static void
_bus_handler (GstBus * bus, GstMessage * message,
GstValidatePipelineMonitor * monitor)
{
GError *err = NULL;
gchar *debug = NULL;
const GstStructure *details = NULL;
gint error_flow = GST_FLOW_OK;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (message, &err, &debug);
gst_message_parse_error_details (message, &details);
if (g_error_matches (err, GST_CORE_ERROR, GST_CORE_ERROR_MISSING_PLUGIN)) {
GST_VALIDATE_REPORT (monitor, MISSING_PLUGIN,
"Error: %s -- Debug message: %s", err->message, debug);
} else if ((g_error_matches (err, GST_STREAM_ERROR,
GST_STREAM_ERROR_FAILED) && details
&& gst_structure_get_int (details, "flow-return", &error_flow)
&& error_flow == GST_FLOW_NOT_NEGOTIATED)
|| g_error_matches (err, GST_STREAM_ERROR, GST_STREAM_ERROR_FORMAT)) {
gchar *report = _generate_not_negotiated_error_report (message);
GST_VALIDATE_REPORT (monitor, NOT_NEGOTIATED, "%s", report);
g_free (report);
} else {
GST_VALIDATE_REPORT (monitor, ERROR_ON_BUS,
"Got error: %s -- Debug message: %s", err->message, debug);
}
GST_VALIDATE_MONITOR_LOCK (monitor);
monitor->got_error = TRUE;
GST_VALIDATE_MONITOR_UNLOCK (monitor);
g_error_free (err);
g_free (debug);
break;
case GST_MESSAGE_WARNING:
gst_message_parse_warning (message, &err, &debug);
GST_VALIDATE_REPORT (monitor, WARNING_ON_BUS,
"Got warning: %s -- Debug message: %s", err->message, debug);
g_error_free (err);
g_free (debug);
break;
case GST_MESSAGE_STATE_CHANGED:
{
if (GST_MESSAGE_SRC (message) == GST_VALIDATE_MONITOR (monitor)->target) {
GstState oldstate, newstate, pending;
gst_message_parse_state_changed (message, &oldstate, &newstate,
&pending);
if (oldstate == GST_STATE_READY && newstate == GST_STATE_PAUSED) {
monitor->print_pos_srcid =
g_timeout_add (PRINT_POSITION_TIMEOUT,
(GSourceFunc) print_position, monitor);
} else if (oldstate >= GST_STATE_PAUSED && newstate <= GST_STATE_READY) {
if (monitor->print_pos_srcid
&& g_source_remove (monitor->print_pos_srcid))
monitor->print_pos_srcid = 0;
monitor->got_error = FALSE;
}
}
break;
}
case GST_MESSAGE_BUFFERING:
{
JsonBuilder *jbuilder = json_builder_new ();
GstBufferingMode mode;
gint percent;
gst_message_parse_buffering (message, &percent);
gst_message_parse_buffering_stats (message, &mode, NULL, NULL, NULL);
json_builder_begin_object (jbuilder);
json_builder_set_member_name (jbuilder, "type");
json_builder_add_string_value (jbuilder, "buffering");
json_builder_set_member_name (jbuilder, "state");
if (percent == 100) {
/* a 100% message means buffering is done */
gst_validate_printf (NULL, "\nDone buffering\n");
json_builder_add_string_value (jbuilder, "done");
if (monitor->buffering) {
monitor->print_pos_srcid =
g_timeout_add (PRINT_POSITION_TIMEOUT,
(GSourceFunc) print_position, monitor);
monitor->buffering = FALSE;
}
} else {
/* buffering... */
if (!monitor->buffering) {
monitor->buffering = TRUE;
gst_validate_printf (NULL, "\nStart buffering\n");
json_builder_add_string_value (jbuilder, "started");
if (monitor->print_pos_srcid
&& g_source_remove (monitor->print_pos_srcid)) {
monitor->print_pos_srcid = 0;
}
} else {
json_builder_add_string_value (jbuilder, "progress");
}
gst_validate_printf (NULL, "%s %d%% \r", "Buffering...", percent);
}
json_builder_set_member_name (jbuilder, "position");
json_builder_add_int_value (jbuilder, percent);
json_builder_end_object (jbuilder);
gst_validate_send (json_builder_get_root (jbuilder));
g_object_unref (jbuilder);
break;
}
case GST_MESSAGE_STREAM_COLLECTION:
{
GstStreamCollection *collection = NULL;
gst_message_parse_stream_collection (message, &collection);
gst_object_replace ((GstObject **) & monitor->stream_collection,
(GstObject *) collection);
gst_object_unref (collection);
break;
}
case GST_MESSAGE_STREAMS_SELECTED:
{
guint i;
if (monitor->streams_selected) {
g_list_free_full (monitor->streams_selected, gst_object_unref);
monitor->streams_selected = NULL;
}
for (i = 0; i < gst_message_streams_selected_get_size (message); i++) {
GstStream *stream =
gst_message_streams_selected_get_stream (message, i);
monitor->streams_selected =
g_list_append (monitor->streams_selected, stream);
}
break;
}
default:
break;
}
}
int BufferingMessage::averageOutputRate() const
{
gint a;
gst_message_parse_buffering_stats(object<GstMessage>(), NULL, NULL, &a, NULL);
return a;
}
qint64 BufferingMessage::bufferingTimeLeft() const
{
gint64 a;
gst_message_parse_buffering_stats(object<GstMessage>(), NULL, NULL, NULL, &a);
return a;
}
BufferingMode BufferingMessage::mode() const
{
GstBufferingMode m;
gst_message_parse_buffering_stats(object<GstMessage>(), &m, NULL, NULL, NULL);
return static_cast<BufferingMode>(m);
}
static gboolean
master_bus_msg (GstBus * bus, GstMessage * msg, gpointer data)
{
GstPipeline *pipeline = data;
switch (GST_MESSAGE_TYPE (msg)) {
case GST_MESSAGE_ERROR:{
GError *err;
gchar *dbg;
gst_message_parse_error (msg, &err, &dbg);
g_printerr ("MASTER: ERROR: %s\n", err->message);
if (dbg != NULL)
g_printerr ("MASTER: ERROR debug information: %s\n", dbg);
g_error_free (err);
g_free (dbg);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, "ipc.master.error");
g_main_loop_quit (loop);
break;
}
case GST_MESSAGE_WARNING:{
GError *err;
gchar *dbg;
gst_message_parse_warning (msg, &err, &dbg);
g_printerr ("MASTER: WARNING: %s\n", err->message);
if (dbg != NULL)
g_printerr ("MASTER: WARNING debug information: %s\n", dbg);
g_error_free (err);
g_free (dbg);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, "ipc.master.warning");
break;
}
case GST_MESSAGE_ASYNC_DONE:
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
GST_DEBUG_GRAPH_SHOW_ALL, "ipc.master.async-done");
break;
case GST_MESSAGE_EOS:
g_print ("EOS on master\n");
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
g_main_loop_quit (loop);
break;
case GST_MESSAGE_BUFFERING:{
gint percent;
GstBufferingMode bufmode;
if (!buffering)
g_print ("\n");
gst_message_parse_buffering (msg, &percent);
g_print ("%s %d%% \r", "Buffering...", percent);
gst_message_parse_buffering_stats (msg, &bufmode, NULL, NULL, NULL);
/* no state management needed for live pipelines */
if (bufmode != GST_BUFFERING_LIVE) {
if (percent == 100) {
/* a 100% message means buffering is done */
if (buffering) {
buffering = FALSE;
gst_element_set_state (GST_ELEMENT (pipeline), desired_state);
g_print ("\n%s\n", gst_element_state_get_name (desired_state));
}
} else {
/* buffering... */
if (!buffering) {
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PAUSED);
buffering = TRUE;
}
}
}
break;
}
case GST_MESSAGE_CLOCK_LOST:{
g_print ("Clock lost, selecting a new one\n");
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PAUSED);
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
break;
}
case GST_MESSAGE_LATENCY:
{
gst_bin_recalculate_latency (GST_BIN (pipeline));
break;
}
case GST_MESSAGE_REQUEST_STATE:{
GstState state;
gchar *name;
name = gst_object_get_path_string (GST_MESSAGE_SRC (msg));
gst_message_parse_request_state (msg, &state);
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS (GST_BIN (pipeline),
GST_DEBUG_GRAPH_SHOW_VERBOSE, "ipc.slave.reqstate");
g_print ("Setting state to %s as requested by %s...\n",
gst_element_state_get_name (state), name);
gst_element_set_state (GST_ELEMENT (pipeline), state);
g_free (name);
break;
}
case GST_MESSAGE_ELEMENT:
{
GstNavigationMessageType mtype = gst_navigation_message_get_type (msg);
if (mtype == GST_NAVIGATION_MESSAGE_EVENT) {
GstEvent *ev = NULL;
if (gst_navigation_message_parse_event (msg, &ev)) {
GstNavigationEventType e_type = gst_navigation_event_get_type (ev);
switch (e_type) {
case GST_NAVIGATION_EVENT_KEY_PRESS:
{
const gchar *key;
if (gst_navigation_event_parse_key_event (ev, &key)) {
GST_INFO ("Key press: %s", key);
if (strcmp (key, "Left") == 0)
key = GST_PLAY_KB_ARROW_LEFT;
else if (strcmp (key, "Right") == 0)
key = GST_PLAY_KB_ARROW_RIGHT;
else if (strcmp (key, "Up") == 0)
key = GST_PLAY_KB_ARROW_UP;
else if (strcmp (key, "Down") == 0)
key = GST_PLAY_KB_ARROW_DOWN;
else if (strcmp (key, "space") == 0)
key = " ";
else if (strlen (key) > 1)
break;
keyboard_cb (key, GST_ELEMENT (pipeline));
}
break;
}
case GST_NAVIGATION_EVENT_MOUSE_BUTTON_PRESS:
{
gint button;
if (gst_navigation_event_parse_mouse_button_event (ev, &button,
NULL, NULL)) {
if (button == 4) {
/* wheel up */
relative_seek (GST_ELEMENT (pipeline), +0.08);
} else if (button == 5) {
/* wheel down */
relative_seek (GST_ELEMENT (pipeline), -0.01);
}
}
break;
}
default:
break;
}
}
if (ev)
gst_event_unref (ev);
}
break;
}
default:
break;
}
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);
}
static void
_bus_handler (GstBus * bus, GstMessage * message,
GstValidatePipelineMonitor * monitor)
{
GError *err = NULL;
gchar *debug = NULL;
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ERROR:
gst_message_parse_error (message, &err, &debug);
if (g_error_matches (err, GST_CORE_ERROR, GST_CORE_ERROR_MISSING_PLUGIN)) {
GST_VALIDATE_REPORT (monitor, MISSING_PLUGIN,
"Error: %s -- Debug message: %s", err->message, debug);
} else {
GST_VALIDATE_REPORT (monitor, ERROR_ON_BUS,
"Got error: %s -- Debug message: %s", err->message, debug);
}
GST_VALIDATE_MONITOR_LOCK (monitor);
monitor->got_error = TRUE;
GST_VALIDATE_MONITOR_UNLOCK (monitor);
g_error_free (err);
g_free (debug);
break;
case GST_MESSAGE_WARNING:
gst_message_parse_warning (message, &err, &debug);
GST_VALIDATE_REPORT (monitor, WARNING_ON_BUS,
"Got warning: %s -- Debug message: %s", err->message, debug);
g_error_free (err);
g_free (debug);
break;
case GST_MESSAGE_STATE_CHANGED:
{
if (GST_MESSAGE_SRC (message) == GST_VALIDATE_MONITOR (monitor)->target) {
GstState oldstate, newstate, pending;
gst_message_parse_state_changed (message, &oldstate, &newstate,
&pending);
if (oldstate == GST_STATE_READY && newstate == GST_STATE_PAUSED) {
monitor->print_pos_srcid =
g_timeout_add (PRINT_POSITION_TIMEOUT,
(GSourceFunc) print_position, monitor);
} else if (oldstate >= GST_STATE_PAUSED && newstate <= GST_STATE_READY) {
if (monitor->print_pos_srcid
&& g_source_remove (monitor->print_pos_srcid))
monitor->print_pos_srcid = 0;
monitor->got_error = FALSE;
}
}
break;
}
case GST_MESSAGE_BUFFERING:
{
GstBufferingMode mode;
gint percent;
gst_message_parse_buffering (message, &percent);
gst_message_parse_buffering_stats (message, &mode, NULL, NULL, NULL);
if (percent == 100) {
/* a 100% message means buffering is done */
if (monitor->buffering) {
monitor->print_pos_srcid =
g_timeout_add (PRINT_POSITION_TIMEOUT,
(GSourceFunc) print_position, monitor);
monitor->buffering = FALSE;
}
} else {
/* buffering... */
if (!monitor->buffering) {
monitor->buffering = TRUE;
if (monitor->print_pos_srcid
&& g_source_remove (monitor->print_pos_srcid))
monitor->print_pos_srcid = 0;
}
}
break;
}
default:
break;
}
}
void __mm_player_streaming_buffering(mm_player_streaming_t* streamer, GstMessage *buffering_msg)
{
GstBufferingMode mode = GST_BUFFERING_STREAM;
gint byte_in_rate = 0;
gint byte_out_rate = 0;
gint64 buffering_left = -1;
gdouble buffering_time = DEFAULT_BUFFERING_TIME;
gdouble low_percent = 0.0;
gdouble high_percent = 0.0;
guint high_percent_byte = 0;
gint buffer_percent = 0;
guint buffer_criteria = 0;
return_if_fail ( streamer );
return_if_fail ( buffering_msg );
return_if_fail ( GST_IS_MESSAGE ( buffering_msg ) );
return_if_fail ( GST_MESSAGE_TYPE ( buffering_msg ) == GST_MESSAGE_BUFFERING );
/* update when buffering has started. */
if ( !streamer->is_buffering )
{
debug_log ( "buffering has started.\n" );
streamer->is_buffering = TRUE;
streamer->buffering_percent = -1;
streamer->need_update = TRUE;
}
/* update buffer percent */
gst_message_parse_buffering ( buffering_msg, &buffer_percent );
if ( streamer->buffering_percent < buffer_percent )
{
debug_log ( "buffering %d%%....\n", buffer_percent );
streamer->buffering_percent = buffer_percent;
}
if ( streamer->buffering_percent == MAX_BUFFER_PERCENT )
{
debug_log ( "buffering had done.\n" );
streamer->is_buffering = FALSE;
}
if (!streamer->need_update)
{
debug_log ( "don't need to update buffering stats during buffering.\n" );
return;
}
/* Note : Parse the buffering message to get the in/out throughput.
* avg_in is the network throughput and avg_out is the consumed throughtput by the linkded element.
*/
gst_message_parse_buffering_stats ( buffering_msg, &mode, &byte_in_rate, &byte_out_rate, &buffering_left );
if (streamer->buffer_max_bitrate > 0)
{
buffer_criteria = GET_BYTE_FROM_BIT(streamer->buffer_max_bitrate);
byte_out_rate = GET_BYTE_FROM_BIT(streamer->buffer_max_bitrate /3);
}
else if (streamer->buffer_avg_bitrate > 0)
{
buffer_criteria = GET_BYTE_FROM_BIT(streamer->buffer_avg_bitrate * 3);
byte_out_rate = GET_BYTE_FROM_BIT(streamer->buffer_avg_bitrate);
}
debug_log ( "in rate is %d, out rate is %d (bytes/sec).\n", byte_in_rate, byte_out_rate );
if ( byte_in_rate > 0 && byte_out_rate > 0)
buffering_time = byte_out_rate / byte_in_rate;
else if (byte_in_rate <= 0 && byte_out_rate > 0)
buffering_time = MAX_BUFFERING_TIME;
else
buffering_time = DEFAULT_BUFFERING_TIME;
streaming_set_buffering_time(streamer, buffering_time);
/* calculate buffer low/high percent */
low_percent = DEFAULT_BUFFER_LOW_PERCENT;
if ( buffer_criteria > 0 )
{
high_percent_byte = buffer_criteria * streamer->buffering_time;
high_percent = ( (gdouble)high_percent_byte * 100.0 ) / (gdouble)streamer->buffer_size;
}
else
{
high_percent_byte = streamer->buffer_high_percent * streamer->buffer_size / 100;
high_percent= streamer->buffer_high_percent;
}
if ( streamer->buffer_size < high_percent_byte )
{
debug_log ( "buffer size[%d bytes] is smaller than high threshold[%d bytes]. update it. \n",
streamer->buffer_size, high_percent_byte );
streaming_set_buffer_size(streamer, high_percent_byte * 1.1);
}
streaming_set_buffer_percent(streamer, low_percent, high_percent);
streamer->need_update = FALSE;
return;
}
