if (sc)
{
std::list<std::string> extensions;
+ extensions.push_back("mp2");
extensions.push_back("mp3");
extensions.push_back("ogg");
extensions.push_back("mpg");
extensions.push_back("wave");
extensions.push_back("mkv");
extensions.push_back("avi");
+ extensions.push_back("divx");
extensions.push_back("dat");
extensions.push_back("flac");
extensions.push_back("mp4");
+ extensions.push_back("m4a");
sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions);
}
eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
{
+ m_seekTimeout = eTimer::create(eApp);
m_stream_tags = 0;
- m_audioStreams.clear();
- m_subtitleStreams.clear();
m_currentAudioStream = 0;
m_currentSubtitleStream = 0;
m_subtitle_widget = 0;
m_currentTrickRatio = 0;
- CONNECT(m_seekTimeout.timeout, eServiceMP3::seekTimeoutCB);
+ CONNECT(m_seekTimeout->timeout, eServiceMP3::seekTimeoutCB);
CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
GstElement *source = 0;
-
- GstElement *decoder = 0, *conv = 0, *flt = 0, *sink = 0; /* for audio */
-
- GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 0;
+ GstElement *decoder = 0, *conv = 0, *flt = 0, *parser = 0, *sink = 0; /* for audio */
+ GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 0, *audiodemux = 0, *id3demux;
m_state = stIdle;
eDebug("SERVICEMP3 construct!");
ext = filename;
sourceStream sourceinfo;
+ sourceinfo.is_video = FALSE;
+ sourceinfo.audiotype = atUnknown;
if ( (strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin") && strcasecmp(ext, ".dat") ) == 0 )
+ {
sourceinfo.containertype = ctMPEGPS;
+ sourceinfo.is_video = TRUE;
+ }
else if ( strcasecmp(ext, ".ts") == 0 )
+ {
sourceinfo.containertype = ctMPEGTS;
+ sourceinfo.is_video = TRUE;
+ }
else if ( strcasecmp(ext, ".mkv") == 0 )
+ {
sourceinfo.containertype = ctMKV;
- else if ( strcasecmp(ext, ".avi") == 0 )
+ sourceinfo.is_video = TRUE;
+ }
+ else if ( strcasecmp(ext, ".avi") == 0 || strcasecmp(ext, ".divx") == 0)
+ {
sourceinfo.containertype = ctAVI;
+ sourceinfo.is_video = TRUE;
+ }
else if ( strcasecmp(ext, ".mp4") == 0 )
+ {
sourceinfo.containertype = ctMP4;
+ sourceinfo.is_video = TRUE;
+ }
+ else if ( strcasecmp(ext, ".m4a") == 0 )
+ {
+ sourceinfo.containertype = ctMP4;
+ sourceinfo.audiotype = atAAC;
+ }
+ else if ( strcasecmp(ext, ".mp3") == 0 )
+ sourceinfo.audiotype = atMP3;
else if ( (strncmp(filename, "/autofs/", 8) || strncmp(filename+strlen(filename)-13, "/track-", 7) || strcasecmp(ext, ".wav")) == 0 )
sourceinfo.containertype = ctCDA;
if ( strcasecmp(ext, ".dat") == 0 )
+ {
sourceinfo.containertype = ctVCD;
+ sourceinfo.is_video = TRUE;
+ }
if ( (strncmp(filename, "http://", 7)) == 0 )
sourceinfo.is_streaming = TRUE;
- sourceinfo.is_video = ( sourceinfo.containertype && sourceinfo.containertype != ctCDA );
-
eDebug("filename=%s, containertype=%d, is_video=%d, is_streaming=%d", filename, sourceinfo.containertype, sourceinfo.is_video, sourceinfo.is_streaming);
int all_ok = 0;
if (!m_gst_pipeline)
m_error_message = "failed to create GStreamer pipeline!\n";
- if ( sourceinfo.containertype == ctCDA )
+ if ( sourceinfo.is_streaming )
+ {
+ eDebug("play webradio!");
+ source = gst_element_factory_make ("neonhttpsrc", "http-source");
+ if (source)
+ {
+ g_object_set (G_OBJECT (source), "location", filename, NULL);
+ g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
+ }
+ else
+ m_error_message = "GStreamer plugin neonhttpsrc not available!\n";
+ }
+ else if ( sourceinfo.containertype == ctCDA )
{
source = gst_element_factory_make ("cdiocddasrc", "cda-source");
if (source)
+ {
g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL);
- else
- sourceinfo.containertype = ctNone;
+ int track = atoi(filename+18);
+ eDebug("play audio CD track #%i",track);
+ if (track > 0)
+ g_object_set (G_OBJECT (source), "track", track, NULL);
+ }
+ }
+ else if ( sourceinfo.containertype == ctVCD )
+ {
+ int fd = open(filename,O_RDONLY);
+ char tmp[128*1024];
+ int ret = read(fd, tmp, 128*1024);
+ close(fd);
+ if ( ret == -1 ) // this is a "REAL" VCD
+ source = gst_element_factory_make ("vcdsrc", "vcd-source");
+ if (source)
+ g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL);
}
- if ( !sourceinfo.is_streaming && sourceinfo.containertype != ctCDA )
+ if ( !source && !sourceinfo.is_streaming )
{
source = gst_element_factory_make ("filesrc", "file-source");
if (source)
else
m_error_message = "GStreamer can't open filesrc " + (std::string)filename + "!\n";
}
- else if ( sourceinfo.is_streaming )
- {
- source = gst_element_factory_make ("neonhttpsrc", "http-source");
- if (source)
- g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
- else
- m_error_message = "GStreamer plugin neonhttpsrc not available!\n";
- }
- else
- {
- int track = atoi(filename+18);
- eDebug("play audio CD track #%i",track);
- if (track > 0)
- g_object_set (G_OBJECT (source), "track", track, NULL);
- }
if ( sourceinfo.is_video )
{
/* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
audio = gst_element_factory_make("dvbaudiosink", "audiosink");
if (!audio)
m_error_message += "failed to create Gstreamer element dvbaudiosink\n";
-
+
video = gst_element_factory_make("dvbvideosink", "videosink");
if (!video)
m_error_message += "failed to create Gstreamer element dvbvideosink\n";
queue_audio = gst_element_factory_make("queue", "queue_audio");
queue_video = gst_element_factory_make("queue", "queue_video");
- char demux_type[14];
+ std::string demux_type;
switch (sourceinfo.containertype)
{
case ctMPEGTS:
- strcat(demux_type, "flutsdemux");
+ demux_type = "flutsdemux";
break;
case ctMPEGPS:
case ctVCD:
- strcat(demux_type, "flupsdemux");
+ demux_type = "flupsdemux";
break;
case ctMKV:
- strcat(demux_type, "matroskademux");
+ demux_type = "matroskademux";
break;
case ctAVI:
- strcat(demux_type, "avidemux");
+ demux_type = "avidemux";
break;
case ctMP4:
- strcat(demux_type, "qtdemux");
+ demux_type = "qtdemux";
break;
default:
break;
}
- videodemux = gst_element_factory_make(demux_type, "videodemux");
+ videodemux = gst_element_factory_make(demux_type.c_str(), "videodemux");
if (!videodemux)
- m_error_message = "GStreamer plugin " + (std::string)demux_type + " not available!\n";
+ m_error_message = "GStreamer plugin " + demux_type + " not available!\n";
switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
if (!switch_audio)
}
} else /* is audio */
{
-
- /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
- decoder = gst_element_factory_make ("decodebin", "decoder");
- if (!decoder)
- m_error_message += "failed to create Gstreamer element decodebin\n";
-
- conv = gst_element_factory_make ("audioconvert", "converter");
- if (!conv)
- m_error_message += "failed to create Gstreamer element audioconvert\n";
-
- flt = gst_element_factory_make ("capsfilter", "flt");
- if (!flt)
- m_error_message += "failed to create Gstreamer element capsfilter\n";
-
- /* for some reasons, we need to set the sample format to depth/width=16, because auto negotiation doesn't work. */
- /* endianness, however, is not required to be set anymore. */
- if (flt)
+ std::string demux_type;
+ switch ( sourceinfo.containertype )
+ {
+ case ctMP4:
+ demux_type = "qtdemux";
+ break;
+ default:
+ break;
+ }
+ if ( demux_type.length() )
{
- GstCaps *caps = gst_caps_new_simple("audio/x-raw-int", /* "endianness", G_TYPE_INT, 4321, */ "depth", G_TYPE_INT, 16, "width", G_TYPE_INT, 16, /*"channels", G_TYPE_INT, 2, */NULL);
- g_object_set (G_OBJECT (flt), "caps", caps, NULL);
- gst_caps_unref(caps);
+ audiodemux = gst_element_factory_make(demux_type.c_str(), "audiodemux");
+ if (!audiodemux)
+ m_error_message = "GStreamer plugin " + demux_type + " not available!\n";
+ }
+ switch ( sourceinfo.audiotype )
+ {
+ case atMP3:
+ {
+ id3demux = gst_element_factory_make("id3demux", "id3demux");
+ if ( !id3demux )
+ {
+ m_error_message += "failed to create Gstreamer element id3demux\n";
+ break;
+ }
+ parser = gst_element_factory_make("mp3parse", "audiosink");
+ if ( !parser )
+ {
+ m_error_message += "failed to create Gstreamer element mp3parse\n";
+ break;
+ }
+ sink = gst_element_factory_make("dvbaudiosink", "audiosink2");
+ if ( !sink )
+ m_error_message += "failed to create Gstreamer element dvbaudiosink\n";
+ else
+ all_ok = 1;
+ break;
+ }
+ case atAAC:
+ {
+ if ( !audiodemux )
+ {
+ m_error_message += "cannot parse raw AAC audio\n";
+ break;
+ }
+ sink = gst_element_factory_make("dvbaudiosink", "audiosink");
+ if (!sink)
+ m_error_message += "failed to create Gstreamer element dvbaudiosink\n";
+ else
+ all_ok = 1;
+ break;
+ }
+ case atAC3:
+ {
+ if ( !audiodemux )
+ {
+ m_error_message += "cannot parse raw AC3 audio\n";
+ break;
+ }
+ sink = gst_element_factory_make("dvbaudiosink", "audiosink");
+ if ( !sink )
+ m_error_message += "failed to create Gstreamer element dvbaudiosink\n";
+ else
+ all_ok = 1;
+ break;
+ }
+ default:
+ { /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
+ decoder = gst_element_factory_make ("decodebin", "decoder");
+ if (!decoder)
+ m_error_message += "failed to create Gstreamer element decodebin\n";
+
+ conv = gst_element_factory_make ("audioconvert", "converter");
+ if (!conv)
+ m_error_message += "failed to create Gstreamer element audioconvert\n";
+
+ flt = gst_element_factory_make ("capsfilter", "flt");
+ if (!flt)
+ m_error_message += "failed to create Gstreamer element capsfilter\n";
+
+ /* for some reasons, we need to set the sample format to depth/width=16, because auto negotiation doesn't work. */
+ /* endianness, however, is not required to be set anymore. */
+ if (flt)
+ {
+ GstCaps *caps = gst_caps_new_simple("audio/x-raw-int", /* "endianness", G_TYPE_INT, 4321, */ "depth", G_TYPE_INT, 16, "width", G_TYPE_INT, 16, /*"channels", G_TYPE_INT, 2, */NULL);
+ g_object_set (G_OBJECT (flt), "caps", caps, NULL);
+ gst_caps_unref(caps);
+ }
+
+ sink = gst_element_factory_make ("alsasink", "alsa-output");
+ if (!sink)
+ m_error_message += "failed to create Gstreamer element alsasink\n";
+
+ if (source && decoder && conv && sink)
+ all_ok = 1;
+ break;
+ }
}
- sink = gst_element_factory_make ("alsasink", "alsa-output");
- if (!sink)
- m_error_message += "failed to create Gstreamer element alsasink\n";
-
- if (source && decoder && conv && sink)
- all_ok = 1;
}
if (m_gst_pipeline && all_ok)
{
}
gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, audio, queue_audio, video, queue_video, switch_audio, NULL);
- if ( sourceinfo.containertype == ctVCD )
+ if ( sourceinfo.containertype == ctVCD && gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"file-source") )
{
+ eDebug("this is a fake video cd... we use filesrc ! cdxaparse !");
GstElement *cdxaparse = gst_element_factory_make("cdxaparse", "cdxaparse");
gst_bin_add(GST_BIN(m_gst_pipeline), cdxaparse);
gst_element_link(source, cdxaparse);
} else /* is audio*/
{
- queue_audio = gst_element_factory_make("queue", "queue_audio");
-
- g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
- g_signal_connect (decoder, "unknown-type", G_CALLBACK(gstCBunknownType), this);
-
- g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
-
- /* gst_bin will take the 'floating references' */
- gst_bin_add_many (GST_BIN (m_gst_pipeline),
- source, queue_audio, decoder, NULL);
-
- /* in decodebin's case we can just connect the source with the decodebin, and decodebin will take care about id3demux (or whatever is required) */
- gst_element_link_many(source, queue_audio, decoder, NULL);
-
- /* create audio bin with the audioconverter, the capsfilter and the audiosink */
- audio = gst_bin_new ("audiobin");
-
- GstPad *audiopad = gst_element_get_static_pad (conv, "sink");
- gst_bin_add_many(GST_BIN(audio), conv, flt, sink, NULL);
- gst_element_link_many(conv, flt, sink, NULL);
- gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));
- gst_object_unref(audiopad);
- gst_bin_add (GST_BIN(m_gst_pipeline), audio);
+ if ( decoder )
+ {
+ queue_audio = gst_element_factory_make("queue", "queue_audio");
+
+ g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
+ g_signal_connect (decoder, "unknown-type", G_CALLBACK(gstCBunknownType), this);
+
+ g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
+
+ /* gst_bin will take the 'floating references' */
+ gst_bin_add_many (GST_BIN (m_gst_pipeline),
+ source, queue_audio, decoder, NULL);
+
+ /* in decodebin's case we can just connect the source with the decodebin, and decodebin will take care about id3demux (or whatever is required) */
+ gst_element_link_many(source, queue_audio, decoder, NULL);
+
+ /* create audio bin with the audioconverter, the capsfilter and the audiosink */
+ audio = gst_bin_new ("audiobin");
+
+ GstPad *audiopad = gst_element_get_static_pad (conv, "sink");
+ gst_bin_add_many(GST_BIN(audio), conv, flt, sink, NULL);
+ gst_element_link_many(conv, flt, sink, NULL);
+ gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));
+ gst_object_unref(audiopad);
+ gst_bin_add (GST_BIN(m_gst_pipeline), audio);
+ }
+ else
+ {
+ gst_bin_add_many (GST_BIN (m_gst_pipeline), source, sink, NULL);
+ if ( parser && id3demux )
+ {
+ gst_bin_add_many (GST_BIN (m_gst_pipeline), parser, id3demux, NULL);
+ gst_element_link(source, id3demux);
+ g_signal_connect(id3demux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+ gst_element_link(parser, sink);
+ }
+ if ( audiodemux )
+ {
+ gst_bin_add (GST_BIN (m_gst_pipeline), audiodemux);
+ g_signal_connect(audiodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+ gst_element_link(source, audiodemux);
+ }
+ audioStream audio;
+ audio.type = sourceinfo.audiotype;
+ m_audioStreams.push_back(audio);
+ }
}
} else
{
{
m_currentTrickRatio = ratio;
if (ratio)
- m_seekTimeout.start(1000, 0);
+ m_seekTimeout->start(1000, 0);
else
- m_seekTimeout.stop();
+ m_seekTimeout->stop();
return 0;
}
if (ppos < 0)
{
ppos = 0;
- m_seekTimeout.stop();
+ m_seekTimeout->stop();
}
if (ppos > len)
{
ppos = 0;
stop();
- m_seekTimeout.stop();
+ m_seekTimeout->stop();
return;
}
seekTo(ppos);
#endif
switch (GST_MESSAGE_TYPE (msg))
{
- case GST_MESSAGE_EOS:
- m_event((iPlayableService*)this, evEOF);
- break;
- case GST_MESSAGE_ERROR:
- {
- gchar *debug;
- GError *err;
-
- gst_message_parse_error (msg, &err, &debug);
- g_free (debug);
- eWarning("Gstreamer error: %s (%i)", err->message, err->code );
- if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_DECODE )
+ case GST_MESSAGE_EOS:
+ m_event((iPlayableService*)this, evEOF);
+ break;
+ case GST_MESSAGE_ERROR:
{
- if ( g_strrstr(sourceName, "videosink") )
- m_event((iPlayableService*)this, evUser+11);
+ gchar *debug;
+ GError *err;
+
+ gst_message_parse_error (msg, &err, &debug);
+ g_free (debug);
+ eWarning("Gstreamer error: %s (%i)", err->message, err->code );
+ if ( err->domain == GST_STREAM_ERROR && err->code == GST_STREAM_ERROR_CODEC_NOT_FOUND )
+ {
+ if ( g_strrstr(sourceName, "videosink") )
+ m_event((iPlayableService*)this, evUser+11);
+ }
+ g_error_free(err);
+ break;
}
- g_error_free(err);
- /* TODO: signal error condition to user */
- 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_PREPEND);
- if (result)
+ case GST_MESSAGE_INFO:
{
- if (m_stream_tags)
- gst_tag_list_free(m_stream_tags);
- m_stream_tags = result;
+ 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;
}
-
- gchar *g_audiocodec;
- if ( gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec) && m_audioStreams.size() == 0 )
+ case GST_MESSAGE_TAG:
{
- GstPad* pad = gst_element_get_pad (GST_ELEMENT(source), "src");
- GstCaps* caps = gst_pad_get_caps(pad);
- GstStructure* str = gst_caps_get_structure(caps, 0);
- if ( !str )
- break;
- audioStream audio;
- audio.type = gstCheckAudioPad(str);
- m_audioStreams.push_back(audio);
+ GstTagList *tags, *result;
+ gst_message_parse_tag(msg, &tags);
+
+ result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND);
+ if (result)
+ {
+ if (m_stream_tags)
+ gst_tag_list_free(m_stream_tags);
+ m_stream_tags = result;
+ }
+
+ gchar *g_audiocodec;
+ if ( gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec) && m_audioStreams.size() == 0 )
+ {
+ GstPad* pad = gst_element_get_pad (GST_ELEMENT(source), "src");
+ GstCaps* caps = gst_pad_get_caps(pad);
+ GstStructure* str = gst_caps_get_structure(caps, 0);
+ if ( !str )
+ break;
+ audioStream audio;
+ audio.type = gstCheckAudioPad(str);
+ m_audioStreams.push_back(audio);
+ }
+
+ 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);
+ write(fd, GST_BUFFER_DATA(buf_image), GST_BUFFER_SIZE(buf_image));
+ close(fd);
+ m_event((iPlayableService*)this, evUser+13);
+ }
+
+ gst_tag_list_free(tags);
+ m_event((iPlayableService*)this, evUpdatedInfo);
+ break;
}
-
- gst_tag_list_free(tags);
- m_event((iPlayableService*)this, evUpdatedInfo);
- break;
- }
- case GST_MESSAGE_ASYNC_DONE:
- {
- GstTagList *tags;
- for (std::vector<audioStream>::iterator IterAudioStream(m_audioStreams.begin()); IterAudioStream != m_audioStreams.end(); ++IterAudioStream)
+ case GST_MESSAGE_ASYNC_DONE:
{
- if ( IterAudioStream->pad )
+ GstTagList *tags;
+ for (std::vector<audioStream>::iterator IterAudioStream(m_audioStreams.begin()); IterAudioStream != m_audioStreams.end(); ++IterAudioStream)
{
- g_object_get(IterAudioStream->pad, "tags", &tags, NULL);
- gchar *g_language;
- if ( gst_is_tag_list(tags) && gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
+ if ( IterAudioStream->pad )
{
- eDebug("found audio language %s",g_language);
- IterAudioStream->language_code = std::string(g_language);
- g_free (g_language);
+ g_object_get(IterAudioStream->pad, "tags", &tags, NULL);
+ gchar *g_language;
+ if ( tags && gst_is_tag_list(tags) && gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
+ {
+ eDebug("found audio language %s",g_language);
+ IterAudioStream->language_code = std::string(g_language);
+ g_free (g_language);
+ }
}
}
- }
- for (std::vector<subtitleStream>::iterator IterSubtitleStream(m_subtitleStreams.begin()); IterSubtitleStream != m_subtitleStreams.end(); ++IterSubtitleStream)
- {
- if ( IterSubtitleStream->pad )
+ for (std::vector<subtitleStream>::iterator IterSubtitleStream(m_subtitleStreams.begin()); IterSubtitleStream != m_subtitleStreams.end(); ++IterSubtitleStream)
{
- g_object_get(IterSubtitleStream->pad, "tags", &tags, NULL);
- gchar *g_language;
- if ( gst_is_tag_list(tags) && gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
+ if ( IterSubtitleStream->pad )
{
- eDebug("found subtitle language %s",g_language);
- IterSubtitleStream->language_code = std::string(g_language);
- g_free (g_language);
+ g_object_get(IterSubtitleStream->pad, "tags", &tags, NULL);
+ gchar *g_language;
+ if ( tags && gst_is_tag_list(tags) && gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
+ {
+ eDebug("found subtitle language %s",g_language);
+ IterSubtitleStream->language_code = std::string(g_language);
+ g_free (g_language);
+ }
}
}
}
- }
- case GST_MESSAGE_ELEMENT:
- {
- if ( gst_is_missing_plugin_message(msg) )
+ case GST_MESSAGE_ELEMENT:
{
- gchar *description = gst_missing_plugin_message_get_description(msg);
- if ( description )
+ if ( gst_is_missing_plugin_message(msg) )
{
- m_error_message = "GStreamer plugin " + (std::string)description + " not available!\n";
- g_free(description);
- m_event((iPlayableService*)this, evUser+12);
+ gchar *description = gst_missing_plugin_message_get_description(msg);
+ if ( description )
+ {
+ m_error_message = "GStreamer plugin " + (std::string)description + " not available!\n";
+ g_free(description);
+ m_event((iPlayableService*)this, evUser+12);
+ }
+ }
+ else if (const GstStructure *msgstruct = gst_message_get_structure(msg))
+ {
+ const gchar *eventname;
+ if ( eventname = gst_structure_get_name(msgstruct) )
+ {
+ if (!strcmp(eventname, "eventSizeChanged"))
+ {
+ gint aspect_ratio, width, height = 0;
+ gst_structure_get_int (msgstruct, "aspect_ratio", &aspect_ratio);
+ gst_structure_get_int (msgstruct, "width", &width);
+ gst_structure_get_int (msgstruct, "height", &height);
+ eDebug("****** decoder threw eventSizeChanged! aspect_ratio=%i, width=%i, height=%i", aspect_ratio, width, height);
+ }
+ if (!strcmp(eventname, "eventFrameRateChanged"))
+ {
+ gint frame_rate = 0;
+ gst_structure_get_int (msgstruct, "frame_rate", &frame_rate);
+ eDebug("****** decoder threw eventFrameRateChanged! frame_rate=%i", frame_rate);
+ }
+ if (!strcmp(eventname, "eventProgressiveChanged"))
+ {
+ gint progressive = 0;
+ gst_structure_get_int (msgstruct, "progressive", &progressive);
+ eDebug("****** decoder threw eventProgressiveChanged! progressive=%i", progressive);
+ }
+ }
}
}
- }
- default:
- break;
+ default:
+ break;
}
g_free (sourceName);
}
}
else
{
- gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink"));
- _this->m_audioStreams.push_back(audio);
+ GstElement *queue_audio = gst_bin_get_by_name(pipeline , "queue_audio");
+ if ( queue_audio )
+ {
+ gst_pad_link(pad, gst_element_get_static_pad(queue_audio, "sink"));
+ _this->m_audioStreams.push_back(audio);
+ }
+ else
+ gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline , "audiosink"), "sink"));
}
}
if (g_strrstr(type,"video"))
git.cweiske.de / enigma2.git / blobdiff