Add source files

7 files changed, 1120 insertions, 0 deletions

diff --git a/src/debug/local-play.c b/src/debug/local-play.c
new file mode 100644
index 0000000..497d567
--- /dev/null
+++ b/src/debug/local-play.c
@@ -0,0 +1,147 @@
+/* License: LGPL-2.1
+ * vim: set sts=2 sw=2 et : */
+
+#include "local-play.h"
+
+static void
+pad_has_video_caps (TranscodeServerCtx *ctx,
+                    GstPad             *decodebin_pad)
+{
+  GstPad *sink_pad;
+  GstElement *bin;
+  GstElement *videoconv, *videoscale, *videosink;
+
+  bin = gst_bin_new (NULL);
+  videoconv = gst_element_factory_make ("videoconvert", "videoconv");
+  videoscale = gst_element_factory_make ("videoscale", "videoscale");
+  videosink = gst_element_factory_make ("autovideosink", "videosink");
+
+  gst_bin_add_many (GST_BIN (bin), videoconv, videoscale, videosink, NULL);
+  gst_element_link_many (videoconv, videoscale, videosink, NULL);
+
+  sink_pad = gst_element_get_static_pad (videoconv, "sink");
+  gst_element_add_pad (bin, gst_ghost_pad_new ("sink", sink_pad));
+  gst_object_unref (sink_pad);
+
+  gst_bin_add (GST_BIN (ctx->pipeline), bin);
+
+  sink_pad = gst_element_get_static_pad (bin, "sink");
+  if (gst_pad_link (decodebin_pad, sink_pad) != GST_PAD_LINK_OK) {
+    g_critical ("ERROR: Unable to link video pads");
+    soup_message_set_status (ctx->msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+    goto out;
+  }
+
+  /* Since these elements have been added to the pipeline after the state was
+   * changed from GST_STATE_NULL, they have to be moved to PAUSED manually */
+  gst_element_set_state (bin, GST_STATE_PAUSED);
+out:
+  gst_object_unref (sink_pad);
+  return;
+}
+
+static void
+pad_has_audio_caps (TranscodeServerCtx *ctx,
+                    GstPad             *decodebin_pad)
+{
+  GstPad *sink_pad;
+  GstElement *bin;
+  GstElement *audioconv, *audioresample, *audiosink;
+
+  bin = gst_bin_new (NULL);
+  audioconv = gst_element_factory_make ("audioconvert", "audioconv");
+  audioresample = gst_element_factory_make ("audioresample", "audioresample");
+  audiosink = gst_element_factory_make ("autoaudiosink", "audiosink");
+
+  gst_bin_add_many (GST_BIN (bin), audioconv, audioresample, audiosink, NULL);
+  gst_element_link_many (audioconv, audioresample, audiosink, NULL);
+
+  sink_pad = gst_element_get_static_pad (audioconv, "sink");
+  gst_element_add_pad (bin, gst_ghost_pad_new ("sink", sink_pad));
+  gst_object_unref (sink_pad);
+
+  gst_bin_add (GST_BIN (ctx->pipeline), bin);
+
+  sink_pad = gst_element_get_static_pad (bin, "sink");
+  if (gst_pad_link (decodebin_pad, sink_pad) != GST_PAD_LINK_OK) {
+    g_critical ("ERROR: Unable to link video pads");
+    soup_message_set_status (ctx->msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+    goto out;
+  }
+
+  /* Since these elements have been added to the pipeline after the state was
+   * changed from GST_STATE_NULL, they have to be moved to PAUSED manually */
+  gst_element_set_state (bin, GST_STATE_PAUSED);
+out:
+  gst_object_unref (sink_pad);
+  return;
+}
+
+static gboolean
+structure_printf (GQuark field_id,
+                  const GValue *value,
+                  gpointer user_data)
+{
+  char *char_value;
+
+  char_value = g_strdup_value_contents (value);
+  g_print ("%s = %s\n", g_quark_to_string (field_id), char_value);
+  g_free (char_value);
+  return TRUE;
+}
+
+static void
+decodebin_pad_added (GstElement *decodebin,
+                     GstPad     *src_pad,
+                     gpointer    user_data)
+{
+  GstCaps *pad_caps;
+  GstStructure *structure;
+  TranscodeServerCtx *ctx = user_data;
+
+  if (!gst_pad_has_current_caps (src_pad)) {
+    g_critical ("Decodebin pad doesn't have current caps");
+    soup_message_set_status (ctx->msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+    return;
+  }
+
+  pad_caps = gst_pad_get_current_caps (src_pad);
+  structure = gst_caps_get_structure (pad_caps, 0);
+  gst_structure_foreach (structure, structure_printf, NULL);
+
+  if (gst_structure_has_name (structure, "audio/x-raw"))
+    pad_has_audio_caps (ctx, src_pad);
+  else if (gst_structure_has_name (structure, "video/x-raw"))
+    pad_has_video_caps (ctx, src_pad);
+  else
+    g_critical ("Found unsupported caps: %s",
+                gst_structure_get_name (structure));
+
+  gst_caps_unref (pad_caps);
+}
+
+void
+stp_play_from_msg (TranscodeServerCtx *ctx)
+{
+  GstBus *bus;
+  GstElement *src, *decodebin;
+
+  g_debug ("Constructing pipeline\n");
+  src = gst_element_factory_make ("appsrc", "src");
+  g_object_set (src, "is-live", TRUE,
+                "stream-type", 0, NULL);
+  ctx->appsrc = src;
+
+  decodebin = gst_element_factory_make ("decodebin", "decodebin");
+  gst_bin_add_many (GST_BIN (ctx->pipeline), src, decodebin, NULL);
+  gst_element_link (src, decodebin);
+
+  g_signal_connect (decodebin, "pad-added",
+                    G_CALLBACK (decodebin_pad_added), ctx);
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE (ctx->pipeline));
+  gst_bus_add_signal_watch (bus);
+  g_signal_connect (bus, "message", G_CALLBACK (stp_on_gst_bus_message), ctx);
+  g_object_unref (bus);
+}
+
diff --git a/src/debug/local-play.h b/src/debug/local-play.h
new file mode 100644
index 0000000..a56a572
--- /dev/null
+++ b/src/debug/local-play.h
@@ -0,0 +1,11 @@
+/* License: LGPL-2.1
+ * vim: set sts=2 sw=2 et : */
+
+#ifndef _SST_LOCAL_PLAY
+#define _SST_LOCAL_PLAY
+
+#include "../lib.h"
+
+void stp_play_from_msg (TranscodeServerCtx *ctx);
+
+#endif /* _SST_LOCAL_PLAY */
diff --git a/src/encode.c b/src/encode.c
new file mode 100644
index 0000000..caaaa46
--- /dev/null
+++ b/src/encode.c
@@ -0,0 +1,166 @@
+/* License: LGPL-2.1
+ * vim: set sts=2 sw=2 et : */
+
+#include "encode.h"
+
+#include <gst/pbutils/encoding-profile.h>
+
+static gboolean
+on_autoplug_continue (GstElement *decodebin,
+                      GstPad     *srcpad,
+                      GstCaps    *caps,
+                      GstElement *encodebin)
+{
+  gboolean ret;
+  char *name;
+  GstPad *sinkpad;
+
+  name = gst_caps_to_string (caps);
+  g_signal_emit_by_name (encodebin, "request-pad", caps, &sinkpad);
+
+  if (sinkpad != NULL)
+    ret = FALSE, g_debug ("encodebin can passthrough %s\n", name);
+  else
+    ret = TRUE, g_debug ("encodebin cannot passthrough %s\n", name);
+
+  g_free (name);
+  return ret;
+}
+
+static void
+on_decodebin_pad_added (GstElement *decodebin,
+                        GstPad     *srcpad,
+                        GstElement *encodebin)
+{
+  GstCaps *caps;
+  GstPad *sinkpad = NULL;
+  char *name;
+
+  caps = gst_pad_query_caps (srcpad, NULL);
+  name = gst_caps_to_string (caps);
+
+  /* FIXME: We don't try to fetch a compatible pad for raw audio because
+   * that somehow always fails to link. Transmageddon does the same. */
+  if (!g_str_has_prefix (name, "audio/x-raw"))
+    /* If we successfully requested a compatible sink pad in 
+     * "autoplug-continue", we can fetch that here. */
+    sinkpad = gst_element_get_compatible_pad (encodebin, srcpad, NULL);
+
+  if (!sinkpad) {
+    /* We request a sink pad for the decoded stream */
+    g_signal_emit_by_name (encodebin, "request-pad", caps, &sinkpad);
+    if (!sinkpad) {
+      g_printerr ("Failed to request a new sink pad for %s\n", name);
+      goto out;
+    }
+    g_debug ("Requested a new sink pad for %s\n", name);
+  } else {
+    g_debug ("Found an existing sink pad for %s\n", name);
+  }
+
+  if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
+    g_printerr ("Couldn't link pads for %s\n", name);
+
+out:
+  g_free (name);
+  gst_caps_unref (caps);
+  return;
+}
+
+static GstEncodingProfile *
+create_webm_profile (void)
+{
+  GstEncodingProfile *t;
+  GstEncodingContainerProfile *prof;
+  GstCaps *caps;
+  GstPreset *vp8preset;
+
+  caps = gst_caps_from_string ("video/webm");
+  prof = gst_encoding_container_profile_new ("WebM audio/video", "Standard WEBM/VP8/VORBIS", caps, NULL);
+  gst_caps_unref (caps);
+
+  vp8preset = GST_PRESET (gst_element_factory_make ("vp8enc", "vp8preset"));
+  /* FIXME: This thing still doesn't encode fast enough in real time */
+  g_object_set (vp8preset,
+                "cpu-used", 5,
+                "end-usage", 1,
+                "max-quantizer", 56,
+                "threads", 4,
+                "undershoot", 95,
+                NULL);
+  gst_preset_save_preset (vp8preset, "stp_vp8preset");
+
+  caps = gst_caps_from_string ("video/x-vp8");
+  t = (GstEncodingProfile*) gst_encoding_video_profile_new (caps, "stp_vp8preset",
+                                                            NULL, 0);
+  gst_encoding_container_profile_add_profile (prof, t);
+  gst_caps_unref (caps);
+
+  caps = gst_caps_from_string ("audio/x-vorbis");
+  t = (GstEncodingProfile*) gst_encoding_audio_profile_new (caps, NULL, NULL, 0);
+  gst_encoding_container_profile_add_profile (prof, t);
+  gst_caps_unref (caps);
+
+  /* What about application/x-ass? */
+
+  return (GstEncodingProfile*) prof;
+}
+
+void
+stp_encode_from_msg (TranscodeServerCtx *ctx)
+{
+  GstBus *bus;
+  GstElement *src, *decodebin, *encodebin;
+  GstElement *tee, *q1, *fakesink;
+  GstEncodingProfile *profile;
+
+  g_debug ("Constructing pipeline\n");
+  
+  src = gst_element_factory_make ("appsrc", "src");
+  g_object_set (src, "is-live", TRUE,
+                "emit-signals", FALSE,
+                "stream-type", 0, NULL);
+  ctx->appsrc = src;
+
+  decodebin = gst_element_factory_make ("decodebin", "decodebin");
+  
+  gst_bin_add_many (GST_BIN (ctx->pipeline), src, decodebin, NULL);
+  gst_element_link (src, decodebin);
+
+  /* TODO: Allow setting of a scaling factor to
+   * allow realtime encoding of all streams */
+  profile = create_webm_profile ();
+  encodebin = gst_element_factory_make ("encodebin", "encodebin");
+  g_object_set (encodebin,
+                "profile", profile,
+                "avoid-reencoding", TRUE, NULL);
+  tee = gst_element_factory_make ("tee", "tee");
+  q1 = gst_element_factory_make ("queue", "q1");
+#ifndef DEBUG
+  fakesink = gst_element_factory_make ("filesink", "filesink");
+  g_object_set (fakesink, "location", "debug-output.webm", NULL);
+#else
+  fakesink = gst_element_factory_make ("fakesink", "fakesink");
+#endif
+
+  gst_bin_add_many (GST_BIN (ctx->pipeline), encodebin, tee, q1, fakesink, NULL);
+  gst_element_link_many (encodebin, tee, q1, fakesink, NULL);
+
+  /* The pads of decodebin and encodebin are dynamic,
+   * so those will be linked when streams/pads are added */
+
+  /* When decodebin finds a stream that can be decoded, we check 
+   * if we can pass that directly to encodebin instead of letting
+   * decodebin find a decoding element automatically */
+  g_signal_connect (decodebin, "autoplug-continue",
+                    G_CALLBACK (on_autoplug_continue), encodebin);
+  /* When decodebin exposes a source pad, we need to request a
+   * corresponding sink pad on decodebin */
+  g_signal_connect (decodebin, "pad-added",
+                    G_CALLBACK (on_decodebin_pad_added), encodebin);
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE (ctx->pipeline));
+  gst_bus_add_signal_watch (bus);
+  g_signal_connect (bus, "message", G_CALLBACK (stp_on_gst_bus_message), ctx);
+  g_object_unref (bus);
+}
diff --git a/src/encode.h b/src/encode.h
new file mode 100644
index 0000000..9c0d897
--- /dev/null
+++ b/src/encode.h
@@ -0,0 +1,11 @@
+/* License: LGPL-2.1
+ * vim: set sts=2 sw=2 et : */
+
+#ifndef _SST_ENCODE
+#define _SST_ENCODE
+
+#include "lib.h"
+
+void stp_encode_from_msg (TranscodeServerCtx *ctx);
+
+#endif /* _SST_ENCODE */
diff --git a/src/lib.c b/src/lib.c
new file mode 100644
index 0000000..584da39
--- /dev/null
+++ b/src/lib.c
@@ -0,0 +1,166 @@
+/* License: LGPL-2.1
+ * vim: set sts=2 sw=2 et : */
+
+#include "lib.h"
+
+#include <poll.h>
+#include <unistd.h>
+#include <sys/stat.h>
+#include <glib-object.h>
+#include <gio/gunixinputstream.h>
+
+gboolean
+stp_on_gst_bus_message (GstBus             *bus,
+                        GstMessage         *msg,
+                        TranscodeServerCtx *ctx)
+{
+  GError *error = NULL;
+  char *tmp = NULL;
+
+  switch (GST_MESSAGE_TYPE (msg)) {
+    case GST_MESSAGE_ERROR:
+      gst_message_parse_error (msg, &error, &tmp);
+      g_printerr ("ERROR from element %s: %s\n",
+                  GST_OBJECT_NAME (msg->src), error->message);
+      g_printerr ("Debug info: %s\n", tmp);
+      g_error_free (error);
+      g_free (tmp);
+      /* Setting the server response will only work if the request
+       * hasn't already finished, so we check that */
+      if (!ctx->request_finished)
+        soup_message_set_status (ctx->msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+      g_hash_table_remove (ctx->parent_ctx_table, ctx->path);
+      break;
+    case GST_MESSAGE_EOS:
+      g_print ("End of file\n");
+      g_hash_table_remove (ctx->parent_ctx_table, ctx->path);
+      break;
+    default:
+      //g_print ("%s\n", gst_message_type_get_name (msg->type));
+      break;
+  }
+
+  return TRUE;
+}
+
+void
+stp_cleanup_transcode_server_ctx (TranscodeServerCtx *ctx)
+{
+  g_print (">>> Doing server cleanup\n");
+
+  /* Closing the stream should send a HUP on the fd and
+   * trigger cleanup in write_get_response_body_chunk() */
+  g_list_foreach (ctx->clients, (GFunc)stp_close_client_ctx, NULL);
+  g_list_free (ctx->clients);
+
+  /* Cleanup gstreamer pipeline */
+  gst_element_set_state (ctx->pipeline, GST_STATE_NULL);
+  gst_object_unref (ctx->pipeline);
+  ctx->pipeline_is_playing = FALSE;
+
+  g_free (ctx);
+}
+
+static GstPadProbeReturn
+pad_blocked_cleanup_cb (GstPad             *srcpad,
+                        GstPadProbeInfo    *info,
+                        TranscodeClientCtx *ctx)
+{
+  GstPad *sinkpad;
+  GstElement *tee;
+  GstElement *sinkbin = GST_ELEMENT (gst_element_get_parent (ctx->appsink));
+
+  g_print (".");
+  /* Remove the probe, XXX: hence unblocking the pipeline? */
+  gst_pad_remove_probe (srcpad, GST_PAD_PROBE_INFO_ID (info));
+
+  tee = gst_pad_get_parent_element (srcpad);
+  sinkpad = gst_element_get_static_pad (sinkbin, "qsink");
+  gst_pad_unlink (srcpad, sinkpad);
+  gst_element_remove_pad (tee, srcpad);
+
+  gst_object_unref (sinkpad);
+  gst_object_unref (tee);
+
+  gst_bin_remove (GST_BIN (ctx->server_ctx->pipeline), sinkbin);
+  gst_object_unref (sinkbin);
+  gst_object_unref (ctx->appsink);
+
+  g_free (ctx);
+  g_print (" Client cleanup done!\n");
+  return GST_PAD_PROBE_OK;
+}
+
+void
+stp_cleanup_transcode_client_ctx (TranscodeClientCtx *ctx)
+{
+  GstPad *sinkpad, *srcpad;
+  TranscodeServerCtx *server_ctx = ctx->server_ctx;
+  GstElement *sinkbin = GST_ELEMENT (gst_element_get_parent (ctx->appsink));
+
+  g_print (">>> Doing client cleanup .");
+
+  /* If we're cleaning up because the server is shutting down,
+   * this will be NULL */
+  if (server_ctx)
+    server_ctx->clients = g_list_remove (server_ctx->clients, ctx);
+
+  /* Block sinkpad and srcpad, then unlink and remove */
+  sinkpad = gst_element_get_static_pad (sinkbin, "qsink");
+  srcpad = gst_pad_get_peer (sinkpad);
+
+  gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK,
+                     (GstPadProbeCallback) pad_blocked_cleanup_cb,
+                     ctx, NULL);
+  gst_object_unref (sinkbin);
+  g_print (".");
+}
+
+void
+stp_close_client_ctx (TranscodeClientCtx *ctx)
+{
+  g_print (">>> Doing client cleanup\n");
+
+  ctx->server_ctx = NULL;
+
+  g_free (ctx);
+}
+
+/* Returns a copy of the streamheader GstBuffer */
+GstBuffer*
+stp_get_streamheader_from_caps (GstCaps *caps)
+{
+  GArray *array;
+  GstStructure *s;
+  const GValue *value;
+
+  s = gst_caps_get_structure (caps, 0);
+  value = gst_structure_get_value (s, "streamheader");
+  g_return_val_if_fail (G_IS_VALUE (value), NULL);
+
+  array = g_value_peek_pointer (value);
+  value = &g_array_index (array, GValue, 0);
+  g_return_val_if_fail (G_VALUE_TYPE (value) == GST_TYPE_BUFFER, NULL);
+
+  return gst_buffer_copy (g_value_peek_pointer (value));
+}
+
+/* Returns a GstBuffer which has consumed the passed-in data */
+GstBuffer*
+stp_get_gst_buffer (SoupBuffer *chunk)
+{
+  gsize len;
+  const guint8 *d;
+  SoupBuffer *copy;
+  GstBuffer *buffer;
+  GstMemory *memory;
+
+  copy = soup_buffer_copy (chunk);
+  soup_buffer_get_data (copy, &d, &len);
+  buffer = gst_buffer_new ();
+  memory = gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY,
+                                  (guint8*)d, len, 0, len, copy,
+                                  (GDestroyNotify)soup_buffer_free);
+  gst_buffer_append_memory (buffer, memory);
+  return buffer;
+}
diff --git a/src/lib.h b/src/lib.h
new file mode 100644
index 0000000..187bae7
--- /dev/null
+++ b/src/lib.h
@@ -0,0 +1,57 @@
+/* License: LGPL-2.1
+ * vim: set sts=2 sw=2 et : */
+
+#ifndef _SST_LIB
+#define _SST_LIB
+
+#include <glib.h>
+#include <gst/gst.h>
+#include <libsoup/soup.h>
+
+typedef struct _TranscodeServerCtx TranscodeServerCtx;
+typedef struct _TranscodeClientCtx TranscodeClientCtx;
+
+struct _TranscodeServerCtx {
+  SoupMessage *msg;
+  GstElement *pipeline;
+  GstElement *appsrc;
+  /* The pipeline status can be queried from the pipeline itself,
+   * but it's probably better for branching to have it separately too */
+  gboolean pipeline_is_playing;
+  /* Set to TRUE when the incoming stream ends; let's us know if
+   * we need to set the PUT response on EOS/ERROR on the pipeline */
+  gboolean request_finished;
+
+  /* List of client contexts */
+  GList *clients;
+  /* Reference to the parent context hash table */
+  GHashTable *parent_ctx_table;
+  /* Reference to the key in the parent hash table */
+  char *path;
+};
+
+struct _TranscodeClientCtx {
+  SoupMessage *msg;
+  GstElement *appsink;
+  gulong first_sample_handler_id;
+  /* The transcode server context */
+  TranscodeServerCtx *server_ctx;
+};
+
+void stp_cleanup_transcode_server_ctx (TranscodeServerCtx *ctx);
+void stp_cleanup_transcode_client_ctx (TranscodeClientCtx *ctx);
+void stp_close_client_ctx (TranscodeClientCtx *ctx);
+
+gboolean stp_on_gst_bus_message (GstBus             *bus,
+                                 GstMessage         *msg,
+                                 TranscodeServerCtx *ctx);
+
+gboolean stp_copy_chunk_to_fd (SoupBuffer *chunk,
+                               int         fd);
+gboolean stp_copy_fd_to_body (TranscodeClientCtx *ctx);
+gboolean stp_fd_has_data_to_read (int fd);
+
+GstBuffer* stp_get_streamheader_from_caps (GstCaps *caps);
+GstBuffer* stp_get_gst_buffer (SoupBuffer *chunk);
+
+#endif /* _SST_LIB */
diff --git a/src/main.c b/src/main.c
new file mode 100644
index 0000000..a2430f5
--- /dev/null
+++ b/src/main.c
@@ -0,0 +1,562 @@
+/* License: LGPL-2.1
+ * vim: set sts=2 sw=2 et : */
+
+#include "lib.h"
+#include "encode.h"
+
+#ifdef DEBUG
+#include "debug/local-play.h"
+#endif
+
+#include <fcntl.h>
+#include <string.h>
+
+static SoupServer *server;
+
+static TranscodeServerCtx*
+get_server_ctx_from_msg (SoupMessage *msg,
+                         GHashTable  *ctx_table)
+{
+  SoupURI *uri;
+  TranscodeServerCtx *ctx;
+
+  g_object_get (msg, "uri", &uri, NULL);
+  ctx = g_hash_table_lookup (ctx_table, soup_uri_get_path (uri));
+  soup_uri_free (uri);
+
+  return ctx;
+}
+
+static void
+enough_data_pause_message (GstElement         *appsrc,
+                           guint               size,
+                           TranscodeServerCtx *ctx)
+{
+  /* Pausing the request can cause weirdness if
+   * the recv rate is too high. So, we don't do this. */
+  //g_print ("Enough data, pause. ");
+  //soup_server_pause_message (server, ctx->msg);
+}
+
+static void
+need_data_unpause_message (GstElement         *appsrc,
+                           guint               size,
+                           TranscodeServerCtx *ctx)
+{
+  //g_print ("Need data, unpause. \n");
+  //soup_server_unpause_message (server, ctx->msg);
+}
+
+static GstPadProbeReturn
+tee_src_pad_blocked_cb (GstPad             *srcpad,
+                        GstPadProbeInfo    *info,
+                        TranscodeClientCtx *ctx)
+{
+  GstCaps *caps;
+  GstPad *sinkpad;
+  GstBuffer *buffer;
+  GstState state;
+  GstStateChangeReturn ret;
+  TranscodeServerCtx *server_ctx = ctx->server_ctx;
+  GstElement *sinkbin = GST_ELEMENT (gst_element_get_parent (ctx->appsink));
+
+  /* Remove the probe, XXX: hence unblocking the pipeline? */
+  gst_pad_remove_probe (srcpad, GST_PAD_PROBE_INFO_ID (info));
+
+  gst_bin_add (GST_BIN (server_ctx->pipeline), sinkbin);
+
+  /* t. ! queue ! appsink */
+  sinkpad = gst_element_get_static_pad (sinkbin, "qsink");
+  gst_pad_link (srcpad, sinkpad);
+
+  /* Send the WebM stream header through the pipeline first */
+  caps = gst_pad_get_current_caps (srcpad);
+  /* TODO: cache keyframes */
+  buffer = stp_get_streamheader_from_caps (caps);
+  if (!buffer)
+    goto err;
+  gst_pad_push (srcpad, buffer);
+
+  ret = gst_element_get_state (server_ctx->pipeline, &state, NULL, 1);
+  g_print ("Linked pads, removed probe. State was %s:%s.\n",
+           gst_element_state_change_return_get_name (ret),
+           gst_element_state_get_name (state));
+ 
+  if (gst_element_set_state (server_ctx->pipeline, GST_STATE_PLAYING) ==
+      GST_STATE_CHANGE_FAILURE) {
+    g_critical ("Unable to set pipeline back to PLAYING\n");
+    goto err;
+  } else {
+    g_print ("pipeline set to PLAYING successfully\n");
+  }
+
+out:
+  gst_caps_unref (caps);
+  gst_object_unref (sinkpad);
+  gst_object_unref (sinkbin);
+  return GST_PAD_PROBE_OK;
+err:
+  soup_message_set_status (ctx->msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+  stp_cleanup_transcode_client_ctx (ctx);
+  /* FIXME: PROBE_OK for errors too? What to do here? */
+  goto out;
+}
+
+static GstFlowReturn
+write_first_client_chunk_cb (GstElement         *appsink,
+                             TranscodeClientCtx *ctx)
+{
+  GstMapInfo info;
+  GstSample *sample;
+  GstBuffer *buffer;
+  GstMemory *memory;
+
+  g_assert (ctx->first_sample_handler_id);
+
+  g_signal_handler_disconnect (appsink, ctx->first_sample_handler_id);
+  ctx->first_sample_handler_id = 0;
+
+  /* XXX: If no samples are available, this will block till a sample is available */
+  g_signal_emit_by_name (appsink, "pull-sample", &sample);
+  buffer = gst_sample_get_buffer (sample);
+  memory = gst_buffer_get_all_memory (buffer); /* copy */
+  gst_memory_map (memory, &info, GST_MAP_READ);
+
+  soup_message_body_append (ctx->msg->response_body, SOUP_MEMORY_COPY,
+                            info.data, info.size); /* copy */
+
+  gst_memory_unmap (memory, &info);
+  gst_memory_unref (memory);
+  gst_sample_unref (sample);
+
+  soup_server_unpause_message (server, ctx->msg);
+  g_print (".");
+  return GST_FLOW_OK;
+}
+
+static void
+write_next_client_chunk_cb (SoupMessage        *msg,
+                            TranscodeClientCtx *ctx)
+{
+  GstMapInfo info;
+  GstSample *sample;
+  GstBuffer *buffer;
+  GstMemory *memory;
+  gboolean eos;
+
+  g_signal_emit_by_name (ctx->appsink, "pull-sample", &sample);
+  if (!sample) {
+    g_print ("Null sample, ending stream\n");
+    g_object_get (ctx->appsink, "eos", &eos, NULL);
+    if (eos)
+      soup_message_set_status (msg, SOUP_STATUS_OK);
+    else
+      soup_message_set_status (msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+    soup_message_body_complete (msg->response_body);
+    return;
+  }
+
+  buffer = gst_sample_get_buffer (sample);
+  memory = gst_buffer_get_all_memory (buffer); /* copy */
+  gst_memory_map (memory, &info, GST_MAP_READ);
+
+  soup_message_body_append (msg->response_body, SOUP_MEMORY_COPY,
+                            info.data, info.size); /* copy */
+
+  gst_memory_unmap (memory, &info);
+  gst_memory_unref (memory);
+  gst_sample_unref (sample);
+
+  soup_server_unpause_message (server, msg);
+  g_print (".");
+  return;
+}
+
+static void
+client_finished_cb (SoupMessage        *msg,
+                    TranscodeClientCtx *ctx)
+{
+  stp_cleanup_transcode_client_ctx (ctx);
+}
+
+static void
+handle_request_cb (SoupServer        *server,
+                   SoupMessage       *msg,
+                   const char        *path,
+                   GHashTable        *query,
+                   SoupClientContext *client,
+                   GHashTable        *ctx_table)
+{
+  TranscodeServerCtx *server_ctx;
+  
+  if (!msg)
+    return;
+
+  g_print ("Handling %s request on path %s\n", msg->method, path);
+
+  if (msg->method == SOUP_METHOD_PUT ||
+      msg->method == SOUP_METHOD_POST)
+    goto PUT;
+  else if (msg->method == SOUP_METHOD_GET)
+    goto GET;
+  else
+    /* We return NOT_IMPLEMENTED for this
+     * when we get the request headers */
+    g_assert_not_reached ();
+
+PUT:
+  {
+    /* The PUT request has finished streaming. Cleanup
+     * will be done on EOS/ERROR in the gst pipeline. */
+    server_ctx = get_server_ctx_from_msg (msg, ctx_table);
+    server_ctx->request_finished = TRUE;
+    soup_message_set_status (server_ctx->msg, SOUP_STATUS_OK);
+    return;
+  }
+
+GET:
+  {
+    /* A GET request was received. We connect from the pipeline to the
+     * client requesting the stream and start writing the response. */
+    GstElement *bin, *tee, *q2, *appsink;
+    GstPadTemplate *template;
+    GstPad *srcpad, *sinkpadq2;
+    TranscodeClientCtx *client_ctx;
+    GstStateChangeReturn state;
+ 
+    server_ctx = get_server_ctx_from_msg (msg, ctx_table);
+    if (server_ctx == NULL ||
+        server_ctx->pipeline == NULL) {
+      /* There's no request streaming on this path (yet) */
+      soup_message_set_status (msg, SOUP_STATUS_NOT_FOUND);
+      return;
+    }
+
+    /* Check if the pipeline successfully started PLAYING */
+    state = gst_element_get_state (server_ctx->pipeline,
+                                   NULL, NULL, 100*GST_MSECOND);
+    switch (state) {
+      case GST_STATE_CHANGE_SUCCESS:
+        break;
+      case GST_STATE_CHANGE_FAILURE:
+        /* PUT stream should've  */
+        g_critical ("GET request, but state change failure?");
+      case GST_STATE_CHANGE_ASYNC:
+        soup_message_set_status (msg, SOUP_STATUS_SERVICE_UNAVAILABLE);
+        return;
+      default:
+        g_assert_not_reached ();
+    }
+ 
+    /* Connect appsink to tee, and start streaming */
+    client_ctx = g_new0 (TranscodeClientCtx, 1);
+    client_ctx->msg = msg;
+    client_ctx->server_ctx = server_ctx;
+ 
+    bin = gst_bin_new (NULL);
+
+    /* queue ! appsink */
+    q2 = gst_element_factory_make ("queue", "q2");
+    appsink = gst_element_factory_make ("appsink", NULL);
+    g_object_set (appsink, "drop", FALSE,
+                  "emit-signals", TRUE,
+                  "max-buffers", 0, NULL);
+    gst_bin_add_many (GST_BIN (bin), q2, appsink, NULL);
+    gst_element_link (q2, appsink);
+
+    sinkpadq2 = gst_element_get_static_pad (q2, "sink");
+    gst_element_add_pad (bin, gst_ghost_pad_new ("qsink", sinkpadq2));
+    gst_object_unref (sinkpadq2);
+
+    /* Set to PAUSED so the bin can pre-roll */
+    if (gst_element_set_state (bin, GST_STATE_PAUSED) ==
+        GST_STATE_CHANGE_FAILURE) {
+      g_critical ("Unable to set appsink to PAUSED\n");
+      soup_message_set_status (client_ctx->msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+      stp_cleanup_transcode_client_ctx (client_ctx);
+    } else {
+      g_print ("sinkbin set to PAUSED successfully\n");
+    }
+
+    /* Request a new src pad from the tee in the pipeline */
+    tee = gst_bin_get_by_name (GST_BIN (server_ctx->pipeline), "tee");
+    template = gst_pad_template_new ("teesrc", GST_PAD_SRC,
+                                     GST_PAD_REQUEST, GST_CAPS_ANY);
+    srcpad = gst_element_request_pad (tee, template, NULL, GST_CAPS_ANY);
+    
+    gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_BLOCK,
+                       (GstPadProbeCallback) tee_src_pad_blocked_cb,
+                       client_ctx, NULL);
+    gst_object_unref (srcpad);
+
+    gst_object_ref (appsink);
+    client_ctx->appsink = appsink;
+
+    client_ctx->first_sample_handler_id =
+      g_signal_connect (appsink, "new-sample",
+                        G_CALLBACK (write_first_client_chunk_cb), client_ctx);
+    g_signal_connect (client_ctx->msg, "wrote-chunk",
+                      G_CALLBACK (write_next_client_chunk_cb), client_ctx);
+    g_signal_connect (client_ctx->msg, "finished",
+                      G_CALLBACK (client_finished_cb), client_ctx);
+
+    server_ctx->clients = g_list_prepend (server_ctx->clients, client_ctx);
+
+    /* Since we set the response as chunked, the server will 
+     * automatically pause the message for us, and the response 
+     * won't be completed till we call soup_message_body_complete() */
+    soup_message_set_status (msg, SOUP_STATUS_OK);
+  }
+}
+
+static void
+got_request_body_chunk (SoupMessage        *msg,
+                        SoupBuffer         *chunk,
+                        TranscodeServerCtx *ctx)
+{
+  GstBuffer *buffer;
+  GstFlowReturn ret;
+
+  /* XXX: Unfortunately, there doesn't seem to be a way to cancel the
+   * reception of further chunks when we've prematurely set the response
+   * (say due to an error), so we check if the status code is set and skip
+   * all further chunks */
+  if (msg->status_code != 0)
+    return;
+
+  /* We need to update this every chunk */
+  ctx->msg = msg;
+
+  /* Actually just a ref */
+  buffer = stp_get_gst_buffer (chunk);
+  g_signal_emit_by_name (ctx->appsrc, "push-buffer", buffer, &ret);
+  gst_buffer_unref (buffer);
+  if (ret != GST_FLOW_OK) {
+    g_critical ("Unable to push buffer\n");
+    soup_message_set_status (msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+    stp_cleanup_transcode_server_ctx (ctx);
+    return;
+  }
+
+  /* We can only set the pipeline to playing when 
+   * the first chunk has been received */
+  if (!ctx->pipeline_is_playing) {
+    if (gst_element_set_state (ctx->pipeline, GST_STATE_PLAYING) ==
+        GST_STATE_CHANGE_FAILURE) {
+      g_critical ("Unable to set pipeline to PLAYING\n");
+      soup_message_set_status (msg, SOUP_STATUS_INTERNAL_SERVER_ERROR);
+      stp_cleanup_transcode_server_ctx (ctx);
+    } else {
+      g_print ("Set pipeline to PLAYING\n");
+      ctx->pipeline_is_playing = TRUE;
+    }
+  }
+}
+
+static void
+got_request_headers (SoupMessage *msg,
+                     GHashTable  *ctx_table)
+{
+  TranscodeServerCtx *ctx;
+  SoupURI *uri;
+
+#ifdef DEBUG
+  SoupMessageHeadersIter iter;
+  const char *name, *value;
+  soup_message_headers_iter_init (&iter, msg->request_headers);
+  while (soup_message_headers_iter_next (&iter, &name, &value))
+    g_print ("%s: %s\n", name, value);
+#endif
+
+  g_object_get (msg, "uri", &uri, NULL);
+  g_print ("%s on uri %s\n", msg->method, soup_uri_get_path (uri));
+
+  if (msg->method == SOUP_METHOD_PUT ||
+      msg->method == SOUP_METHOD_POST)
+    goto PUT;
+  else if (msg->method == SOUP_METHOD_GET)
+    goto GET;
+  else {
+    /* Nothing to do; server doesn't implement this method */
+    soup_message_set_status (msg, SOUP_STATUS_NOT_IMPLEMENTED);
+    goto out;
+  }
+
+PUT: {
+    if (g_hash_table_contains (ctx_table, soup_uri_get_path (uri))) {
+      /* There's already a request streaming on this path */
+      g_print ("Recv duplicate request on the same URI: %s\n",
+               soup_uri_get_path (uri));
+      soup_message_set_status (msg, SOUP_STATUS_CONFLICT);
+      goto out;
+    }
+
+    ctx = g_new0 (TranscodeServerCtx, 1);
+    ctx->msg = msg;
+    ctx->parent_ctx_table = ctx_table;
+    ctx->path = g_strdup (soup_uri_get_path (uri));
+    g_hash_table_insert (ctx_table, ctx->path, ctx);
+
+    /* We only do this when we receive the first chunk */
+    if (!ctx->pipeline) {
+      ctx->pipeline = gst_pipeline_new ("pipe");
+      /* The chunked request is copied into this stream 
+       * for consumption by the gst pipeline */
+#ifdef DEBUG
+      stp_play_from_msg (ctx);
+#else
+      stp_encode_from_msg (ctx);
+#endif
+      g_signal_connect (ctx->appsrc, "need-data",
+                        G_CALLBACK (need_data_unpause_message), ctx);
+      g_signal_connect (ctx->appsrc, "enough-data",
+                        G_CALLBACK (enough_data_pause_message), ctx);
+    }
+    
+    /* The request body isn't fixed length, so tell libsoup to
+     * not collect chunks for forming a complete request body */
+    soup_message_body_set_accumulate (msg->request_body, FALSE);
+
+    g_signal_connect (msg, "got-chunk",
+                      G_CALLBACK (got_request_body_chunk), ctx);
+    goto out;
+   }
+
+GET: {
+    if (!g_hash_table_contains (ctx_table, soup_uri_get_path (uri))) {
+      g_print ("No stream on URI: %s\n", soup_uri_get_path (uri));
+      soup_message_set_status (msg, SOUP_STATUS_NOT_FOUND);
+      goto out;
+    }
+
+    /* Our response will be chunked, and not with a fixed Content-Length */
+    soup_message_headers_set_encoding (msg->response_headers, SOUP_ENCODING_CHUNKED);
+    /* The request body isn't fixed length, so tell libsoup to
+     * not collect chunks for forming a complete request body */
+    soup_message_body_set_accumulate (msg->response_body, FALSE);
+   }
+
+out:
+  soup_uri_free (uri);
+  return;
+}
+
+static void
+request_started_cb (SoupServer        *server,
+                    SoupMessage       *msg,
+                    SoupClientContext *client,
+                    GHashTable        *ctx_table)
+{
+  g_print ("New %s request started\n", msg->method);
+
+  /* SoupMessage is useful only once we have headers,
+   * so we do all initialization there */
+  g_signal_connect (msg, "got-headers",
+                    G_CALLBACK (got_request_headers), ctx_table);
+}
+
+static void
+request_read_cb (SoupServer        *server,
+                 SoupMessage       *msg,
+                 SoupClientContext *client,
+                 GHashTable        *ctx_table)
+{
+  if (msg->method == SOUP_METHOD_PUT ||
+      msg->method == SOUP_METHOD_POST)
+    goto PUT;
+  else if (msg->method == SOUP_METHOD_GET)
+    goto GET;
+  else
+    /* We return NOT_IMPLEMENTED for this
+     * when we get the request headers */
+    g_assert_not_reached ();
+
+PUT:
+  {
+    gboolean ret;
+    TranscodeServerCtx *ctx;
+
+    ctx = get_server_ctx_from_msg (msg, ctx_table);
+    g_return_if_fail (ctx);
+
+    /* Incoming stream has ended */
+    g_signal_emit_by_name (ctx->appsrc, "end-of-stream", &ret);
+    if (!ret)
+      g_printerr ("Unable to emit end-of-stream\n");
+  }
+
+GET:
+  {
+    /* Nothing to do */
+  }
+
+  g_print ("%s request read successfully\n", msg->method);
+}
+
+static void
+request_finished_cb (SoupServer        *server,
+                     SoupMessage       *msg,
+                     SoupClientContext *client,
+                     GHashTable        *ctx_table)
+{
+  g_print ("%s request ended\n", msg->method);
+}
+
+static void
+request_aborted_cb (SoupServer        *server,
+                    SoupMessage       *msg,
+                    SoupClientContext *client,
+                    GHashTable        *ctx_table)
+{
+  g_print ("%s request aborted!\n", msg->method);
+  if (msg->method == SOUP_METHOD_PUT ||
+      msg->method == SOUP_METHOD_POST)
+    goto PUT;
+  else if (msg->method == SOUP_METHOD_GET)
+    goto GET;
+  else
+    /* We return NOT_IMPLEMENTED for this
+     * when we get the request headers */
+    g_assert_not_reached ();
+
+PUT:
+  {
+    /* FIXME: Is there anything to do here? */
+  }
+
+GET:
+  {
+    /* We do all cleanup in the "finished" msg signal, nothing to do here */
+  }
+}
+
+int
+main (int   argc,
+      char *argv[])
+{
+  GHashTable *ctx_table;
+
+  gst_init (&argc, &argv);
+
+  /* Keys are paths, and values are TranscodeServerCtxs */
+  ctx_table = g_hash_table_new_full (g_str_hash, g_str_equal,
+                                     (GDestroyNotify)g_free,
+                                     (GDestroyNotify)stp_cleanup_transcode_server_ctx);
+  server = soup_server_new (SOUP_SERVER_SERVER_HEADER,
+                            "soup-transcode-proxy ",
+                            SOUP_SERVER_PORT, 8000, NULL);
+  soup_server_add_handler (server, NULL,
+                           (SoupServerCallback)handle_request_cb,
+                           ctx_table, NULL);
+  g_signal_connect (server, "request-started",
+                    G_CALLBACK (request_started_cb), ctx_table);
+  g_signal_connect (server, "request-read",
+                    G_CALLBACK (request_read_cb), ctx_table);
+  g_signal_connect (server, "request-finished",
+                    G_CALLBACK (request_finished_cb), ctx_table);
+  g_signal_connect (server, "request-aborted",
+                    G_CALLBACK (request_aborted_cb), ctx_table);
+
+  soup_server_run (server);
+  return 0;
+}