void eServiceMP3::gstTextpadHasCAPS_synced(GstPad *pad)
{
GstCaps *caps;
g_object_get (G_OBJECT (pad), "caps", &caps, NULL);
eDebug("gstTextpadHasCAPS:: signal::caps = %s", gst_caps_to_string(caps));
if (caps)
{
subtitleStream subs;
// eDebug("gstGhostpadHasCAPS_synced %p %d", pad, m_subtitleStreams.size());
if (m_currentSubtitleStream >= 0 && m_currentSubtitleStream < m_subtitleStreams.size())
subs = m_subtitleStreams[m_currentSubtitleStream];
else {
subs.type = stUnknown;
subs.pad = pad;
}
if ( subs.type == stUnknown )
{
GstTagList *tags;
// eDebug("gstGhostpadHasCAPS::m_subtitleStreams[%i].type == stUnknown...", m_currentSubtitleStream);
gchar *g_lang;
g_signal_emit_by_name (m_gst_playbin, "get-text-tags", m_currentSubtitleStream, &tags);
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);
subs.language_code = std::string(g_lang);
subs.type = getSubtitleType(pad);
if (m_currentSubtitleStream >= 0 && m_currentSubtitleStream < m_subtitleStreams.size())
m_subtitleStreams[m_currentSubtitleStream] = subs;
else
m_subtitleStreams.push_back(subs);
g_free (g_lang);
}
// eDebug("gstGhostpadHasCAPS:: m_gst_prev_subtitle_caps=%s equal=%i",gst_caps_to_string(m_gst_prev_subtitle_caps),gst_caps_is_equal(m_gst_prev_subtitle_caps, caps));
gst_caps_unref (caps);
}
}
void printEvent(const DBEventEIT& event, const QString& name)
{
printf("\n------------Event - %s------------\n", name.toLocal8Bit().constData());
printf("Title %s\n", event.title.toLocal8Bit().constData());
printf("Subtitle %s\n", event.subtitle.toLocal8Bit().constData());
printf("Description %s\n", event.description.toLocal8Bit().constData());
printf("Season %3u\n", event.season);
printf("Episode %3u\n", event.episode);
printf("Total episodes %3u\n", event.totalepisodes);
printf("Part number %3u\n", event.partnumber);
printf("Part total %3u\n", event.parttotal);
printf("SubtitleType %s\n", getSubtitleType(event.subtitleType).toLocal8Bit().constData());
printf("Audio props %s\n", getAudioProps(event.audioProps).toLocal8Bit().constData());
printf("Video props %s\n", getVideoProps(event.videoProps).toLocal8Bit().constData());
printf("\n");
}
void printEvent(const DBEventEIT& event, const QString& name)
{
printf("\n------------Event - %s------------\n", name.toLocal8Bit().constData());
printf("Title %s\n", event.title.toLocal8Bit().constData());
printf("Subtitle %s\n", event.subtitle.toLocal8Bit().constData());
printf("Description %s\n", event.description.toLocal8Bit().constData());
printf("Season %3u\n", event.season);
printf("Episode %3u\n", event.episode);
printf("Total episodes %3u\n", event.totalepisodes);
printf("Part number %3u\n", event.partnumber);
printf("Part total %3u\n", event.parttotal);
printf("SubtitleType %s\n", getSubtitleType(event.subtitleType).toLocal8Bit().constData());
printf("Audio props %s\n", getAudioProps(event.audioProps).toLocal8Bit().constData());
printf("Video props %s\n", getVideoProps(event.videoProps).toLocal8Bit().constData());
if (event.credits && !event.credits->empty())
{
printf("Credits %3zu\n", event.credits->size());
}
if (!event.items.isEmpty())
{
printf("Items %3d\n", event.items.count());
}
printf("\n");
}
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);
}
