Attempt to migrate audio hacks to FarStream (non-functional so far)

diff --git a/chat.c b/chat.c
index e7430dd38085f2fd6fe08ecce45bb6d7c510cfc2..d5ce2ec0fab6249a2d127f46b2bdd2541bb1d346 100644 (file)
--- a/chat.c
+++ b/chat.c
@@ -26,6 +26,7 @@
 #include <debug.h>
 #include <media.h>
 #include <mediamanager.h>
+#include <media-gst.h>
 
 #include "chime.h"
 #include "chime-connection-private.h"
@@ -239,11 +240,12 @@ static void on_audio_state(ChimeCall *call, ChimeAudioState audio_state, struct
        purple_debug(PURPLE_DEBUG_INFO, "chime", "Audio state %d\n", audio_state);
 
        if (audio_state == CHIME_AUDIO_STATE_AUDIO_MUTED && chat->audio_outpipeline) {
-               gst_element_set_state(chat->audio_outpipeline, GST_STATE_PAUSED);
+               //gst_element_set_state(chat->audio_outpipeline, GST_STATE_PAUSED);
        } else if (audio_state == CHIME_AUDIO_STATE_AUDIO && chat->audio_outpipeline) {
-               gst_element_set_state(chat->audio_outpipeline, GST_STATE_PLAYING);
+               //gst_element_set_state(chat->audio_outpipeline, GST_STATE_PLAYING);
        } else if (audio_state == CHIME_AUDIO_STATE_AUDIO && !chat->audio_inpipeline) {
                // GStreamer - server-to-speakers
+#if 0          
                chat->audio_inpipeline = gst_pipeline_new("dirt-pipeline");
 
                GstAppSrc *src = GST_APP_SRC(gst_element_factory_make("appsrc", "appsrc"));
@@ -287,6 +289,57 @@ static void on_audio_state(ChimeCall *call, ChimeAudioState audio_state, struct
                chime_call_install_gst_app_callbacks(chat->call, src, GST_APP_SINK(appsink));
                gst_element_set_state(chat->audio_inpipeline, GST_STATE_PLAYING);
                gst_element_set_state(chat->audio_outpipeline, GST_STATE_PLAYING);
+#else
+               PurpleMediaManager *mgr = purple_media_manager_get();
+                       const gchar *name = chime_call_get_alert_body(chat->call);
+       chat->media = purple_media_manager_create_media(purple_media_manager_get(),
+                                                              chat->conv->account,
+                                                              "fsrawconference",
+                                                              name,
+                                                              TRUE);
+       if (chat->media) {
+               gboolean r = purple_media_add_stream(chat->media, "chime", name,
+                                                    PURPLE_MEDIA_AUDIO, TRUE,
+                                                    "app", 0, NULL);
+               gchar *srcname = g_strdup_printf("chime_src_%p", call);
+               gchar *sinkname = g_strdup_printf("chime_sink_%p", call);
+               gchar *srcpipe = g_strdup_printf("appsrc name=%s ! opusdec", srcname);
+               gchar *sinkpipe = g_strdup_printf("opusenc channels=1 bitrate=16000 bitrate-type=vbr ! appsink name=%s", sinkname);
+
+               PurpleMediaCandidate *cand =
+                       purple_media_candidate_new(NULL, 1,
+                                                  PURPLE_MEDIA_CANDIDATE_TYPE_HOST,
+                                                  PURPLE_MEDIA_NETWORK_PROTOCOL_UDP,
+                                                  sinkpipe, 0);
+               g_object_set(cand, "username", srcpipe, NULL);
+               g_free(sinkpipe);
+               g_free(srcpipe);
+
+               GList *cands = g_list_append (NULL, cand);
+               GList *codecs = g_list_append(NULL,
+                                              purple_media_codec_new(1, "audio/x-raw, format=(string)S16LE, rate=16000, layout=(string)interleaved, channels=(int)1", PURPLE_MEDIA_AUDIO, 0));
+               purple_media_add_remote_candidates(chat->media, "chime", name, cands);
+               purple_media_set_remote_codecs(chat->media, "chime", name, codecs);
+
+               GstElement *pipeline = purple_media_manager_get_pipeline(mgr);
+               GstElement *appsrc = gst_bin_get_by_name(GST_BIN(pipeline), srcname);
+               GstElement *appsink = gst_bin_get_by_name(GST_BIN(pipeline), sinkname);
+               g_free(srcname);
+               g_free(sinkname);
+               printf("Got src %p sink %p\n", appsrc, appsink);
+               gst_app_src_set_size(GST_APP_SRC(appsrc), -1);
+               gst_app_src_set_max_bytes(GST_APP_SRC(appsrc), 100);
+               gst_base_src_set_live(GST_BASE_SRC(appsrc), TRUE);
+               gst_app_src_set_stream_type(GST_APP_SRC(appsrc), GST_APP_STREAM_TYPE_STREAM);
+
+               chime_call_install_gst_app_callbacks(chat->call, GST_APP_SRC(appsrc), GST_APP_SINK(appsink));
+
+               purple_media_stream_info(chat->media, PURPLE_MEDIA_INFO_ACCEPT, "chime", name, FALSE);
+
+               GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(purple_media_manager_get_pipeline(mgr)), GST_DEBUG_GRAPH_SHOW_ALL, "chime conference graph");
+       }
+
+#endif
        }
 
 }
@@ -401,43 +454,6 @@ static void on_chat_name(ChimeObject *obj, GParamSpec *ignored, struct chime_cha
                purple_conversation_set_name(chat->conv, name);
 }
 
-#if 0
-static void audio_joined(GObject *source, GAsyncResult *result, gpointer _chat)
-{
-       ChimeConnection *cxn = CHIME_CONNECTION(source);
-       struct chime_chat *chat = _chat;
-
-       chat->audio = chime_connection_join_call_audio_finish(cxn, result, NULL);
-       if (!chat->audio)
-               return;
-
-#if 0 /* FIXME make this work... */
-       const gchar *name = chime_call_get_alert_body(chat->call);
-       chat->media = purple_media_manager_create_media(purple_media_manager_get(),
-                                                              chat->conv->account,
-                                                              "fsrawconference",
-                                                              name,
-                                                              TRUE);
-       printf("media for %s %p\n", name, chat->media);
-       if (chat->media) {
-               gboolean r = purple_media_add_stream(chat->media, "chime", name,
-                                                    PURPLE_MEDIA_AUDIO, TRUE,
-                                                    "app", 0, NULL);
-               printf("Add stream %s\n", r ? "succeeded" : "failed");
-               GList *cands = g_list_append (NULL,
-                                             purple_media_candidate_new(NULL, 1,
-                                                                        PURPLE_MEDIA_CANDIDATE_TYPE_HOST, PURPLE_MEDIA_NETWORK_PROTOCOL_UDP, "/tmp/src1", 0));
-
-               GList *codecs = g_list_append(NULL,
-                                              purple_media_codec_new(1, "audio/x-raw, format=(string)S16LE, layout=(string)interleaved, rate=(int)44100, channels=(int)1", PURPLE_MEDIA_AUDIO, 0));
-               purple_media_add_remote_candidates(chat->media, "chime", name, cands);
-               purple_media_set_send_codec(chat->media, "chime", codecs->data);
-               purple_media_set_remote_codecs(chat->media, "chime", name, codecs);
-               purple_media_stream_info(chat->media, PURPLE_MEDIA_INFO_ACCEPT, "chime", name, FALSE);
-       }
-#endif
-}
-#endif
 struct chime_chat *do_join_chat(PurpleConnection *conn, ChimeConnection *cxn, ChimeObject *obj, JsonNode *first_msg, ChimeMeeting *meeting)
 {
        if (!obj)
@@ -456,11 +472,6 @@ struct chime_chat *do_join_chat(PurpleConnection *conn, ChimeConnection *cxn, Ch
                if (chat->call) {
                        g_signal_connect(chat->call, "audio-state", G_CALLBACK(on_audio_state), chat);
                        g_signal_connect(chat->call, "participants-changed", G_CALLBACK(on_call_participants), chat);
-#if 0
-                       /* Don't try this at home, kids! (yet) */
-                       if (getenv("CHIME_AUDIO"))
-                               chime_connection_join_call_audio_async(cxn, chat->call, NULL, audio_joined, chat);
-#endif
                }
        }