_app_buffer_new(g_memdup(mem, size), size, g_free, NULL);
-	gst_app_src_push_buffer(src, gstbuf);
-}
-
-static GstFlowReturn
-pull_buffer(GstAppSink *sink, gpointer user_data)
-{
-	struct context *ctx = user_data;
-	GstBuffer *buf;
-	uint8_t data[1024];
-	uint32_t write = 0, pos = 0;
-	MumbleProto__UDPTunnel tunnel;
-	static int seq = 0;
-
-	buf = gst_app_sink_pull_buffer(ctx->sink);
-
-	if (++seq <= 2) {
-		gst_buffer_unref(buf);
-		return GST_FLOW_OK;
-	}
-	if (GST_BUFFER_SIZE(buf) > 127) {
-		g_printerr("GOT TOO BIG BUFFER\n");
-		return GST_FLOW_ERROR;
-	}
-
-	data[pos++] = (udp_voice_celt_alpha) | (0 << 4);
-
-	encode_varint(&data[pos], &write, ++ctx->sequence, 1024-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);
-
-	gst_buffer_unref(buf);
-
-	mumble_proto__udptunnel__init(&tunnel);
-	tunnel.packet.data = data;
-	tunnel.packet.len = pos;
-	cmumble_send_msg(ctx, &tunnel.base);
-
-	return GST_FLOW_OK;
-}
-
-static void
 recv_udp_tunnel(MumbleProto__UDPTunnel *tunnel, struct context *ctx)
 {
 	int64_t session, sequence;
@@ -100,7 +51,7 @@ recv_udp_tunnel(MumbleProto__UDPTunnel *tunnel, struct context *ctx)
 		if (frame_len == 0 || frame_len > len-pos)
 			break;
 
-		appsrc_push(user->src, &data[pos], frame_len);
+		cmumble_audio_push(ctx, user, &data[pos], frame_len);
 
 		pos += frame_len;
 		sequence++;
@@ -177,9 +128,6 @@ recv_user_remove(MumbleProto__UserRemove *remove, struct context *ctx)
 	}
 }
 
-static int
-user_create_playback_pipeline(struct context *ctx, struct user *user);
-
 static void
 recv_user_state(MumbleProto__UserState *state, struct context *ctx)
 {
@@ -200,7 +148,8 @@ recv_user_state(MumbleProto__UserState *state, struct context *ctx)
 	user->name = g_strdup(state->name);
 	user->user_id = state->user_id;
 
-	user_create_playback_pipeline(ctx, user);
+
+	cmumble_audio_create_playback_pipeline(ctx, user);
 	g_print("receive user: %s\n", user->name);
 	ctx->users = g_list_prepend(ctx->users, user);
 }
@@ -239,135 +188,6 @@ static const struct {
 	.SuggestConfig		= NULL,
 };
 
-static void
-set_pulse_states(gpointer data, gpointer user_data)
-{
-	GstElement *elm = data;
-	struct user *user = user_data;
-	GstStructure *props;
-	gchar *name;
-
-	if (g_strcmp0(G_OBJECT_TYPE_NAME(elm), "GstPulseSink") != 0 ||
-	    g_object_class_find_property(G_OBJECT_GET_CLASS(elm),
-					 "stream-properties") == NULL)
-		goto out;
-
-	/* configure pulseaudio to use:
-	 * load-module module-device-manager "do_routing=1"
-	 * or new users may join to default output which is not headset?
-	 * Also consider setting device.intended_roles = "phone" for your
-	 * wanted default output (if you dont have a usb headset dev). */
-
-	name = g_strdup_printf("cmumble [%s]", user->name);
-
-	props = gst_structure_new("props",
-				  "application.name", G_TYPE_STRING, name,
-				  "media.role", G_TYPE_STRING, "phone",
-				  NULL);
-					
-	g_object_set(elm, "stream-properties", props, NULL);
-	gst_structure_free(props);
-	g_free(name);
-
-out:
-	g_object_unref(G_OBJECT(elm));
-}
-
-static int
-user_create_playback_pipeline(struct context *ctx, struct user *user)
-{
-	GstElement *pipeline, *sink_bin;
-	GError *error = NULL;
-	char *desc = "appsrc name=src ! celtdec ! audioconvert ! autoaudiosink name=sink";
-
-	pipeline = gst_parse_launch(desc, &error);
-	if (error) {
-		g_printerr("Failed to create pipeline: %s\n", error->message);
-		return -1;
-	}
-
-	user->pipeline = pipeline;
-	user->src = GST_APP_SRC(gst_bin_get_by_name(GST_BIN(pipeline), "src"));
-
-	/* Important! */
-	gst_base_src_set_live(GST_BASE_SRC(user->src), TRUE); 
-	gst_base_src_set_do_timestamp(GST_BASE_SRC(user->src), TRUE);
-	gst_base_src_set_format(GST_BASE_SRC(user->src), GST_FORMAT_TIME);
-
-	gst_app_src_set_stream_type(user->src, GST_APP_STREAM_TYPE_STREAM); 
-
-	gst_element_set_state(pipeline, GST_STATE_PLAYING);
-
-	sink_bin = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
-	GstIterator *iter = gst_bin_iterate_sinks(GST_BIN(sink_bin));
-	gst_iterator_foreach(iter, set_pulse_states, user);
-	gst_iterator_free(iter);
-
-	/* Setup Celt Decoder */
-	appsrc_push(user->src, ctx->celt_header_packet, sizeof(CELTHeader));
-	/* fake vorbiscomment buffer */
-	appsrc_push(user->src, NULL, 0);
-
-	return 0;
-}
-
-static int
-setup_playback_gst_pipeline(struct context *ctx)
-{
-#define SAMPLERATE 48000
-#define CHANNELS 1
-	ctx->celt_mode = celt_mode_create(SAMPLERATE,
-					  SAMPLERATE / 100, NULL);
-	celt_header_init(&ctx->celt_header, ctx->celt_mode, CHANNELS);
-	celt_header_to_packet(&ctx->celt_header,
-			      ctx->celt_header_packet, sizeof(CELTHeader));
-
-	return 0;
-}
-
-static int
-setup_recording_gst_pipeline(struct context *ctx)
-{
-	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";
-
-	pipeline = gst_parse_launch(desc, &error);
-	if (error) {
-		g_printerr("Failed to create pipeline: %s\n", error->message);
-		return -1;
-	}
-	sink = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
-	ctx->sink = GST_APP_SINK(sink);
-	ctx->record_pipeline = pipeline;
-
-	cutter = gst_bin_get_by_name(GST_BIN(pipeline), "cutter");
-	g_object_set(G_OBJECT(cutter),
-		     "threshold_dB", -45.0, "leaky", TRUE, NULL);
-	
-	gst_app_sink_set_emit_signals(ctx->sink, TRUE);
-	gst_app_sink_set_drop(ctx->sink, FALSE);;
-	g_signal_connect(sink, "new-buffer", G_CALLBACK(pull_buffer), ctx);
-	
-	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(ctx->sink, caps);
-	gst_caps_unref(caps);
-
-	gst_element_set_state(pipeline, GST_STATE_PLAYING);
-
-	ctx->sequence = 0;
-
-	return 0;
-}
-
 int main(int argc, char **argv)
 {
 	char *host = "localhost";
@@ -411,12 +231,8 @@ int main(int argc, char **argv)
 
 	gst_init(&argc, &argv);
 
-	if (setup_playback_gst_pipeline(&ctx) < 0)
+	if (cmumble_audio_init(&ctx) < 0)
 		return 1;
-
-	if (setup_recording_gst_pipeline(&ctx) < 0)
-		return 1;
-
 	cmumble_io_init(&ctx);
 
 	g_main_loop_run(ctx.loop);
@@ -424,6 +240,7 @@ int main(int argc, char **argv)
 	g_main_loop_unref(ctx.loop);
 
 	cmumble_io_fini(&ctx);
+	cmumble_audio_init(&ctx);
 	cmumble_connection_fini(&ctx);
 
 	return 0;
diff --git a/src/cmumble.h b/src/cmumble.h
index 5de3fb1..2ea1203 100644
--- a/src/cmumble.h
+++ b/src/cmumble.h
@@ -1,41 +1,28 @@
 #ifndef _CMUMBLE_H_
 #define _CMUMBLE_H_
 
-#include <gst/gst.h>
-#include <gst/app/gstappsrc.h>
-#include <gst/app/gstappsink.h>
-#include <gst/app/gstappbuffer.h>
-
 #include <glib.h>
 #include <glib-object.h>
 #include <gio/gio.h>
 
-#include <celt/celt.h>
-#include <celt/celt_header.h>
-
 #include "mumble.pb-c.h"
 #include "message.h"
 #include "io.h"
 #include "connection.h"
+#include "audio.h"
 
 typedef void (*callback_t)(ProtobufCMessage *msg, struct context *);
 
 struct context {
 	struct cmumble_connection con;
 	struct cmumble_io io;
+	struct cmumble_audio audio;
 	const callback_t *callbacks;
 	GMainLoop *loop;
 
 	uint32_t session;
 	gboolean authenticated;
 
-	uint8_t celt_header_packet[sizeof(CELTHeader)];
-	CELTHeader celt_header;
-	CELTMode *celt_mode;
-
-	GstElement *record_pipeline;
-	GstAppSink *sink;
-
 	int64_t sequence;
 
 	GList *users;