Switch to using RTP for audio

We get a jitterbuffer for free this way, which makes it a lot more usable.

It also means I can drop a lot of the timestamp mangling crap and just
transpose the protocol's sample counts into RTP headers... I think. The
idea is to drop the 'opusrtppay' part and feed RTP directly into the
appsrc, and the converse for the sink.

It does require farstream 0.2.8 *and* to remove libgstsrtp.so for now.

diff --git a/chat.c b/chat.c
index 2f19ea1a8589d2707a13eb6aeff94596fa33b714..32089e780a44f90085ae509b9dd41abfa58fac08 100644 (file)
--- a/chat.c
+++ b/chat.c
@@ -238,7 +238,6 @@ static void participants_closed_cb(gpointer _chat)
 
 static void call_stream_info(PurpleMedia *media, PurpleMediaInfoType type, gchar *id, const gchar *participant, gboolean local, struct chime_chat *chat)
 {
-       printf("%s %d\n", __func__, type);
        if (type == PURPLE_MEDIA_INFO_MUTE) {
                chime_call_set_local_mute(chat->call, TRUE);
        } else if (type == PURPLE_MEDIA_INFO_UNMUTE) {
@@ -248,7 +247,6 @@ static void call_stream_info(PurpleMedia *media, PurpleMediaInfoType type, gchar
 
 static void call_media_changed(PurpleMedia *media, PurpleMediaState state, const gchar *id, const gchar *participant, struct chime_chat *chat)
 {
-
        if (state == PURPLE_MEDIA_STATE_END && !id && !participant) {
                if (chat->media) {
                        chat->media = NULL;
@@ -271,11 +269,11 @@ static void on_audio_state(ChimeCall *call, ChimeAudioState audio_state, struct
                PurpleMediaManager *mgr = purple_media_manager_get();
                chat->media = purple_media_manager_create_media(purple_media_manager_get(),
                                                                chat->conv->account,
-                                                               "fsrawconference",
+                                                               "fsrtpconference",
                                                                name,
                                                                TRUE);
                if (chat->media) {
-                       const gchar *caps = "audio/x-raw,format=(string)S16LE,layout=(string)interleaved,channels=(int)1";
+                       const gchar *caps = "application/x-rtp,media=(string)audio,payload=(int)96,encoding-name=(string)OPUS";
 
                        gboolean r = purple_media_add_stream(chat->media, "chime", name,
                                                             PURPLE_MEDIA_AUDIO, TRUE,
@@ -283,9 +281,8 @@ static void on_audio_state(ChimeCall *call, ChimeAudioState audio_state, struct
                        gchar *srcname = g_strdup_printf("chime_src_%p", call);
                        gchar *sinkname = g_strdup_printf("chime_sink_%p", call);
                        /* Without the capsfilter (even though there's another capsfilter almost immediately after it in the pipeline, it doesn't work */
-                       gchar *srcpipe = g_strdup_printf("appsrc name=%s format=time caps=audio/x-opus,channel-mapping-family=0 ! opusdec ! capsfilter caps=%s",
-                                                        srcname, caps);
-                       gchar *sinkpipe = g_strdup_printf("opusenc bitrate=16000 bitrate-type=vbr ! appsink name=%s async=false", sinkname);
+                       gchar *srcpipe = g_strdup_printf("appsrc name=%s format=time caps=audio/x-opus,channel-mapping-family=0,channels=1 ! rtpopuspay ! capsfilter caps=%s", srcname, caps);
+                       gchar *sinkpipe = g_strdup_printf("rtpopusdepay ! appsink name=%s async=false", sinkname);
 
                        PurpleMediaCandidate *cand =
                                purple_media_candidate_new(NULL, 1,
@@ -298,7 +295,12 @@ static void on_audio_state(ChimeCall *call, ChimeAudioState audio_state, struct
 
                        GList *cands = g_list_append (NULL, cand);
                        GList *codecs = g_list_append(NULL,
-                                                     purple_media_codec_new(1, caps, PURPLE_MEDIA_AUDIO, 0));
+                                                     purple_media_codec_new(96, "OPUS", PURPLE_MEDIA_AUDIO, 0));
+                       //                      g_object_set(codecs->data, "channels", 1, NULL);
+                       //                      purple_media_codec_add_optional_parameter(codecs->data, "farstream-recv-profile", "rtpopusdepay ! opusdec");
+                       //                      purple_media_codec_add_optional_parameter(codecs->data, "farstream-send-profile", "opusenc bitrate=16000 bitrate-type=vbr ! rtpopuspay");
+                       purple_media_codec_add_optional_parameter(codecs->data, "sprop-stereo", "0");
+                       purple_media_codec_add_optional_parameter(codecs->data, "stereo", "0");
                        purple_media_add_remote_candidates(chat->media, "chime", name, cands);
                        purple_media_set_remote_codecs(chat->media, "chime", name, codecs);