From 2f5ca7693694fe917b9e322f6dc1692a5cbaacbf Mon Sep 17 00:00:00 2001
From: hark <hark@puscii.nl>
Date: Fri, 25 Mar 2016 03:33:09 +0000
Subject: experimental gstreamer audiobackend (not working)

---
 src/conversation/gnunet_gst_def.h | 190 ++++++++++++++++++++++++++++++++++++++
 1 file changed, 190 insertions(+)
 create mode 100644 src/conversation/gnunet_gst_def.h

(limited to 'src/conversation/gnunet_gst_def.h')

diff --git a/src/conversation/gnunet_gst_def.h b/src/conversation/gnunet_gst_def.h
new file mode 100644
index 000000000..2e6903db4
--- /dev/null
+++ b/src/conversation/gnunet_gst_def.h
@@ -0,0 +1,190 @@
+#include <getopt.h>
+#include <string.h> 
+#include <stdio.h>
+#include <ctype.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <time.h>
+#include <regex.h> 
+
+
+#include "gnunet/platform.h"
+#include "gnunet/gnunet_util_lib.h"
+#include "gnunet/gnunet_protocols.h"
+//#include "gnunet/conversation.h" doesn't get installed
+#include "conversation.h"
+#include "gnunet/gnunet_constants.h"
+#include "gnunet/gnunet_core_service.h"
+#include "gnunet/gnunet_common.h"
+
+/*
+#include <gst/gst.h>
+#include <gst/audio/gstaudiobasesrc.h>
+#include <gst/app/gstappsrc.h>
+*/
+
+/* huh
+#include <glib-2.0/glib.h>
+
+#include <gstreamer-1.0/gst/gst.h>
+#include <gstreamer-1.0/gst/pbutils/pbutils.h>
+#include <gstreamer-1.0/gst/video/videooverlay.h>
+#include <gstreamer-1.0/gst/audio/gstaudiobasesrc.h>
+#include <gstreamer-1.0/gst/app/gstappsrc.h>
+*/
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiobasesrc.h>
+#include <gst/app/gstappsrc.h>
+#include <glib.h>
+#include <gst/app/gstappsink.h>
+
+// sockets
+#include <netinet/in.h>
+#include <sys/socket.h>
+#include <fcntl.h>
+#include <arpa/inet.h>
+
+#include <sys/types.h>
+#include <sys/socket.h>
+#include <netdb.h>
+
+
+//glib stuff
+//#include <glib.h>
+#include <glib-2.0/glib/gprintf.h>
+#include <glib-unix.h>
+
+// static struct AudioMessage *audio_message;
+
+
+
+typedef struct GNUNET_gstData GNUNET_gstData;
+struct GNUNET_gstData {
+  //general
+  GstPipeline *pipeline;
+
+  // things
+  struct AudioMessage *audio_message;
+  struct GNUNET_SERVER_MessageStreamTokenizer *stdin_mst;
+  GstElement *appsrc;
+  GstElement *appsink;
+  //settings
+  int audiobackend;  
+  int dropsilence;
+  int usertp;
+  int pure_ogg;
+  char *jack_pp_in;
+  char *jack_pp_out;
+};
+
+
+
+
+#define DEBUG_READ_PURE_OGG 1
+#define DEBUG_RECORD_PURE_OGG 1
+
+
+/**
+ * How much data to read in one go
+ */
+#define MAXLINE 4096
+
+/**
+ * Max number of microseconds to buffer in audiosink.
+ * Default is 1000
+ */
+#define BUFFER_TIME 1000
+
+/**
+ * Min number of microseconds to buffer in audiosink.
+ * Default is 1000
+ */
+#define LATENCY_TIME 1000
+
+
+/**
+ * Number of channels.
+ * Must be one of the following (from libopusenc documentation):
+ * 1, 2
+ */
+#define OPUS_CHANNELS 1
+
+/**
+ * Maximal size of a single opus packet.
+ */
+#define MAX_PAYLOAD_SIZE (1024 / OPUS_CHANNELS)
+
+/**
+ * Size of a single frame fed to the encoder, in ms.
+ * Must be one of the following (from libopus documentation):
+ * 2.5, 5, 10, 20, 40 or 60
+ */
+#define OPUS_FRAME_SIZE 40
+
+/**
+ * Expected packet loss to prepare for, in percents.
+ */
+#define PACKET_LOSS_PERCENTAGE 1
+
+/**
+ * Set to 1 to enable forward error correction.
+ * Set to 0 to disable.
+ */
+#define INBAND_FEC_MODE 1
+
+/**
+ * Max number of microseconds to buffer in audiosource.
+ * Default is 200000
+ */
+#define BUFFER_TIME 1000 /* 1ms */
+
+/**
+ * Min number of microseconds to buffer in audiosource.
+ * Default is 10000
+ */
+#define LATENCY_TIME 1000 /* 1ms */
+
+/**
+ * Maximum delay in multiplexing streams, in ns.
+ * Setting this to 0 forces page flushing, which
+ * decreases delay, but increases overhead.
+ */
+#define OGG_MAX_DELAY 0
+
+/**
+ * Maximum delay for sending out a page, in ns.
+ * Setting this to 0 forces page flushing, which
+ * decreases delay, but increases overhead.
+ */
+#define OGG_MAX_PAGE_DELAY 0
+
+#define SAMPLING_RATE 48000
+
+enum {
+    AUTO,
+    JACK,
+    ALSA,
+    FAKE,
+    TEST
+};
+
+enum {
+    SOURCE,
+    SINK
+};
+
+enum {
+    ENCODER,
+    DECODER
+};
+
+enum {
+    FAIL,
+    OK
+};
+
+enum {
+    SPEAKER,
+    MICROPHONE
+};
-- 
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