static void
_check_message_received (GstBus * bus, GstMessage * message, gpointer user_data)
{
GMainLoop *main_loop = (GMainLoop *) user_data;
GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "%s: %" GST_PTR_FORMAT,
GST_MESSAGE_TYPE_NAME (message), message);
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_STATE_CHANGED:
if (GST_IS_PIPELINE (GST_MESSAGE_SRC (message))) {
GstState oldstate, newstate, pending;
gst_message_parse_state_changed (message, &oldstate, &newstate,
&pending);
switch (GST_STATE_TRANSITION (oldstate, newstate)) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
goto done;
break;
default:
break;
}
}
break;
case GST_MESSAGE_ERROR:
GST_WARNING_OBJECT (GST_MESSAGE_SRC (message), "error: %" GST_PTR_FORMAT,
message);
_check_run_main_loop_error = TRUE;
goto done;
case GST_MESSAGE_SEGMENT_DONE:
case GST_MESSAGE_EOS:
if (GST_IS_PIPELINE (GST_MESSAGE_SRC (message))) {
GST_INFO_OBJECT (GST_MESSAGE_SRC (message), "%s: %" GST_PTR_FORMAT,
GST_MESSAGE_TYPE_NAME (message), message);
goto done;
}
break;
default:
GST_WARNING_OBJECT (GST_MESSAGE_SRC (message), "unexpected messges: %s",
GST_MESSAGE_TYPE_NAME (message));
}
return;
done:
if (g_main_loop_is_running (main_loop))
g_main_loop_quit (main_loop);
}
static gboolean
gst_switchui_handle_message (GstBus * bus, GstMessage * message, gpointer data)
{
GstSwitchUI *switchui = (GstSwitchUI *) data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_EOS:
gst_switchui_handle_eos (switchui);
break;
case GST_MESSAGE_ERROR:
{
GError *error = NULL;
gchar *debug;
gst_message_parse_error (message, &error, &debug);
gst_switchui_handle_error (switchui, error, debug);
}
break;
case GST_MESSAGE_WARNING:
{
GError *error = NULL;
gchar *debug;
gst_message_parse_warning (message, &error, &debug);
gst_switchui_handle_warning (switchui, error, debug);
}
break;
case GST_MESSAGE_INFO:
{
GError *error = NULL;
gchar *debug;
gst_message_parse_info (message, &error, &debug);
gst_switchui_handle_info (switchui, error, debug);
}
break;
case GST_MESSAGE_TAG:
{
GstTagList *tag_list;
gst_message_parse_tag (message, &tag_list);
if (verbose)
g_print ("tag\n");
}
break;
case GST_MESSAGE_STATE_CHANGED:
{
GstState oldstate, newstate, pending;
gst_message_parse_state_changed (message, &oldstate, &newstate, &pending);
if (GST_ELEMENT (message->src) == switchui->pipeline) {
if (verbose)
g_print ("state change from %s to %s\n",
gst_element_state_get_name (oldstate),
gst_element_state_get_name (newstate));
switch (GST_STATE_TRANSITION (oldstate, newstate)) {
case GST_STATE_CHANGE_NULL_TO_READY:
gst_switchui_handle_null_to_ready (switchui);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_switchui_handle_ready_to_paused (switchui);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_switchui_handle_paused_to_playing (switchui);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
gst_switchui_handle_playing_to_paused (switchui);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_switchui_handle_paused_to_ready (switchui);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_switchui_handle_ready_to_null (switchui);
break;
default:
if (verbose)
g_print ("unknown state change from %s to %s\n",
gst_element_state_get_name (oldstate),
gst_element_state_get_name (newstate));
}
}
}
break;
case GST_MESSAGE_BUFFERING:
{
int percent;
gst_message_parse_buffering (message, &percent);
//g_print("buffering %d\n", percent);
if (!switchui->paused_for_buffering && percent < 100) {
g_print ("pausing for buffing\n");
switchui->paused_for_buffering = TRUE;
gst_element_set_state (switchui->pipeline, GST_STATE_PAUSED);
} else if (switchui->paused_for_buffering && percent == 100) {
g_print ("unpausing for buffing\n");
switchui->paused_for_buffering = FALSE;
gst_element_set_state (switchui->pipeline, GST_STATE_PLAYING);
}
}
break;
case GST_MESSAGE_STATE_DIRTY:
case GST_MESSAGE_CLOCK_PROVIDE:
case GST_MESSAGE_CLOCK_LOST:
case GST_MESSAGE_NEW_CLOCK:
case GST_MESSAGE_STRUCTURE_CHANGE:
case GST_MESSAGE_STREAM_STATUS:
break;
case GST_MESSAGE_STEP_DONE:
case GST_MESSAGE_APPLICATION:
case GST_MESSAGE_ELEMENT:
case GST_MESSAGE_SEGMENT_START:
case GST_MESSAGE_SEGMENT_DONE:
//case GST_MESSAGE_DURATION:
case GST_MESSAGE_LATENCY:
case GST_MESSAGE_ASYNC_START:
case GST_MESSAGE_ASYNC_DONE:
case GST_MESSAGE_REQUEST_STATE:
case GST_MESSAGE_STEP_START:
case GST_MESSAGE_QOS:
default:
if (verbose) {
g_print ("message: %s\n", GST_MESSAGE_TYPE_NAME (message));
}
break;
}
return TRUE;
}
void Pipeline::handleBusMessage(GstMessage *message)
{
switch (GST_MESSAGE_TYPE(message)) {
case GST_MESSAGE_ELEMENT:
{
// The only message we are handling here is the prepare-xwindow-id one
if (gst_structure_has_name (message->structure, "prepare-xwindow-id")) {
gst_x_overlay_set_xwindow_id(GST_X_OVERLAY(viewfinder), windowId);
}
break;
}
case GST_MESSAGE_ERROR:
{
GError *gerror = 0;
gchar *debug = 0;
gst_message_parse_error(message, &gerror, &debug);
qCritical() << "Debug" << debug << " Error " << gerror->message;
g_free(debug);
g_error_free(gerror);
break;
}
case GST_MESSAGE_WARNING:
{
GError *gerror = 0;
gchar *debug = 0;
gst_message_parse_warning(message, &gerror, &debug);
qWarning() << "Debug" << debug << " Warning " << gerror->message;
g_free(debug);
g_error_free(gerror);
break;
}
case GST_MESSAGE_INFO:
{
GError *gerror = 0;
gchar *debug = 0;
gst_message_parse_info(message, &gerror, &debug);
qDebug() << "Debug" << debug << " Info " << gerror->message;
g_free(debug);
g_error_free(gerror);
break;
}
case GST_MESSAGE_STATE_CHANGED:
{
if (GST_ELEMENT(GST_MESSAGE_SRC(message)) == camerabin) {
GstState oldstate, newstate, pending;
gst_message_parse_state_changed(message, &oldstate, &newstate, &pending);
qDebug() << Q_FUNC_INFO << gst_element_state_get_name(oldstate)
<< "->" << gst_element_state_get_name(newstate) << "=>"
<< gst_element_state_get_name(pending);
GstStateChange stateTransition =
GST_STATE_TRANSITION(oldstate, newstate);
switch (stateTransition) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
QMetaObject::invokeMethod(this, "pipelinePlaying", Qt::QueuedConnection);
break;
default:
break;
}
}
break;
}
default:
break;
}
}
void eServiceMP3Record::gstBusCall(GstMessage *msg)
{
if (!msg)
return;
ePtr<iRecordableService> ptr = this;
gchar *sourceName;
GstObject *source;
source = GST_MESSAGE_SRC(msg);
if (!GST_IS_OBJECT(source))
return;
sourceName = gst_object_get_name(source);
switch (GST_MESSAGE_TYPE (msg))
{
case GST_MESSAGE_EOS:
eDebug("[eMP3ServiceRecord] gstBusCall eos event");
// Stream end -> stop recording
m_event((iRecordableService*)this, evGstRecordEnded);
break;
case GST_MESSAGE_STATE_CHANGED:
{
if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_recording_pipeline))
break;
GstState old_state, new_state;
gst_message_parse_state_changed(msg, &old_state, &new_state, NULL);
if(old_state == new_state)
break;
GstStateChange transition = (GstStateChange)GST_STATE_TRANSITION(old_state, new_state);
eDebug("[eMP3ServiceRecord] gstBusCall state transition %s -> %s", gst_element_state_get_name(old_state), gst_element_state_get_name(new_state));
switch(transition)
{
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
{
if (m_streamingsrc_timeout)
m_streamingsrc_timeout->stop();
break;
}
default:
break;
}
break;
}
case GST_MESSAGE_ERROR:
{
gchar *debug;
GError *err;
gst_message_parse_error(msg, &err, &debug);
g_free(debug);
if (err->code != GST_STREAM_ERROR_CODEC_NOT_FOUND)
eWarning("[eServiceMP3Record] gstBusCall Gstreamer error: %s (%i) from %s", err->message, err->code, sourceName);
g_error_free(err);
break;
}
case GST_MESSAGE_ELEMENT:
{
const GstStructure *msgstruct = gst_message_get_structure(msg);
if (msgstruct)
{
if (gst_is_missing_plugin_message(msg))
{
GstCaps *caps = NULL;
gst_structure_get (msgstruct, "detail", GST_TYPE_CAPS, &caps, NULL);
if (caps)
{
std::string codec = (const char*) gst_caps_to_string(caps);
eDebug("[eServiceMP3Record] gstBusCall cannot record because of incompatible codecs %s", codec.c_str());
gst_caps_unref(caps);
}
}
else
{
const gchar *eventname = gst_structure_get_name(msgstruct);
if (eventname)
{
if (!strcmp(eventname, "redirect"))
{
const char *uri = gst_structure_get_string(msgstruct, "new-location");
eDebug("[eServiceMP3Record] gstBusCall redirect to %s", uri);
gst_element_set_state (m_recording_pipeline, GST_STATE_NULL);
g_object_set(G_OBJECT (m_source), "uri", uri, NULL);
gst_element_set_state (m_recording_pipeline, GST_STATE_PLAYING);
}
}
}
}
break;
}
case GST_MESSAGE_STREAM_STATUS:
{
GstStreamStatusType type;
GstElement *owner;
gst_message_parse_stream_status (msg, &type, &owner);
if (type == GST_STREAM_STATUS_TYPE_CREATE)
{
if (GST_IS_PAD(source))
owner = gst_pad_get_parent_element(GST_PAD(source));
else if (GST_IS_ELEMENT(source))
owner = GST_ELEMENT(source);
else
owner = 0;
if (owner)
{
GstState state;
gst_element_get_state(m_recording_pipeline, &state, NULL, 0LL);
GstElementFactory *factory = gst_element_get_factory(GST_ELEMENT(owner));
const gchar *name = gst_plugin_feature_get_name(GST_PLUGIN_FEATURE(factory));
if (!strcmp(name, "souphttpsrc") && (state == GST_STATE_READY) && !m_streamingsrc_timeout->isActive())
{
m_streamingsrc_timeout->start(HTTP_TIMEOUT*1000, true);
g_object_set (G_OBJECT (owner), "timeout", HTTP_TIMEOUT, NULL);
eDebug("[eServiceMP3Record] gstBusCall setting timeout on %s to %is", name, HTTP_TIMEOUT);
}
}
if (GST_IS_PAD(source))
gst_object_unref(owner);
}
break;
}
default:
break;
}
g_free(sourceName);
}
static gboolean
bus_message (GstBus * bus, GstMessage * message, App * app)
{
gchar *sourceName;
GstObject *source;
gchar *string;
GstState current_state;
if (!message)
return FALSE;
source = GST_MESSAGE_SRC (message);
if (!GST_IS_OBJECT (source))
return FALSE;
sourceName = gst_object_get_name (source);
if (gst_message_get_structure (message))
string = gst_structure_to_string (gst_message_get_structure (message));
else
string = g_strdup (GST_MESSAGE_TYPE_NAME (message));
GST_DEBUG("gst_message from %s: %s", sourceName, string);
g_free (string);
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
{
GError *gerror;
gchar *debug;
gst_message_parse_error (message, &gerror, &debug);
gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
g_error_free (gerror);
g_free (debug);
g_main_loop_quit (app->loop);
break;
}
case GST_MESSAGE_WARNING:
{
GError *gerror;
gchar *debug;
gst_message_parse_warning (message, &gerror, &debug);
gst_object_default_error (GST_MESSAGE_SRC (message), gerror, debug);
g_error_free (gerror);
g_free (debug);
// g_main_loop_quit (app->loop);
break;
}
case GST_MESSAGE_EOS:
g_message ("received EOS");
g_main_loop_quit (app->loop);
break;
case GST_MESSAGE_ASYNC_DONE:
break;
case GST_MESSAGE_ELEMENT:
{
const GstStructure *msgstruct = gst_message_get_structure (message);
if (msgstruct) {
const gchar *eventname = gst_structure_get_name (msgstruct);
if (!strcmp (eventname, "seekable"))
app->is_seekable = TRUE;
}
break;
}
case GST_MESSAGE_STATE_CHANGED:
{
GstState old_state, new_state;
GstStateChange transition;
if (GST_MESSAGE_SRC (message) != GST_OBJECT (app->tsdemux))
break;
gst_message_parse_state_changed (message, &old_state, &new_state, NULL);
transition = (GstStateChange) GST_STATE_TRANSITION (old_state, new_state);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
{
}
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
}
break;
}
case GST_MESSAGE_SEGMENT_DONE:
{
GST_DEBUG ("GST_MESSAGE_SEGMENT_DONE!!!");
do_seek (app);
}
default:
break;
}
gst_element_get_state (app->pipeline, ¤t_state, NULL, 0);
if (app->current_segment == 0 && app->segment_count /*&& app->is_seekable*/
&& current_state == GST_STATE_PLAYING)
do_seek (app);
GST_DEBUG_BIN_TO_DOT_FILE(GST_BIN(app->pipeline),GST_DEBUG_GRAPH_SHOW_ALL,"bdremux_pipelinegraph_message");
return TRUE;
}
void eServiceMP3::gstBusCall(GstMessage *msg)
{
if (!msg)
return;
gchar *sourceName;
GstObject *source;
source = GST_MESSAGE_SRC(msg);
if (!GST_IS_OBJECT(source))
return;
sourceName = gst_object_get_name(source);
#if 0
gchar *string;
if (gst_message_get_structure(msg))
string = gst_structure_to_string(gst_message_get_structure(msg));
else
string = g_strdup(GST_MESSAGE_TYPE_NAME(msg));
eDebug("eTsRemoteSource::gst_message from %s: %s", sourceName, string);
g_free(string);
#endif
switch (GST_MESSAGE_TYPE (msg))
{
case GST_MESSAGE_EOS:
m_event((iPlayableService*)this, evEOF);
break;
case GST_MESSAGE_STATE_CHANGED:
{
if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_gst_playbin))
break;
GstState old_state, new_state;
gst_message_parse_state_changed(msg, &old_state, &new_state, NULL);
if(old_state == new_state)
break;
eDebug("eServiceMP3::state transition %s -> %s", gst_element_state_get_name(old_state), gst_element_state_get_name(new_state));
GstStateChange transition = (GstStateChange)GST_STATE_TRANSITION(old_state, new_state);
switch(transition)
{
case GST_STATE_CHANGE_NULL_TO_READY:
{
} break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
{
GstElement *subsink = gst_bin_get_by_name(GST_BIN(m_gst_playbin), "subtitle_sink");
if (subsink)
{
#ifdef GSTREAMER_SUBTITLE_SYNC_MODE_BUG
/*
* HACK: disable sync mode for now, gstreamer suffers from a bug causing sparse streams to loose sync, after pause/resume / skip
* see: https://bugzilla.gnome.org/show_bug.cgi?id=619434
* Sideeffect of using sync=false is that we receive subtitle buffers (far) ahead of their
* display time.
* Not too far ahead for subtitles contained in the media container.
* But for external srt files, we could receive all subtitles at once.
* And not just once, but after each pause/resume / skip.
* So as soon as gstreamer has been fixed to keep sync in sparse streams, sync needs to be re-enabled.
*/
g_object_set (G_OBJECT (subsink), "sync", FALSE, NULL);
#endif
#if 0
/* we should not use ts-offset to sync with the decoder time, we have to do our own decoder timekeeping */
g_object_set (G_OBJECT (subsink), "ts-offset", -2L * GST_SECOND, NULL);
/* late buffers probably will not occur very often */
g_object_set (G_OBJECT (subsink), "max-lateness", 0L, NULL);
/* avoid prerolling (it might not be a good idea to preroll a sparse stream) */
g_object_set (G_OBJECT (subsink), "async", TRUE, NULL);
#endif
eDebug("eServiceMP3::subsink properties set!");
gst_object_unref(subsink);
}
setAC3Delay(ac3_delay);
setPCMDelay(pcm_delay);
} break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
{
if ( m_sourceinfo.is_streaming && m_streamingsrc_timeout )
m_streamingsrc_timeout->stop();
} break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
{
} break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
} break;
case GST_STATE_CHANGE_READY_TO_NULL:
{
} break;
}
break;
}
case GST_MESSAGE_ERROR:
{
gchar *debug;
GError *err;
gst_message_parse_error (msg, &err, &debug);
g_free (debug);
eWarning("Gstreamer error: %s (%i) from %s", err->message, err->code, sourceName );
if ( err->domain == GST_STREAM_ERROR )
{
if ( err->code == GST_STREAM_ERROR_CODEC_NOT_FOUND )
{
if ( g_strrstr(sourceName, "videosink") )
m_event((iPlayableService*)this, evUser+11);
else if ( g_strrstr(sourceName, "audiosink") )
m_event((iPlayableService*)this, evUser+10);
}
}
g_error_free(err);
break;
}
case GST_MESSAGE_INFO:
{
gchar *debug;
GError *inf;
gst_message_parse_info (msg, &inf, &debug);
g_free (debug);
if ( inf->domain == GST_STREAM_ERROR && inf->code == GST_STREAM_ERROR_DECODE )
{
if ( g_strrstr(sourceName, "videosink") )
m_event((iPlayableService*)this, evUser+14);
}
g_error_free(inf);
break;
}
case GST_MESSAGE_TAG:
{
GstTagList *tags, *result;
gst_message_parse_tag(msg, &tags);
result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_REPLACE);
if (result)
{
if (m_stream_tags)
gst_tag_list_free(m_stream_tags);
m_stream_tags = result;
}
const GValue *gv_image = gst_tag_list_get_value_index(tags, GST_TAG_IMAGE, 0);
if ( gv_image )
{
GstBuffer *buf_image;
buf_image = gst_value_get_buffer (gv_image);
int fd = open("/tmp/.id3coverart", O_CREAT|O_WRONLY|O_TRUNC, 0644);
int ret = write(fd, GST_BUFFER_DATA(buf_image), GST_BUFFER_SIZE(buf_image));
close(fd);
eDebug("eServiceMP3::/tmp/.id3coverart %d bytes written ", ret);
m_event((iPlayableService*)this, evUser+13);
}
gst_tag_list_free(tags);
m_event((iPlayableService*)this, evUpdatedInfo);
break;
}
case GST_MESSAGE_ASYNC_DONE:
{
if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_gst_playbin))
break;
GstTagList *tags;
gint i, active_idx, n_video = 0, n_audio = 0, n_text = 0;
g_object_get (m_gst_playbin, "n-video", &n_video, NULL);
g_object_get (m_gst_playbin, "n-audio", &n_audio, NULL);
g_object_get (m_gst_playbin, "n-text", &n_text, NULL);
eDebug("eServiceMP3::async-done - %d video, %d audio, %d subtitle", n_video, n_audio, n_text);
if ( n_video + n_audio <= 0 )
stop();
active_idx = 0;
m_audioStreams.clear();
m_subtitleStreams.clear();
for (i = 0; i < n_audio; i++)
{
audioStream audio;
gchar *g_codec, *g_lang;
GstPad* pad = 0;
g_signal_emit_by_name (m_gst_playbin, "get-audio-pad", i, &pad);
GstCaps* caps = gst_pad_get_negotiated_caps(pad);
if (!caps)
continue;
GstStructure* str = gst_caps_get_structure(caps, 0);
const gchar *g_type = gst_structure_get_name(str);
eDebug("AUDIO STRUCT=%s", g_type);
audio.type = gstCheckAudioPad(str);
g_codec = g_strdup(g_type);
g_lang = g_strdup_printf ("und");
g_signal_emit_by_name (m_gst_playbin, "get-audio-tags", i, &tags);
if ( tags && gst_is_tag_list(tags) )
{
gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_codec);
gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_lang);
gst_tag_list_free(tags);
}
audio.language_code = std::string(g_lang);
audio.codec = std::string(g_codec);
eDebug("eServiceMP3::audio stream=%i codec=%s language=%s", i, g_codec, g_lang);
m_audioStreams.push_back(audio);
g_free (g_lang);
g_free (g_codec);
gst_caps_unref(caps);
}
for (i = 0; i < n_text; i++)
{
gchar *g_codec = NULL, *g_lang = NULL;
g_signal_emit_by_name (m_gst_playbin, "get-text-tags", i, &tags);
subtitleStream subs;
g_lang = g_strdup_printf ("und");
if ( tags && gst_is_tag_list(tags) )
{
gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_lang);
gst_tag_list_get_string(tags, GST_TAG_SUBTITLE_CODEC, &g_codec);
gst_tag_list_free(tags);
}
subs.language_code = std::string(g_lang);
eDebug("eServiceMP3::subtitle stream=%i language=%s codec=%s", i, g_lang, g_codec);
GstPad* pad = 0;
g_signal_emit_by_name (m_gst_playbin, "get-text-pad", i, &pad);
if ( pad )
g_signal_connect (G_OBJECT (pad), "notify::caps", G_CALLBACK (gstTextpadHasCAPS), this);
subs.type = getSubtitleType(pad, g_codec);
m_subtitleStreams.push_back(subs);
g_free (g_lang);
}
m_event((iPlayableService*)this, evUpdatedInfo);
if ( m_errorInfo.missing_codec != "" )
{
if ( m_errorInfo.missing_codec.find("video/") == 0 || ( m_errorInfo.missing_codec.find("audio/") == 0 && getNumberOfTracks() == 0 ) )
m_event((iPlayableService*)this, evUser+12);
}
break;
}
case GST_MESSAGE_ELEMENT:
{
if (const GstStructure *msgstruct = gst_message_get_structure(msg))
{
if ( gst_is_missing_plugin_message(msg) )
{
GstCaps *caps;
gst_structure_get (msgstruct, "detail", GST_TYPE_CAPS, &caps, NULL);
std::string codec = (const char*) gst_caps_to_string(caps);
gchar *description = gst_missing_plugin_message_get_description(msg);
if ( description )
{
eDebug("eServiceMP3::m_errorInfo.missing_codec = %s", codec.c_str());
m_errorInfo.error_message = "GStreamer plugin " + (std::string)description + " not available!\n";
m_errorInfo.missing_codec = codec.substr(0,(codec.find_first_of(',')));
g_free(description);
}
gst_caps_unref(caps);
}
else
{
const gchar *eventname = gst_structure_get_name(msgstruct);
if ( eventname )
{
if (!strcmp(eventname, "eventSizeChanged") || !strcmp(eventname, "eventSizeAvail"))
{
gst_structure_get_int (msgstruct, "aspect_ratio", &m_aspect);
gst_structure_get_int (msgstruct, "width", &m_width);
gst_structure_get_int (msgstruct, "height", &m_height);
if (strstr(eventname, "Changed"))
m_event((iPlayableService*)this, evVideoSizeChanged);
}
else if (!strcmp(eventname, "eventFrameRateChanged") || !strcmp(eventname, "eventFrameRateAvail"))
{
gst_structure_get_int (msgstruct, "frame_rate", &m_framerate);
if (strstr(eventname, "Changed"))
m_event((iPlayableService*)this, evVideoFramerateChanged);
}
else if (!strcmp(eventname, "eventProgressiveChanged") || !strcmp(eventname, "eventProgressiveAvail"))
{
gst_structure_get_int (msgstruct, "progressive", &m_progressive);
if (strstr(eventname, "Changed"))
m_event((iPlayableService*)this, evVideoProgressiveChanged);
}
}
}
}
break;
}
case GST_MESSAGE_BUFFERING:
{
GstBufferingMode mode;
gst_message_parse_buffering(msg, &(m_bufferInfo.bufferPercent));
gst_message_parse_buffering_stats(msg, &mode, &(m_bufferInfo.avgInRate), &(m_bufferInfo.avgOutRate), &(m_bufferInfo.bufferingLeft));
m_event((iPlayableService*)this, evBuffering);
break;
}
case GST_MESSAGE_STREAM_STATUS:
{
GstStreamStatusType type;
GstElement *owner;
gst_message_parse_stream_status (msg, &type, &owner);
if ( type == GST_STREAM_STATUS_TYPE_CREATE && m_sourceinfo.is_streaming )
{
if ( GST_IS_PAD(source) )
owner = gst_pad_get_parent_element(GST_PAD(source));
else if ( GST_IS_ELEMENT(source) )
owner = GST_ELEMENT(source);
else
owner = 0;
if ( owner )
{
GstElementFactory *factory = gst_element_get_factory(GST_ELEMENT(owner));
const gchar *name = gst_plugin_feature_get_name(GST_PLUGIN_FEATURE(factory));
if (!strcmp(name, "souphttpsrc"))
{
m_streamingsrc_timeout->start(HTTP_TIMEOUT*1000, true);
g_object_set (G_OBJECT (owner), "timeout", HTTP_TIMEOUT, NULL);
eDebug("eServiceMP3::GST_STREAM_STATUS_TYPE_CREATE -> setting timeout on %s to %is", name, HTTP_TIMEOUT);
}
}
if ( GST_IS_PAD(source) )
gst_object_unref(owner);
}
break;
}
default:
break;
}
g_free (sourceName);
}
GstBusSyncReply Gst_bus_call(GstBus * bus, GstMessage *msg, gpointer user_data)
{
gchar * sourceName;
// source
GstObject * source;
source = GST_MESSAGE_SRC(msg);
if (!GST_IS_OBJECT(source))
return GST_BUS_DROP;
sourceName = gst_object_get_name(source);
switch (GST_MESSAGE_TYPE(msg))
{
case GST_MESSAGE_EOS:
{
g_message("End-of-stream");
end_eof = 1;
break;
}
case GST_MESSAGE_ERROR:
{
gchar * debug;
GError *err;
gst_message_parse_error(msg, &err, &debug);
g_free (debug);
lt_info_c( "%s:%s - GST_MESSAGE_ERROR: %s (%i) from %s\n", FILENAME, __FUNCTION__, err->message, err->code, sourceName );
if ( err->domain == GST_STREAM_ERROR )
{
if ( err->code == GST_STREAM_ERROR_CODEC_NOT_FOUND )
{
if ( g_strrstr(sourceName, "videosink") )
lt_info_c( "%s:%s - GST_MESSAGE_ERROR: videosink\n", FILENAME, __FUNCTION__ ); //FIXME: how shall playback handle this event???
else if ( g_strrstr(sourceName, "audiosink") )
lt_info_c( "%s:%s - GST_MESSAGE_ERROR: audioSink\n", FILENAME, __FUNCTION__ ); //FIXME: how shall playback handle this event???
}
}
g_error_free(err);
end_eof = 1; // NOTE: just to exit
break;
}
case GST_MESSAGE_INFO:
{
gchar *debug;
GError *inf;
gst_message_parse_info (msg, &inf, &debug);
g_free (debug);
if ( inf->domain == GST_STREAM_ERROR && inf->code == GST_STREAM_ERROR_DECODE )
{
if ( g_strrstr(sourceName, "videosink") )
lt_info_c( "%s:%s - GST_MESSAGE_INFO: videosink\n", FILENAME, __FUNCTION__ ); //FIXME: how shall playback handle this event???
}
g_error_free(inf);
break;
}
case GST_MESSAGE_TAG:
{
GstTagList *tags, *result;
gst_message_parse_tag(msg, &tags);
result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_REPLACE);
if (result)
{
if (m_stream_tags)
gst_tag_list_free(m_stream_tags);
m_stream_tags = result;
}
const GValue *gv_image = gst_tag_list_get_value_index(tags, GST_TAG_IMAGE, 0);
if ( gv_image )
{
GstBuffer *buf_image;
buf_image = gst_value_get_buffer (gv_image);
int fd = open("/tmp/.id3coverart", O_CREAT|O_WRONLY|O_TRUNC, 0644);
if(fd >= 0)
{
int ret = write(fd, GST_BUFFER_DATA(buf_image), GST_BUFFER_SIZE(buf_image));
close(fd);
lt_info_c( "%s:%s - GST_MESSAGE_INFO: cPlayback::state /tmp/.id3coverart %d bytes written\n", FILENAME, __FUNCTION__ , ret);
}
//FIXME: how shall playback handle this event???
}
gst_tag_list_free(tags);
lt_info_c( "%s:%s - GST_MESSAGE_INFO: update info tags\n", FILENAME, __FUNCTION__); //FIXME: how shall playback handle this event???
break;
}
case GST_MESSAGE_STATE_CHANGED:
{
if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_gst_playbin))
break;
GstState old_state, new_state;
gst_message_parse_state_changed(msg, &old_state, &new_state, NULL);
if(old_state == new_state)
break;
lt_info_c( "%s:%s - GST_MESSAGE_STATE_CHANGED: state transition %s -> %s\n", FILENAME, __FUNCTION__, gst_element_state_get_name(old_state), gst_element_state_get_name(new_state));
GstStateChange transition = (GstStateChange)GST_STATE_TRANSITION(old_state, new_state);
switch(transition)
{
case GST_STATE_CHANGE_NULL_TO_READY:
{
} break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
{
GstIterator *children;
if (audioSink)
{
gst_object_unref(GST_OBJECT(audioSink));
audioSink = NULL;
}
if (videoSink)
{
gst_object_unref(GST_OBJECT(videoSink));
videoSink = NULL;
}
children = gst_bin_iterate_recurse(GST_BIN(m_gst_playbin));
audioSink = GST_ELEMENT_CAST(gst_iterator_find_custom(children, (GCompareFunc)match_sinktype, (gpointer)"GstDVBAudioSink"));
videoSink = GST_ELEMENT_CAST(gst_iterator_find_custom(children, (GCompareFunc)match_sinktype, (gpointer)"GstDVBVideoSink"));
gst_iterator_free(children);
} break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
{
} break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
{
} break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
if (audioSink)
{
gst_object_unref(GST_OBJECT(audioSink));
audioSink = NULL;
}
if (videoSink)
{
gst_object_unref(GST_OBJECT(videoSink));
videoSink = NULL;
}
} break;
case GST_STATE_CHANGE_READY_TO_NULL:
{
} break;
}
break;
}
#if 0
case GST_MESSAGE_ELEMENT:
{
if(gst_structure_has_name(gst_message_get_structure(msg), "prepare-xwindow-id"))
{
// set window id
gst_x_overlay_set_xwindow_id(GST_X_OVERLAY(GST_MESSAGE_SRC (msg)), glfb->getWindowID());
// reshape window
gst_x_overlay_set_render_rectangle(GST_X_OVERLAY(GST_MESSAGE_SRC (msg)), 0, 0, glfb->getOSDWidth(), glfb->getOSDHeight());
// sync frames
gst_x_overlay_expose(GST_X_OVERLAY(GST_MESSAGE_SRC (msg)));
}
}
#endif
break;
default:
break;
}
return GST_BUS_DROP;
}
static gboolean
gst_PlayRegion_handle_message (GstBus * bus, GstMessage * message, gpointer data)
{
GstPlayRegion *PlayRegion = (GstPlayRegion *) data;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_APPLICATION:
if (gst_message_has_name (message, "ExPrerolled")) {
/* it's our message */
g_print ("we are all prerolled, do seek\n");
gst_element_seek (PlayRegion->pipeline,
1.0, GST_FORMAT_TIME,
GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE,
GST_SEEK_TYPE_SET, 2 * GST_SECOND,
GST_SEEK_TYPE_SET, 5 * GST_SECOND);
gst_element_set_state (PlayRegion->pipeline, GST_STATE_PLAYING);
}
break;
case GST_MESSAGE_EOS:
gst_PlayRegion_handle_eos (PlayRegion);
break;
case GST_MESSAGE_ERROR:
{
GError *error = NULL;
gchar *debug;
gst_message_parse_error (message, &error, &debug);
gst_PlayRegion_handle_error (PlayRegion, error, debug);
}
break;
case GST_MESSAGE_WARNING:
{
GError *error = NULL;
gchar *debug;
gst_message_parse_warning (message, &error, &debug);
gst_PlayRegion_handle_warning (PlayRegion, error, debug);
}
break;
case GST_MESSAGE_INFO:
{
GError *error = NULL;
gchar *debug;
gst_message_parse_info (message, &error, &debug);
gst_PlayRegion_handle_info (PlayRegion, error, debug);
}
break;
case GST_MESSAGE_TAG:
{
GstTagList *tag_list;
gst_message_parse_tag (message, &tag_list);
if (verbose)
g_print ("tag\n");
}
break;
case GST_MESSAGE_STATE_CHANGED:
{
GstState oldstate, newstate, pending;
gst_message_parse_state_changed (message, &oldstate, &newstate, &pending);
if (GST_ELEMENT (message->src) == PlayRegion->pipeline) {
if (verbose)
g_print ("state change from %s to %s\n",
gst_element_state_get_name (oldstate),
gst_element_state_get_name (newstate));
switch (GST_STATE_TRANSITION (oldstate, newstate)) {
case GST_STATE_CHANGE_NULL_TO_READY:
gst_PlayRegion_handle_null_to_ready (PlayRegion);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_PlayRegion_handle_ready_to_paused (PlayRegion);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_PlayRegion_handle_paused_to_playing (PlayRegion);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
gst_PlayRegion_handle_playing_to_paused (PlayRegion);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_PlayRegion_handle_paused_to_ready (PlayRegion);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_PlayRegion_handle_ready_to_null (PlayRegion);
break;
default:
if (verbose)
g_print ("unknown state change from %s to %s\n",
gst_element_state_get_name (oldstate),
gst_element_state_get_name (newstate));
}
}
}
break;
case GST_MESSAGE_BUFFERING:
{
int percent;
gst_message_parse_buffering (message, &percent);
//g_print("buffering %d\n", percent);
if (!PlayRegion->paused_for_buffering && percent < 100) {
g_print ("pausing for buffing\n");
PlayRegion->paused_for_buffering = TRUE;
gst_element_set_state (PlayRegion->pipeline, GST_STATE_PAUSED);
} else if (PlayRegion->paused_for_buffering && percent == 100) {
g_print ("unpausing for buffing\n");
PlayRegion->paused_for_buffering = FALSE;
gst_element_set_state (PlayRegion->pipeline, GST_STATE_PLAYING);
}
}
break;
case GST_MESSAGE_STATE_DIRTY:
case GST_MESSAGE_CLOCK_PROVIDE:
case GST_MESSAGE_CLOCK_LOST:
case GST_MESSAGE_NEW_CLOCK:
case GST_MESSAGE_STRUCTURE_CHANGE:
case GST_MESSAGE_STREAM_STATUS:
break;
case GST_MESSAGE_STEP_DONE:
case GST_MESSAGE_ELEMENT:
case GST_MESSAGE_SEGMENT_START:
case GST_MESSAGE_SEGMENT_DONE:
case GST_MESSAGE_DURATION:
case GST_MESSAGE_LATENCY:
case GST_MESSAGE_ASYNC_START:
case GST_MESSAGE_ASYNC_DONE:
case GST_MESSAGE_REQUEST_STATE:
case GST_MESSAGE_STEP_START:
case GST_MESSAGE_QOS:
default:
if (verbose) {
g_print ("message: %s\n", GST_MESSAGE_TYPE_NAME (message));
}
break;
}
return TRUE;
}