Farstream kind of working now...

diff --git a/chat.c b/chat.c
index d5ce2ec0fab6249a2d127f46b2bdd2541bb1d346..318a5c5f62cd586672b60801a0fa64ce839c75da 100644 (file)
--- a/chat.c
+++ b/chat.c
@@ -235,112 +235,70 @@ static void participants_closed_cb(gpointer _chat)
        g_signal_handlers_disconnect_matched(chat->call, G_SIGNAL_MATCH_FUNC|G_SIGNAL_MATCH_DATA,
                                             0, 0, NULL, G_CALLBACK(on_call_participants), chat);
 }
-static void on_audio_state(ChimeCall *call, ChimeAudioState audio_state, struct chime_chat *chat)
+
+static void call_media_changed(PurpleMedia *media, PurpleMediaState state, const gchar *id, const gchar *participant, struct chime_chat *chat)
 {
-       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);
-       } else if (audio_state == CHIME_AUDIO_STATE_AUDIO && chat->audio_outpipeline) {
-               //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"));
-               GstCaps *audio_caps;
-               audio_caps = gst_caps_from_string("audio/x-opus,channel-mapping-family=0");
-               g_object_set (src, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
-               gst_app_src_set_size(src, -1);
-               gst_app_src_set_max_bytes(src, 100);
-               gst_base_src_set_live(GST_BASE_SRC(src), TRUE);
-               gst_app_src_set_stream_type(src, GST_APP_STREAM_TYPE_STREAM);
-
-               GstElement *opusdec = gst_element_factory_make("opusdec", "opusdec");
-               GstElement *convert = gst_element_factory_make("audioconvert", "audioconvert");
-               GstElement *resample = gst_element_factory_make("audioresample", "audioresample");
-               GstElement *sink = gst_element_factory_make("autoaudiosink", "autoaudiosink");
-               gst_bin_add_many(GST_BIN(chat->audio_inpipeline), GST_ELEMENT(src), opusdec, convert, resample, sink, NULL);
-               if(!gst_element_link_many(GST_ELEMENT(src), opusdec, convert, resample, sink, NULL)) {
-                       printf("Failed to link incoming pipeline\n");
-               }
+       if (state == PURPLE_MEDIA_STATE_END && !id && !participant) {
+               chat->media = NULL;
+               chime_call_set_mute(chat->call, TRUE);
+       }
+}
 
-               // GStreamer - mic-to-server
-               chat->audio_outpipeline = gst_pipeline_new("upstream-audio");
-               GstElement *mic = gst_element_factory_make("autoaudiosrc", "autoaudiosrc");
-               convert = gst_element_factory_make("audioconvert", "audioconvert");
-               g_object_set(convert, "caps", gst_caps_from_string("audio/x-raw,format=S16,channels=1"), NULL);
-               resample = gst_element_factory_make("audioresample", "audioresample");
-               g_object_set(resample, "caps", gst_caps_from_string("audio/x-raw"), NULL);
-               GstElement *opusenc = gst_element_factory_make("opusenc", "opusenc");
-               g_object_set(opusenc, "caps", gst_caps_from_string("audio/x-raw,format=S16,channels=1"), NULL);
-               g_object_set(opusenc,
-                            "bitrate", 16000,
-                            "bitrate-type", "vbr",
-                            NULL);
-               GstElement *appsink = gst_element_factory_make("appsink", "appsink");
-               g_object_set(appsink, "caps", gst_caps_from_string("audio/x-opus,channels=1,channel-mapping-family=0"), NULL);
-               gst_bin_add_many(GST_BIN(chat->audio_outpipeline), mic, convert, resample, opusenc, appsink, NULL);
-               if(!gst_element_link_many(mic, convert, resample, opusenc, appsink, NULL)) {
-                       printf("Failed to link upstream pipeline\n");
-               }
+static void on_audio_state(ChimeCall *call, ChimeAudioState audio_state, struct chime_chat *chat)
+{
+       purple_debug(PURPLE_DEBUG_INFO, "chime", "Audio state %d\n", audio_state);
 
-               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
+       if (audio_state == CHIME_AUDIO_STATE_AUDIO && !chat->media) {
                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");
+               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 format=time caps=audio/x-opus,channel-mapping-family=0 ! opusdec ! audioconvert ! audioresample ", srcname);
+                       gchar *sinkpipe = g_strdup_printf("opusenc bitrate=16000 bitrate-type=vbr ! appsink name=%s async=false", 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=(int)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_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));
+                       g_signal_connect(chat->media, "state-changed", G_CALLBACK(call_media_changed), chat);
+
+                       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
-       }
 
 }
 

diff --git a/chime-call-audio.c b/chime-call-audio.c
index a89d833574626bae42d168eeca7076bf52f59fcc..81b26d3f89a4d3d695c4209c3fa62614e7143414 100644 (file)
--- a/chime-call-audio.c
+++ b/chime-call-audio.c
@@ -61,12 +61,19 @@ static gboolean audio_receive_rt_msg(ChimeCallAudio *audio, gconstpointer pkt, g
                        audio->echo_server_time = TRUE;
                }
                if (msg->audio->has_audio) {
-                       if (audio->audio_src && audio->appsrc_need_data) {
+                       GstClock *clk;
+                       if (audio->audio_src && audio->appsrc_need_data &&
+                           /* It'll have a clock if it's PLAYING. */
+                           (clk = gst_element_get_clock(GST_ELEMENT(audio->audio_src)))) {
                                GstBuffer *buffer = gst_buffer_new_allocate(NULL, msg->audio->audio.len, NULL);
                                gst_buffer_fill(buffer, 0, msg->audio->audio.data, msg->audio->audio.len);
-                               if (msg->audio->sample_time)
-                                       buffer->dts = msg->audio->sample_time;
 
+                               if (!audio->audio_src_timebase) {
+                                       audio->base_src_sample_time = msg->audio->sample_time;
+                                       audio->audio_src_timebase = gst_clock_get_time(clk) - gst_element_get_base_time(GST_ELEMENT(audio->audio_src));
+                                       printf("Timebase %ld sample base %d\n", audio->audio_src_timebase, msg->audio->sample_time);
+                               }
+                               buffer->dts = audio->audio_src_timebase + (uint64_t)NS_PER_SAMPLE * (uint64_t)(msg->audio->sample_time - audio->base_src_sample_time);
                                gst_app_src_push_buffer(GST_APP_SRC(audio->audio_src), buffer);
                        }
                }
@@ -141,7 +148,6 @@ static void do_send_rt_packet(ChimeCallAudio *audio, GstBuffer *buffer)
                pts = GST_BUFFER_PTS(buffer);
                dur = GST_BUFFER_DURATION(buffer);
 
-#define NS_PER_SAMPLE (1000000000 / 16000)
                nr_samples = GST_BUFFER_DURATION(buffer) / NS_PER_SAMPLE;
                printf("buf dts %ld pts %ld dur %ld samples %d\n", dts, pts, dur, nr_samples);
 
@@ -493,6 +499,7 @@ static void chime_appsrc_destroy(gpointer _audio)
 {
        ChimeCallAudio *audio = _audio;
 
+       audio->audio_src_timebase = 0;
        audio->audio_src = NULL;
 }
 

diff --git a/chime-call-audio.h b/chime-call-audio.h
index e4eeb1ad589ee9b637eaa21ba9a119b61e826abc..f8dd2d2c41f9973cb0301548285f4ff5c3966c0c 100644 (file)
--- a/chime-call-audio.h
+++ b/chime-call-audio.h
@@ -28,6 +28,8 @@
 #include <gst/app/gstappsrc.h>
 #include <gst/app/gstappsink.h>
 
+#define NS_PER_SAMPLE (1000000000 / 16000)
+
 struct _ChimeCallAudio {
        ChimeCall *call;
        ChimeAudioState state;
@@ -45,6 +47,9 @@ struct _ChimeCallAudio {
        GstClockTime next_dts;
        gint64 last_send_local_time;
        GstAppSrc *audio_src;
+       GstClockTime audio_src_timebase;
+       GstClockTime audio_src_rate_num, audio_src_rate_denom;
+       uint32_t base_src_sample_time;
        gboolean appsrc_need_data;
 
        GMutex rt_lock;

diff --git a/fs-app-transmitter/fs-app-stream-transmitter.c b/fs-app-transmitter/fs-app-stream-transmitter.c
index 4d2ef2bbaec543fcd25648f8580e31e60ff4904c..755a7cf1334ecb382fba8b0df45e66ce93f1fd8d 100644 (file)
--- a/fs-app-transmitter/fs-app-stream-transmitter.c
+++ b/fs-app-transmitter/fs-app-stream-transmitter.c
@@ -391,7 +391,7 @@ fs_app_stream_transmitter_add_sink (FsAppStreamTransmitter *self,
   if (self->priv->app_sink[candidate->component_id] == NULL)
     return FALSE;
 
-#if 1
+#if 0
   if (candidate->component_id == 1) {
     fs_app_transmitter_sink_set_sending (self->priv->transmitter,
         self->priv->app_sink[candidate->component_id], self->priv->sending);