/* note: this requires gstreamer 0.10.x and a big list of plugins. */
/* it's currently hardcoded to use a big-endian alsasink as sink. */
+#include <lib/base/ebase.h>
#include <lib/base/eerror.h>
+#include <lib/base/init_num.h>
+#include <lib/base/init.h>
+#include <lib/base/nconfig.h>
#include <lib/base/object.h>
-#include <lib/base/ebase.h>
-#include <string>
+#include <lib/dvb/decoder.h>
+#include <lib/components/file_eraser.h>
+#include <lib/gui/esubtitle.h>
#include <lib/service/servicemp3.h>
#include <lib/service/service.h>
-#include <lib/components/file_eraser.h>
-#include <lib/base/init_num.h>
-#include <lib/base/init.h>
+
+#include <string>
+
#include <gst/gst.h>
#include <gst/pbutils/missing-plugins.h>
#include <sys/stat.h>
-/* for subtitles */
-#include <lib/gui/esubtitle.h>
// eServiceFactoryMP3
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("mov");
+ extensions.push_back("m4a");
sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions);
}
RESULT eServiceFactoryMP3::play(const eServiceReference &ref, ePtr<iPlayableService> &ptr)
{
// check resources...
- ptr = new eServiceMP3(ref.path.c_str());
+ ptr = new eServiceMP3(ref);
return 0;
}
RESULT deleteFromDisk(int simulate);
RESULT getListOfFilenames(std::list<std::string> &);
+ RESULT reindex();
};
DEFINE_REF(eMP3ServiceOfflineOperations);
return 0;
}
+RESULT eMP3ServiceOfflineOperations::reindex()
+{
+ return -1;
+}
+
RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &ref, ePtr<iServiceOfflineOperations> &ptr)
{
RESULT eStaticServiceMP3Info::getName(const eServiceReference &ref, std::string &name)
{
- size_t last = ref.path.rfind('/');
- if (last != std::string::npos)
- name = ref.path.substr(last+1);
+ if ( ref.name.length() )
+ name = ref.name;
else
- name = ref.path;
+ {
+ size_t last = ref.path.rfind('/');
+ if (last != std::string::npos)
+ name = ref.path.substr(last+1);
+ else
+ name = ref.path;
+ }
return 0;
}
}
// eServiceMP3
+int eServiceMP3::ac3_delay,
+ eServiceMP3::pcm_delay;
-eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
+eServiceMP3::eServiceMP3(eServiceReference ref)
+ :m_ref(ref), m_pump(eApp, 1)
{
m_seekTimeout = eTimer::create(eApp);
+ m_subtitle_sync_timer = eTimer::create(eApp);
m_stream_tags = 0;
- m_currentAudioStream = 0;
+ m_currentAudioStream = -1;
m_currentSubtitleStream = 0;
m_subtitle_widget = 0;
m_currentTrickRatio = 0;
+ m_subs_to_pull = 0;
+ m_buffer_size = 1*1024*1024;
CONNECT(m_seekTimeout->timeout, eServiceMP3::seekTimeoutCB);
+ CONNECT(m_subtitle_sync_timer->timeout, eServiceMP3::pushSubtitles);
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;
-
+ m_aspect = m_width = m_height = m_framerate = m_progressive = -1;
+
m_state = stIdle;
- eDebug("SERVICEMP3 construct!");
-
- /* FIXME: currently, decodebin isn't possible for
- video streams. in that case, make a manual pipeline. */
+ eDebug("eServiceMP3::construct!");
+ const char *filename = m_ref.path.c_str();
const char *ext = strrchr(filename, '.');
if (!ext)
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;
- else if ( strcasecmp(ext, ".mp4") == 0 )
+ sourceinfo.is_video = TRUE;
+ }
+ else if ( strcasecmp(ext, ".mp4") == 0 || strcasecmp(ext, ".mov") == 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;
- if ( (strncmp(filename, "http://", 7)) == 0 )
+ sourceinfo.is_video = TRUE;
+ }
+ if ( (strncmp(filename, "http://", 7)) == 0 || (strncmp(filename, "udp://", 6)) == 0 || (strncmp(filename, "rtp://", 6)) == 0 || (strncmp(filename, "https://", 8)) == 0 || (strncmp(filename, "mms://", 6)) == 0 || (strncmp(filename, "rtsp://", 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;
+ gchar *uri;
- m_gst_pipeline = gst_pipeline_new ("mediaplayer");
- if (!m_gst_pipeline)
- m_error_message = "failed to create GStreamer pipeline!\n";
-
- if ( sourceinfo.containertype == ctCDA )
+ if ( sourceinfo.is_streaming )
{
- 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;
+ uri = g_strdup_printf ("%s", filename);
}
- if ( !sourceinfo.is_streaming && sourceinfo.containertype != ctCDA )
+ else if ( sourceinfo.containertype == ctCDA )
{
- source = gst_element_factory_make ("filesrc", "file-source");
- if (source)
- g_object_set (G_OBJECT (source), "location", filename, NULL);
- else
- m_error_message = "GStreamer can't open filesrc " + (std::string)filename + "!\n";
+ int i_track = atoi(filename+18);
+ uri = g_strdup_printf ("cdda://%i", i_track);
}
- else if ( sourceinfo.is_streaming )
+ else if ( sourceinfo.containertype == ctVCD )
{
- source = gst_element_factory_make ("neonhttpsrc", "http-source");
- if (source)
- g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
+ 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
+ uri = g_strdup_printf ("vcd://");
else
- m_error_message = "GStreamer plugin neonhttpsrc not available!\n";
+ uri = g_strdup_printf ("file://%s", filename);
}
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
- | queue_video -> dvbvideosink */
-
- 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");
- std::string demux_type;
- switch (sourceinfo.containertype)
- {
- case ctMPEGTS:
- demux_type = "flutsdemux";
- break;
- case ctMPEGPS:
- case ctVCD:
- demux_type = "flupsdemux";
- break;
- case ctMKV:
- demux_type = "matroskademux";
- break;
- case ctAVI:
- demux_type = "avidemux";
- break;
- case ctMP4:
- demux_type = "qtdemux";
- break;
- default:
- break;
- }
- videodemux = gst_element_factory_make(demux_type.c_str(), "videodemux");
- if (!videodemux)
- m_error_message = "GStreamer plugin " + demux_type + " not available!\n";
+ uri = g_strdup_printf ("file://%s", filename);
- switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
- if (!switch_audio)
- m_error_message = "GStreamer plugin input-selector not available!\n";
+ eDebug("eServiceMP3::playbin2 uri=%s", uri);
- if (audio && queue_audio && video && queue_video && videodemux && switch_audio)
- {
- g_object_set (G_OBJECT (queue_audio), "max-size-bytes", 256*1024, NULL);
- g_object_set (G_OBJECT (queue_audio), "max-size-buffers", 0, NULL);
- g_object_set (G_OBJECT (queue_audio), "max-size-time", (guint64)0, NULL);
- g_object_set (G_OBJECT (queue_video), "max-size-buffers", 0, NULL);
- g_object_set (G_OBJECT (queue_video), "max-size-bytes", 2*1024*1024, NULL);
- g_object_set (G_OBJECT (queue_video), "max-size-time", (guint64)0, NULL);
- g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL);
- all_ok = 1;
- }
- } 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";
+ m_gst_playbin = gst_element_factory_make("playbin2", "playbin");
+ if (!m_gst_playbin)
+ m_error_message = "failed to create GStreamer pipeline!\n";
- conv = gst_element_factory_make ("audioconvert", "converter");
- if (!conv)
- m_error_message += "failed to create Gstreamer element audioconvert\n";
+ g_object_set (G_OBJECT (m_gst_playbin), "uri", uri, NULL);
- flt = gst_element_factory_make ("capsfilter", "flt");
- if (!flt)
- m_error_message += "failed to create Gstreamer element capsfilter\n";
+ int flags = 0x47; // ( == GST_PLAY_FLAG_VIDEO | GST_PLAY_FLAG_AUDIO | GST_PLAY_FLAG_NATIVE_VIDEO | GST_PLAY_FLAG_TEXT )
+ g_object_set (G_OBJECT (m_gst_playbin), "flags", flags, NULL);
- /* 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);
- }
+ g_free(uri);
- 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)
+ GstElement *subsink = gst_element_factory_make("appsink", "subtitle_sink");
+ if (!subsink)
+ eDebug("eServiceMP3::sorry, can't play: missing gst-plugin-appsink");
+ else
{
- gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
-
- if ( sourceinfo.containertype == ctCDA )
- {
- queue_audio = gst_element_factory_make("queue", "queue_audio");
- g_object_set (G_OBJECT (sink), "preroll-queue-len", 80, NULL);
- gst_bin_add_many (GST_BIN (m_gst_pipeline), source, queue_audio, conv, sink, NULL);
- gst_element_link_many(source, queue_audio, conv, sink, NULL);
- }
- else if ( sourceinfo.is_video )
- {
- char srt_filename[strlen(filename)+1];
- strncpy(srt_filename,filename,strlen(filename)-3);
- srt_filename[strlen(filename)-3]='\0';
- strcat(srt_filename, "srt");
- struct stat buffer;
- if (stat(srt_filename, &buffer) == 0)
- {
- eDebug("subtitle file found: %s",srt_filename);
- GstElement *subsource = gst_element_factory_make ("filesrc", "srt_source");
- g_object_set (G_OBJECT (subsource), "location", srt_filename, NULL);
- gst_bin_add(GST_BIN (m_gst_pipeline), subsource);
- GstPad *switchpad = gstCreateSubtitleSink(this, stSRT);
- gst_pad_link(gst_element_get_pad (subsource, "src"), switchpad);
- subtitleStream subs;
- subs.pad = switchpad;
- subs.type = stSRT;
- subs.language_code = std::string("und");
- m_subtitleStreams.push_back(subs);
- }
- gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, audio, queue_audio, video, queue_video, switch_audio, NULL);
-
- if ( sourceinfo.containertype == ctVCD )
- {
- GstElement *cdxaparse = gst_element_factory_make("cdxaparse", "cdxaparse");
- gst_bin_add(GST_BIN(m_gst_pipeline), cdxaparse);
- gst_element_link(source, cdxaparse);
- gst_element_link(cdxaparse, videodemux);
- }
- else
- gst_element_link(source, videodemux);
-
- gst_element_link(switch_audio, queue_audio);
- gst_element_link(queue_audio, audio);
- gst_element_link(queue_video, video);
- g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+ m_subs_to_pull_handler_id = g_signal_connect (subsink, "new-buffer", G_CALLBACK (gstCBsubtitleAvail), this);
+ g_object_set (G_OBJECT (m_gst_playbin), "text-sink", subsink, NULL);
+ }
- } else /* is audio*/
+ if ( m_gst_playbin )
+ {
+ gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_playbin)), gstBusSyncHandler, this);
+ char srt_filename[strlen(filename)+1];
+ strncpy(srt_filename,filename,strlen(filename)-3);
+ srt_filename[strlen(filename)-3]='\0';
+ strcat(srt_filename, "srt");
+ struct stat buffer;
+ if (stat(srt_filename, &buffer) == 0)
{
- 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);
+ std::string suburi = "file://" + (std::string)srt_filename;
+ eDebug("eServiceMP3::subtitle uri: %s",suburi.c_str());
+ g_object_set (G_OBJECT (m_gst_playbin), "suburi", suburi.c_str(), NULL);
+ subtitleStream subs;
+ subs.type = stSRT;
+ subs.language_code = std::string("und");
+ m_subtitleStreams.push_back(subs);
}
} else
{
m_event((iPlayableService*)this, evUser+12);
- if (m_gst_pipeline)
- gst_object_unref(GST_OBJECT(m_gst_pipeline));
- if (source)
- gst_object_unref(GST_OBJECT(source));
- if (decoder)
- gst_object_unref(GST_OBJECT(decoder));
- if (conv)
- gst_object_unref(GST_OBJECT(conv));
- if (sink)
- gst_object_unref(GST_OBJECT(sink));
-
- if (audio)
- gst_object_unref(GST_OBJECT(audio));
- if (queue_audio)
- gst_object_unref(GST_OBJECT(queue_audio));
- if (video)
- gst_object_unref(GST_OBJECT(video));
- if (queue_video)
- gst_object_unref(GST_OBJECT(queue_video));
- if (videodemux)
- gst_object_unref(GST_OBJECT(videodemux));
- if (switch_audio)
- gst_object_unref(GST_OBJECT(switch_audio));
-
- eDebug("sorry, can't play: %s",m_error_message.c_str());
- m_gst_pipeline = 0;
+ if (m_gst_playbin)
+ gst_object_unref(GST_OBJECT(m_gst_playbin));
+
+ eDebug("eServiceMP3::sorry, can't play: %s",m_error_message.c_str());
+ m_gst_playbin = 0;
}
- gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
+ setBufferSize(m_buffer_size);
}
eServiceMP3::~eServiceMP3()
{
+ // disconnect subtitle callback
+ GstElement *sink;
+ g_object_get (G_OBJECT (m_gst_playbin), "text-sink", &sink, NULL);
+ if (sink)
+ {
+ g_signal_handler_disconnect (sink, m_subs_to_pull_handler_id);
+ gst_object_unref(sink);
+ }
+
delete m_subtitle_widget;
+
+ // disconnect sync handler callback
+ gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_playbin)), NULL, NULL);
+
if (m_state == stRunning)
stop();
-
+
if (m_stream_tags)
gst_tag_list_free(m_stream_tags);
- if (m_gst_pipeline)
+ if (m_gst_playbin)
{
- gst_object_unref (GST_OBJECT (m_gst_pipeline));
- eDebug("SERVICEMP3 destruct!");
+ gst_object_unref (GST_OBJECT (m_gst_playbin));
+ eDebug("eServiceMP3::destruct!");
}
}
-DEFINE_REF(eServiceMP3);
+DEFINE_REF(eServiceMP3);
RESULT eServiceMP3::connectEvent(const Slot2<void,iPlayableService*,int> &event, ePtr<eConnection> &connection)
{
RESULT eServiceMP3::start()
{
- assert(m_state == stIdle);
-
+ ASSERT(m_state == stIdle);
+
m_state = stRunning;
- if (m_gst_pipeline)
+ if (m_gst_playbin)
{
- eDebug("starting pipeline");
- gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
+ eDebug("eServiceMP3::starting pipeline");
+ gst_element_set_state (m_gst_playbin, GST_STATE_PLAYING);
}
+
m_event(this, evStart);
+
return 0;
}
RESULT eServiceMP3::stop()
{
- assert(m_state != stIdle);
+ ASSERT(m_state != stIdle);
+
if (m_state == stStopped)
return -1;
- eDebug("MP3: %s stop\n", m_filename.c_str());
- gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
+
+ eDebug("eServiceMP3::stop %s", m_ref.path.c_str());
+ gst_element_set_state(m_gst_playbin, GST_STATE_NULL);
m_state = stStopped;
+
return 0;
}
RESULT eServiceMP3::setSlowMotion(int ratio)
{
- /* we can't do slomo yet */
- return -1;
+ if (!ratio)
+ return 0;
+ eDebug("eServiceMP3::setSlowMotion ratio=%f",1/(float)ratio);
+ return trickSeek(1/(float)ratio);
}
RESULT eServiceMP3::setFastForward(int ratio)
{
- m_currentTrickRatio = ratio;
- if (ratio)
- m_seekTimeout->start(1000, 0);
- else
- m_seekTimeout->stop();
- return 0;
+ eDebug("eServiceMP3::setFastForward ratio=%i",ratio);
+ return trickSeek(ratio);
}
void eServiceMP3::seekTimeoutCB()
// iPausableService
RESULT eServiceMP3::pause()
{
- if (!m_gst_pipeline)
+ if (!m_gst_playbin || m_state != stRunning)
return -1;
- GstStateChangeReturn res = gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
- if (res == GST_STATE_CHANGE_ASYNC)
- {
- pts_t ppos;
- getPlayPosition(ppos);
- seekTo(ppos);
- }
+
+ gst_element_set_state(m_gst_playbin, GST_STATE_PAUSED);
+
return 0;
}
RESULT eServiceMP3::unpause()
{
- if (!m_gst_pipeline)
+ if (!m_gst_playbin || m_state != stRunning)
return -1;
- GstStateChangeReturn res;
- res = gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
+ gst_element_set_state(m_gst_playbin, GST_STATE_PLAYING);
+
return 0;
}
RESULT eServiceMP3::getLength(pts_t &pts)
{
- if (!m_gst_pipeline)
+ if (!m_gst_playbin)
return -1;
+
if (m_state != stRunning)
return -1;
-
+
GstFormat fmt = GST_FORMAT_TIME;
gint64 len;
- if (!gst_element_query_duration(m_gst_pipeline, &fmt, &len))
+ if (!gst_element_query_duration(m_gst_playbin, &fmt, &len))
return -1;
-
/* len is in nanoseconds. we have 90 000 pts per second. */
pts = len / 11111;
return 0;
}
-RESULT eServiceMP3::seekTo(pts_t to)
+RESULT eServiceMP3::seekToImpl(pts_t to)
{
- if (!m_gst_pipeline)
- return -1;
-
/* convert pts to nanoseconds */
gint64 time_nanoseconds = to * 11111LL;
- if (!gst_element_seek (m_gst_pipeline, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
+ if (!gst_element_seek (m_gst_playbin, 1.0, GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH,
GST_SEEK_TYPE_SET, time_nanoseconds,
GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
{
- eDebug("SEEK failed");
+ eDebug("eServiceMP3::seekTo failed");
+ return -1;
+ }
+
+ return 0;
+}
+
+RESULT eServiceMP3::seekTo(pts_t to)
+{
+ RESULT ret = -1;
+
+ if (m_gst_playbin) {
+ eSingleLocker l(m_subs_to_pull_lock); // this is needed to dont handle incomming subtitles during seek!
+ if (!(ret = seekToImpl(to)))
+ {
+ m_subtitle_pages.clear();
+ m_subs_to_pull = 0;
+ }
+ }
+
+ return ret;
+}
+
+
+RESULT eServiceMP3::trickSeek(gdouble ratio)
+{
+ if (!m_gst_playbin)
+ return -1;
+ if (!ratio)
+ return seekRelative(0, 0);
+
+ GstEvent *s_event;
+ int flags;
+ flags = GST_SEEK_FLAG_NONE;
+ flags |= GST_SEEK_FLAG_FLUSH;
+// flags |= GstSeekFlags (GST_SEEK_FLAG_ACCURATE);
+ flags |= GST_SEEK_FLAG_KEY_UNIT;
+// flags |= GstSeekFlags (GST_SEEK_FLAG_SEGMENT);
+// flags |= GstSeekFlags (GST_SEEK_FLAG_SKIP);
+
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint64 pos, len;
+ gst_element_query_duration(m_gst_playbin, &fmt, &len);
+ gst_element_query_position(m_gst_playbin, &fmt, &pos);
+
+ if ( ratio >= 0 )
+ {
+ s_event = gst_event_new_seek (ratio, GST_FORMAT_TIME, (GstSeekFlags)flags, GST_SEEK_TYPE_SET, pos, GST_SEEK_TYPE_SET, len);
+
+ eDebug("eServiceMP3::trickSeek with rate %lf to %" GST_TIME_FORMAT " ", ratio, GST_TIME_ARGS (pos));
+ }
+ else
+ {
+ s_event = gst_event_new_seek (ratio, GST_FORMAT_TIME, (GstSeekFlags)(GST_SEEK_FLAG_SKIP|GST_SEEK_FLAG_FLUSH), GST_SEEK_TYPE_NONE, -1, GST_SEEK_TYPE_NONE, -1);
+ }
+
+ if (!gst_element_send_event ( GST_ELEMENT (m_gst_playbin), s_event))
+ {
+ eDebug("eServiceMP3::trickSeek failed");
return -1;
}
+
return 0;
}
+
RESULT eServiceMP3::seekRelative(int direction, pts_t to)
{
- if (!m_gst_pipeline)
+ if (!m_gst_playbin)
return -1;
pts_t ppos;
RESULT eServiceMP3::getPlayPosition(pts_t &pts)
{
- if (!m_gst_pipeline)
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint64 pos;
+ GstElement *sink;
+ pts = 0;
+
+ if (!m_gst_playbin)
return -1;
if (m_state != stRunning)
return -1;
-
- GstFormat fmt = GST_FORMAT_TIME;
- gint64 len;
-
- if (!gst_element_query_position(m_gst_pipeline, &fmt, &len))
+
+ g_object_get (G_OBJECT (m_gst_playbin), "audio-sink", &sink, NULL);
+
+ if (!sink)
+ g_object_get (G_OBJECT (m_gst_playbin), "video-sink", &sink, NULL);
+
+ if (!sink)
return -1;
-
- /* len is in nanoseconds. we have 90 000 pts per second. */
- pts = len / 11111;
+
+ gchar *name = gst_element_get_name(sink);
+ gboolean use_get_decoder_time = strstr(name, "dvbaudiosink") || strstr(name, "dvbvideosink");
+ g_free(name);
+
+ if (use_get_decoder_time)
+ g_signal_emit_by_name(sink, "get-decoder-time", &pos);
+
+ gst_object_unref(sink);
+
+ if (!use_get_decoder_time && !gst_element_query_position(m_gst_playbin, &fmt, &pos)) {
+ eDebug("gst_element_query_position failed in getPlayPosition");
+ return -1;
+ }
+
+ /* pos is in nanoseconds. we have 90 000 pts per second. */
+ pts = pos / 11111;
return 0;
}
RESULT eServiceMP3::getName(std::string &name)
{
- name = m_filename;
- size_t n = name.rfind('/');
- if (n != std::string::npos)
- name = name.substr(n + 1);
+ std::string title = m_ref.getName();
+ if (title.empty())
+ {
+ name = m_ref.path;
+ size_t n = name.rfind('/');
+ if (n != std::string::npos)
+ name = name.substr(n + 1);
+ }
+ else
+ name = title;
return 0;
}
int eServiceMP3::getInfo(int w)
{
- gchar *tag = 0;
+ const gchar *tag = 0;
switch (w)
{
- case sTitle:
- case sArtist:
- case sAlbum:
- case sComment:
- case sTracknumber:
- case sGenre:
- case sVideoType:
- case sTimeCreate:
+ case sServiceref: return m_ref;
+ case sVideoHeight: return m_height;
+ case sVideoWidth: return m_width;
+ case sFrameRate: return m_framerate;
+ case sProgressive: return m_progressive;
+ case sAspect: return m_aspect;
+ case sTagTitle:
+ case sTagArtist:
+ case sTagAlbum:
+ case sTagTitleSortname:
+ case sTagArtistSortname:
+ case sTagAlbumSortname:
+ case sTagDate:
+ case sTagComposer:
+ case sTagGenre:
+ case sTagComment:
+ case sTagExtendedComment:
+ case sTagLocation:
+ case sTagHomepage:
+ case sTagDescription:
+ case sTagVersion:
+ case sTagISRC:
+ case sTagOrganization:
+ case sTagCopyright:
+ case sTagCopyrightURI:
+ case sTagContact:
+ case sTagLicense:
+ case sTagLicenseURI:
+ case sTagCodec:
+ case sTagAudioCodec:
+ case sTagVideoCodec:
+ case sTagEncoder:
+ case sTagLanguageCode:
+ case sTagKeywords:
+ case sTagChannelMode:
case sUser+12:
return resIsString;
- case sCurrentTitle:
+ case sTagTrackGain:
+ case sTagTrackPeak:
+ case sTagAlbumGain:
+ case sTagAlbumPeak:
+ case sTagReferenceLevel:
+ case sTagBeatsPerMinute:
+ case sTagImage:
+ case sTagPreviewImage:
+ case sTagAttachment:
+ return resIsPyObject;
+ case sTagTrackNumber:
tag = GST_TAG_TRACK_NUMBER;
break;
- case sTotalTitles:
+ case sTagTrackCount:
tag = GST_TAG_TRACK_COUNT;
break;
+ case sTagAlbumVolumeNumber:
+ tag = GST_TAG_ALBUM_VOLUME_NUMBER;
+ break;
+ case sTagAlbumVolumeCount:
+ tag = GST_TAG_ALBUM_VOLUME_COUNT;
+ break;
+ case sTagBitrate:
+ tag = GST_TAG_BITRATE;
+ break;
+ case sTagNominalBitrate:
+ tag = GST_TAG_NOMINAL_BITRATE;
+ break;
+ case sTagMinimumBitrate:
+ tag = GST_TAG_MINIMUM_BITRATE;
+ break;
+ case sTagMaximumBitrate:
+ tag = GST_TAG_MAXIMUM_BITRATE;
+ break;
+ case sTagSerial:
+ tag = GST_TAG_SERIAL;
+ break;
+ case sTagEncoderVersion:
+ tag = GST_TAG_ENCODER_VERSION;
+ break;
+ case sTagCRC:
+ tag = "has-crc";
+ break;
default:
return resNA;
}
guint value;
if (gst_tag_list_get_uint(m_stream_tags, tag, &value))
return (int) value;
-
- return 0;
+ return 0;
}
std::string eServiceMP3::getInfoString(int w)
{
- if ( !m_stream_tags )
+ if ( !m_stream_tags && w < sUser && w > 26 )
return "";
- gchar *tag = 0;
+ const gchar *tag = 0;
switch (w)
{
- case sTitle:
+ case sTagTitle:
tag = GST_TAG_TITLE;
break;
- case sArtist:
+ case sTagArtist:
tag = GST_TAG_ARTIST;
break;
- case sAlbum:
+ case sTagAlbum:
tag = GST_TAG_ALBUM;
break;
- case sComment:
- tag = GST_TAG_COMMENT;
- break;
- case sTracknumber:
- tag = GST_TAG_TRACK_NUMBER;
+ case sTagTitleSortname:
+ tag = GST_TAG_TITLE_SORTNAME;
break;
- case sGenre:
- tag = GST_TAG_GENRE;
+ case sTagArtistSortname:
+ tag = GST_TAG_ARTIST_SORTNAME;
break;
- case sVideoType:
- tag = GST_TAG_VIDEO_CODEC;
+ case sTagAlbumSortname:
+ tag = GST_TAG_ALBUM_SORTNAME;
break;
- case sTimeCreate:
+ case sTagDate:
GDate *date;
if (gst_tag_list_get_date(m_stream_tags, GST_TAG_DATE, &date))
{
gchar res[5];
- g_date_strftime (res, sizeof(res), "%Y", date);
+ g_date_strftime (res, sizeof(res), "%Y-%M-%D", date);
return (std::string)res;
}
break;
+ case sTagComposer:
+ tag = GST_TAG_COMPOSER;
+ break;
+ case sTagGenre:
+ tag = GST_TAG_GENRE;
+ break;
+ case sTagComment:
+ tag = GST_TAG_COMMENT;
+ break;
+ case sTagExtendedComment:
+ tag = GST_TAG_EXTENDED_COMMENT;
+ break;
+ case sTagLocation:
+ tag = GST_TAG_LOCATION;
+ break;
+ case sTagHomepage:
+ tag = GST_TAG_HOMEPAGE;
+ break;
+ case sTagDescription:
+ tag = GST_TAG_DESCRIPTION;
+ break;
+ case sTagVersion:
+ tag = GST_TAG_VERSION;
+ break;
+ case sTagISRC:
+ tag = GST_TAG_ISRC;
+ break;
+ case sTagOrganization:
+ tag = GST_TAG_ORGANIZATION;
+ break;
+ case sTagCopyright:
+ tag = GST_TAG_COPYRIGHT;
+ break;
+ case sTagCopyrightURI:
+ tag = GST_TAG_COPYRIGHT_URI;
+ break;
+ case sTagContact:
+ tag = GST_TAG_CONTACT;
+ break;
+ case sTagLicense:
+ tag = GST_TAG_LICENSE;
+ break;
+ case sTagLicenseURI:
+ tag = GST_TAG_LICENSE_URI;
+ break;
+ case sTagCodec:
+ tag = GST_TAG_CODEC;
+ break;
+ case sTagAudioCodec:
+ tag = GST_TAG_AUDIO_CODEC;
+ break;
+ case sTagVideoCodec:
+ tag = GST_TAG_VIDEO_CODEC;
+ break;
+ case sTagEncoder:
+ tag = GST_TAG_ENCODER;
+ break;
+ case sTagLanguageCode:
+ tag = GST_TAG_LANGUAGE_CODE;
+ break;
+ case sTagKeywords:
+ tag = GST_TAG_KEYWORDS;
+ break;
+ case sTagChannelMode:
+ tag = "channel-mode";
+ break;
case sUser+12:
return m_error_message;
default:
return "";
}
+PyObject *eServiceMP3::getInfoObject(int w)
+{
+ const gchar *tag = 0;
+ bool isBuffer = false;
+ switch (w)
+ {
+ case sTagTrackGain:
+ tag = GST_TAG_TRACK_GAIN;
+ break;
+ case sTagTrackPeak:
+ tag = GST_TAG_TRACK_PEAK;
+ break;
+ case sTagAlbumGain:
+ tag = GST_TAG_ALBUM_GAIN;
+ break;
+ case sTagAlbumPeak:
+ tag = GST_TAG_ALBUM_PEAK;
+ break;
+ case sTagReferenceLevel:
+ tag = GST_TAG_REFERENCE_LEVEL;
+ break;
+ case sTagBeatsPerMinute:
+ tag = GST_TAG_BEATS_PER_MINUTE;
+ break;
+ case sTagImage:
+ tag = GST_TAG_IMAGE;
+ isBuffer = true;
+ break;
+ case sTagPreviewImage:
+ tag = GST_TAG_PREVIEW_IMAGE;
+ isBuffer = true;
+ break;
+ case sTagAttachment:
+ tag = GST_TAG_ATTACHMENT;
+ isBuffer = true;
+ break;
+ default:
+ break;
+ }
+
+ if ( isBuffer )
+ {
+ const GValue *gv_buffer = gst_tag_list_get_value_index(m_stream_tags, tag, 0);
+ if ( gv_buffer )
+ {
+ GstBuffer *buffer;
+ buffer = gst_value_get_buffer (gv_buffer);
+ return PyBuffer_FromMemory(GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer));
+ }
+ }
+ else
+ {
+ gdouble value = 0.0;
+ gst_tag_list_get_double(m_stream_tags, tag, &value);
+ return PyFloat_FromDouble(value);
+ }
+
+ return 0;
+}
+
RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
{
ptr = this;
return 0;
}
+RESULT eServiceMP3::audioDelay(ePtr<iAudioDelay> &ptr)
+{
+ ptr = this;
+ return 0;
+}
+
int eServiceMP3::getNumberOfTracks()
{
return m_audioStreams.size();
int eServiceMP3::getCurrentTrack()
{
+ if (m_currentAudioStream == -1)
+ g_object_get (G_OBJECT (m_gst_playbin), "current-audio", &m_currentAudioStream, NULL);
return m_currentAudioStream;
}
RESULT eServiceMP3::selectTrack(unsigned int i)
{
- int ret = selectAudioStream(i);
- /* flush */
pts_t ppos;
getPlayPosition(ppos);
- seekTo(ppos);
+ ppos -= 90000;
+ if (ppos < 0)
+ ppos = 0;
+
+ int ret = selectAudioStream(i);
+ if (!ret) {
+ /* flush */
+ seekTo(ppos);
+ }
return ret;
}
int eServiceMP3::selectAudioStream(int i)
{
- gint nb_sources;
- GstPad *active_pad;
- GstElement *switch_audio = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
- if ( !switch_audio )
+ int current_audio;
+ g_object_set (G_OBJECT (m_gst_playbin), "current-audio", i, NULL);
+ g_object_get (G_OBJECT (m_gst_playbin), "current-audio", ¤t_audio, NULL);
+ if ( current_audio == i )
{
- eDebug("can't switch audio tracks! gst-plugin-selector needed");
- return -1;
+ eDebug ("eServiceMP3::switched to audio stream %i", current_audio);
+ m_currentAudioStream = i;
+ return 0;
}
- g_object_get (G_OBJECT (switch_audio), "n-pads", &nb_sources, NULL);
- if ( (unsigned int)i >= m_audioStreams.size() || i >= nb_sources || (unsigned int)m_currentAudioStream >= m_audioStreams.size() )
- return -2;
- char sinkpad[8];
- sprintf(sinkpad, "sink%d", i);
- g_object_set (G_OBJECT (switch_audio), "active-pad", gst_element_get_pad (switch_audio, sinkpad), NULL);
- g_object_get (G_OBJECT (switch_audio), "active-pad", &active_pad, NULL);
- gchar *name;
- name = gst_pad_get_name (active_pad);
- eDebug ("switched audio to (%s)", name);
- g_free(name);
- m_currentAudioStream = i;
- return 0;
+ return -1;
}
int eServiceMP3::getCurrentChannel()
RESULT eServiceMP3::getTrackInfo(struct iAudioTrackInfo &info, unsigned int i)
{
-// eDebug("eServiceMP3::getTrackInfo(&info, %i)",i);
if (i >= m_audioStreams.size())
return -2;
- if (m_audioStreams[i].type == atMPEG)
+ info.m_description = m_audioStreams[i].codec;
+/* if (m_audioStreams[i].type == atMPEG)
info.m_description = "MPEG";
else if (m_audioStreams[i].type == atMP3)
info.m_description = "MP3";
info.m_description = "PCM";
else if (m_audioStreams[i].type == atOGG)
info.m_description = "OGG";
+ else if (m_audioStreams[i].type == atFLAC)
+ info.m_description = "FLAC";
else
- info.m_description = "???";
+ info.m_description = "???";*/
if (info.m_language.empty())
info.m_language = m_audioStreams[i].language_code;
return 0;
if (gst_message_get_structure(msg))
{
gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
- eDebug("gst_message from %s: %s", sourceName, string);
+ eDebug("eServiceMP3::gst_message from %s: %s", sourceName, string);
g_free(string);
}
else
- eDebug("gst_message from %s: %s (without structure)", sourceName, GST_MESSAGE_TYPE_NAME(msg));
+ eDebug("eServiceMP3::gst_message from %s: %s (without structure)", sourceName, GST_MESSAGE_TYPE_NAME(msg));
#endif
switch (GST_MESSAGE_TYPE (msg))
{
- case GST_MESSAGE_EOS:
- m_event((iPlayableService*)this, evEOF);
- break;
- case GST_MESSAGE_ERROR:
- {
- gchar *debug;
- GError *err;
+ 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;
- 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 )
+ 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 *sink;
+ g_object_get (G_OBJECT (m_gst_playbin), "text-sink", &sink, NULL);
+ if (sink)
+ {
+ g_object_set (G_OBJECT (sink), "max-buffers", 2, NULL);
+ g_object_set (G_OBJECT (sink), "sync", FALSE, NULL);
+ g_object_set (G_OBJECT (sink), "async", FALSE, NULL);
+ g_object_set (G_OBJECT (sink), "emit-signals", TRUE, NULL);
+ gst_object_unref(sink);
+ }
+ setAC3Delay(ac3_delay);
+ setPCMDelay(pcm_delay);
+ } 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_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) 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;
}
- 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:
+ {
+ 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:
{
- if (m_stream_tags)
- gst_tag_list_free(m_stream_tags);
- m_stream_tags = result;
+ 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;
}
-
- gchar *g_audiocodec;
- if ( gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec) && m_audioStreams.size() == 0 )
+ case GST_MESSAGE_ASYNC_DONE:
{
- 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 )
+ if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_gst_playbin))
break;
- audioStream audio;
- audio.type = gstCheckAudioPad(str);
- m_audioStreams.push_back(audio);
- }
- 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)
- {
- if ( IterAudioStream->pad )
+ 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++)
{
- 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) )
+ 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) )
{
- eDebug("found audio language %s",g_language);
- IterAudioStream->language_code = std::string(g_language);
- g_free (g_language);
+ 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_lang;
+// gchar *g_type;
+// GstPad* pad = 0;
+// g_signal_emit_by_name (m_gst_playbin, "get-text-pad", i, &pad);
+// GstCaps* caps = gst_pad_get_negotiated_caps(pad);
+// GstStructure* str = gst_caps_get_structure(caps, 0);
+// g_type = gst_structure_get_name(str);
+// g_signal_emit_by_name (m_gst_playbin, "get-text-tags", i, &tags);
+ subtitleStream subs;
+ subs.type = stPlainText;
+ 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);
+ eDebug("eServiceMP3::subtitle stream=%i language=%s"/* type=%s*/, i, g_lang/*, g_type*/);
+ m_subtitleStreams.push_back(subs);
+ g_free (g_lang);
+// g_free (g_type);
+ }
+ m_event((iPlayableService*)this, evUpdatedEventInfo);
}
- for (std::vector<subtitleStream>::iterator IterSubtitleStream(m_subtitleStreams.begin()); IterSubtitleStream != m_subtitleStreams.end(); ++IterSubtitleStream)
+ case GST_MESSAGE_ELEMENT:
{
- if ( IterSubtitleStream->pad )
+ if ( gst_is_missing_plugin_message(msg) )
{
- 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) )
+ gchar *description = gst_missing_plugin_message_get_description(msg);
+ if ( description )
{
- eDebug("found subtitle language %s",g_language);
- IterSubtitleStream->language_code = std::string(g_language);
- g_free (g_language);
+ m_error_message = "GStreamer plugin " + (std::string)description + " not available!\n";
+ g_free(description);
+ m_event((iPlayableService*)this, evUser+12);
}
}
- }
- }
- case GST_MESSAGE_ELEMENT:
- {
- if ( gst_is_missing_plugin_message(msg) )
- {
- gchar *description = gst_missing_plugin_message_get_description(msg);
- if ( description )
+ else if (const GstStructure *msgstruct = gst_message_get_structure(msg))
{
- m_error_message = "GStreamer plugin " + (std::string)description + " not available!\n";
- g_free(description);
- m_event((iPlayableService*)this, evUser+12);
+ 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;
}
- }
- default:
- 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);
+ }
+ default:
+ break;
}
g_free (sourceName);
}
audiotype_t eServiceMP3::gstCheckAudioPad(GstStructure* structure)
{
- const gchar* type;
- type = gst_structure_get_name(structure);
+ if (!structure)
+ return atUnknown;
+
+ if ( gst_structure_has_name (structure, "audio/mpeg"))
+ {
+ gint mpegversion, layer = -1;
+ if (!gst_structure_get_int (structure, "mpegversion", &mpegversion))
+ return atUnknown;
- if (!strcmp(type, "audio/mpeg")) {
- gint mpegversion, layer = 0;
- gst_structure_get_int (structure, "mpegversion", &mpegversion);
- gst_structure_get_int (structure, "layer", &layer);
- eDebug("mime audio/mpeg version %d layer %d", mpegversion, layer);
- switch (mpegversion) {
- case 1:
+ switch (mpegversion) {
+ case 1:
{
+ gst_structure_get_int (structure, "layer", &layer);
if ( layer == 3 )
return atMP3;
else
return atMPEG;
+ break;
}
- case 2:
- return atMPEG;
- case 4:
- return atAAC;
- default:
- return atUnknown;
- }
+ case 2:
+ return atAAC;
+ case 4:
+ return atAAC;
+ default:
+ return atUnknown;
}
- else
- {
- eDebug("mime %s", type);
- if (!strcmp(type, "audio/x-ac3") || !strcmp(type, "audio/ac3"))
- return atAC3;
- else if (!strcmp(type, "audio/x-dts") || !strcmp(type, "audio/dts"))
- return atDTS;
- else if (!strcmp(type, "audio/x-raw-int"))
- return atPCM;
}
+
+ else if ( gst_structure_has_name (structure, "audio/x-ac3") || gst_structure_has_name (structure, "audio/ac3") )
+ return atAC3;
+ else if ( gst_structure_has_name (structure, "audio/x-dts") || gst_structure_has_name (structure, "audio/dts") )
+ return atDTS;
+ else if ( gst_structure_has_name (structure, "audio/x-raw-int") )
+ return atPCM;
+
return atUnknown;
}
-void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
+void eServiceMP3::gstPoll(const int &msg)
{
- const gchar* type;
- GstCaps* caps;
- GstStructure* str;
- caps = gst_pad_get_caps(pad);
- str = gst_caps_get_structure(caps, 0);
- type = gst_structure_get_name(str);
-
- eDebug("A new pad %s:%s was created", GST_OBJECT_NAME (decodebin), GST_OBJECT_NAME (pad));
-
- eServiceMP3 *_this = (eServiceMP3*)user_data;
- GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
- if (g_strrstr(type,"audio"))
+ /* ok, we have a serious problem here. gstBusSyncHandler sends
+ us the wakup signal, but likely before it was posted.
+ the usleep, an EVIL HACK (DON'T DO THAT!!!) works around this.
+
+ I need to understand the API a bit more to make this work
+ proplerly. */
+ if (msg == 1)
{
- audioStream audio;
- audio.type = _this->gstCheckAudioPad(str);
- GstElement *switch_audio = gst_bin_get_by_name(pipeline , "switch_audio");
- if ( switch_audio )
- {
- GstPad *sinkpad = gst_element_get_request_pad (switch_audio, "sink%d");
- gst_pad_link(pad, sinkpad);
- audio.pad = sinkpad;
- _this->m_audioStreams.push_back(audio);
-
- if ( _this->m_audioStreams.size() == 1 )
- {
- _this->selectAudioStream(0);
- gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING);
- }
- else
- g_object_set (G_OBJECT (switch_audio), "select-all", FALSE, NULL);
- }
- else
+ GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_playbin));
+ GstMessage *message;
+ usleep(1);
+ while ((message = gst_bus_pop (bus)))
{
- gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink"));
- _this->m_audioStreams.push_back(audio);
+ gstBusCall(bus, message);
+ gst_message_unref (message);
}
}
- if (g_strrstr(type,"video"))
- {
- gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_video"), "sink"));
- }
- if (g_strrstr(type,"application/x-ssa") || g_strrstr(type,"application/x-ass"))
- {
- GstPad *switchpad = _this->gstCreateSubtitleSink(_this, stSSA);
- gst_pad_link(pad, switchpad);
- subtitleStream subs;
- subs.pad = switchpad;
- subs.type = stSSA;
- _this->m_subtitleStreams.push_back(subs);
- }
- if (g_strrstr(type,"text/plain"))
- {
- GstPad *switchpad = _this->gstCreateSubtitleSink(_this, stPlainText);
- gst_pad_link(pad, switchpad);
- subtitleStream subs;
- subs.pad = switchpad;
- subs.type = stPlainText;
- _this->m_subtitleStreams.push_back(subs);
- }
-}
-
-GstPad* eServiceMP3::gstCreateSubtitleSink(eServiceMP3* _this, subtype_t type)
-{
- GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
- GstElement *switch_subparse = gst_bin_get_by_name(pipeline,"switch_subparse");
- if ( !switch_subparse )
- {
- switch_subparse = gst_element_factory_make ("input-selector", "switch_subparse");
- GstElement *sink = gst_element_factory_make("fakesink", "sink_subtitles");
- gst_bin_add_many(pipeline, switch_subparse, sink, NULL);
- gst_element_link(switch_subparse, sink);
- g_object_set (G_OBJECT(sink), "signal-handoffs", TRUE, NULL);
- g_object_set (G_OBJECT(sink), "sync", TRUE, NULL);
- g_object_set (G_OBJECT(sink), "async", FALSE, NULL);
- g_signal_connect(sink, "handoff", G_CALLBACK(_this->gstCBsubtitleAvail), _this);
-
- // order is essential since requested sink pad names can't be explicitely chosen
- GstElement *switch_substream_plain = gst_element_factory_make ("input-selector", "switch_substream_plain");
- gst_bin_add(pipeline, switch_substream_plain);
- GstPad *sinkpad_plain = gst_element_get_request_pad (switch_subparse, "sink%d");
- gst_pad_link(gst_element_get_pad (switch_substream_plain, "src"), sinkpad_plain);
-
- GstElement *switch_substream_ssa = gst_element_factory_make ("input-selector", "switch_substream_ssa");
- GstElement *ssaparse = gst_element_factory_make("ssaparse", "ssaparse");
- gst_bin_add_many(pipeline, switch_substream_ssa, ssaparse, NULL);
- GstPad *sinkpad_ssa = gst_element_get_request_pad (switch_subparse, "sink%d");
- gst_element_link(switch_substream_ssa, ssaparse);
- gst_pad_link(gst_element_get_pad (ssaparse, "src"), sinkpad_ssa);
-
- GstElement *switch_substream_srt = gst_element_factory_make ("input-selector", "switch_substream_srt");
- GstElement *srtparse = gst_element_factory_make("subparse", "srtparse");
- gst_bin_add_many(pipeline, switch_substream_srt, srtparse, NULL);
- GstPad *sinkpad_srt = gst_element_get_request_pad (switch_subparse, "sink%d");
- gst_element_link(switch_substream_srt, srtparse);
- gst_pad_link(gst_element_get_pad (srtparse, "src"), sinkpad_srt);
- g_object_set (G_OBJECT(srtparse), "subtitle-encoding", "ISO-8859-15", NULL);
- }
-
- switch (type)
- {
- case stSSA:
- return gst_element_get_request_pad (gst_bin_get_by_name(pipeline,"switch_substream_ssa"), "sink%d");
- case stSRT:
- return gst_element_get_request_pad (gst_bin_get_by_name(pipeline,"switch_substream_srt"), "sink%d");
- case stPlainText:
- default:
- break;
- }
- return gst_element_get_request_pad (gst_bin_get_by_name(pipeline,"switch_substream_plain"), "sink%d");
+ else
+ pullSubtitle();
}
-void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
-{
- eServiceMP3 *_this = (eServiceMP3*)user_data;
- GstElement *decoder = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"decoder");
- gst_pad_link(pad, gst_element_get_static_pad (decoder, "sink"));
-}
+eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
-void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)
+void eServiceMP3::gstCBsubtitleAvail(GstElement *appsink, gpointer user_data)
{
eServiceMP3 *_this = (eServiceMP3*)user_data;
- GstCaps *caps;
- GstStructure *str;
- GstPad *audiopad;
-
- /* only link once */
- GstElement *audiobin = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"audiobin");
- audiopad = gst_element_get_static_pad (audiobin, "sink");
- if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
- eDebug("audio already linked!");
- g_object_unref (audiopad);
- return;
- }
-
- /* check media type */
- caps = gst_pad_get_caps (pad);
- str = gst_caps_get_structure (caps, 0);
- eDebug("gst new pad! %s", gst_structure_get_name (str));
-
- if (!g_strrstr (gst_structure_get_name (str), "audio")) {
- gst_caps_unref (caps);
- gst_object_unref (audiopad);
- return;
- }
-
- gst_caps_unref (caps);
- gst_pad_link (pad, audiopad);
+ eSingleLocker l(_this->m_subs_to_pull_lock);
+ ++_this->m_subs_to_pull;
+ _this->m_pump.send(2);
}
-void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
+void eServiceMP3::pullSubtitle()
{
- GstStructure *str;
-
- /* check media type */
- caps = gst_pad_get_caps (pad);
- str = gst_caps_get_structure (caps, 0);
- eDebug("unknown type: %s - this can't be decoded.", gst_structure_get_name (str));
- gst_caps_unref (caps);
-}
-
-void eServiceMP3::gstPoll(const int&)
-{
- /* ok, we have a serious problem here. gstBusSyncHandler sends
- us the wakup signal, but likely before it was posted.
- the usleep, an EVIL HACK (DON'T DO THAT!!!) works around this.
-
- I need to understand the API a bit more to make this work
- proplerly. */
- usleep(1);
-
- GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline));
- GstMessage *message;
- while ((message = gst_bus_pop (bus)))
+ GstElement *sink;
+ g_object_get (G_OBJECT (m_gst_playbin), "text-sink", &sink, NULL);
+ if (sink)
{
- gstBusCall(bus, message);
- gst_message_unref (message);
+ while (m_subs_to_pull && m_subtitle_pages.size() < 2)
+ {
+ GstBuffer *buffer;
+ {
+ eSingleLocker l(m_subs_to_pull_lock);
+ --m_subs_to_pull;
+ g_signal_emit_by_name (sink, "pull-buffer", &buffer);
+ }
+ if (buffer)
+ {
+ gint64 buf_pos = GST_BUFFER_TIMESTAMP(buffer);
+ gint64 duration_ns = GST_BUFFER_DURATION(buffer);
+ size_t len = GST_BUFFER_SIZE(buffer);
+ unsigned char line[len+1];
+ memcpy(line, GST_BUFFER_DATA(buffer), len);
+ line[len] = 0;
+ eDebug("got new subtitle @ buf_pos = %lld ns (in pts=%lld): '%s' ", buf_pos, buf_pos/11111, line);
+ ePangoSubtitlePage page;
+ gRGB rgbcol(0xD0,0xD0,0xD0);
+ page.m_elements.push_back(ePangoSubtitlePageElement(rgbcol, (const char*)line));
+ page.show_pts = buf_pos / 11111L;
+ page.m_timeout = duration_ns / 1000000;
+ m_subtitle_pages.push_back(page);
+ pushSubtitles();
+ gst_buffer_unref(buffer);
+ }
+ }
+ gst_object_unref(sink);
}
+ else
+ eDebug("no subtitle sink!");
}
-eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
-
-void eServiceMP3::gstCBsubtitleAvail(GstElement *element, GstBuffer *buffer, GstPad *pad, gpointer user_data)
+void eServiceMP3::pushSubtitles()
{
- gint64 duration_ns = GST_BUFFER_DURATION(buffer);
- const unsigned char *text = (unsigned char *)GST_BUFFER_DATA(buffer);
- eDebug("gstCBsubtitleAvail: %s",text);
- eServiceMP3 *_this = (eServiceMP3*)user_data;
- if ( _this->m_subtitle_widget )
+ ePangoSubtitlePage page;
+ pts_t running_pts;
+ while ( !m_subtitle_pages.empty() )
{
- ePangoSubtitlePage page;
- gRGB rgbcol(0xD0,0xD0,0xD0);
- page.m_elements.push_back(ePangoSubtitlePageElement(rgbcol, (const char*)text));
- page.m_timeout = duration_ns / 1000000;
- (_this->m_subtitle_widget)->setPage(page);
+ getPlayPosition(running_pts);
+ page = m_subtitle_pages.front();
+ gint64 diff_ms = ( page.show_pts - running_pts ) / 90;
+ eDebug("eServiceMP3::pushSubtitles show_pts = %lld running_pts = %lld diff = %lld", page.show_pts, running_pts, diff_ms);
+ if (diff_ms < -100)
+ {
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint64 now;
+ if (gst_element_query_position(m_gst_playbin, &fmt, &now) != -1)
+ {
+ now /= 11111;
+ diff_ms = abs((now - running_pts) / 90);
+ eDebug("diff < -100ms check decoder/pipeline diff: decoder: %lld, pipeline: %lld, diff: %lld", running_pts, now, diff_ms);
+ if (diff_ms > 100000)
+ {
+ eDebug("high decoder/pipeline difference.. assume decoder has now started yet.. check again in 1sec");
+ m_subtitle_sync_timer->start(1000, true);
+ break;
+ }
+ }
+ else
+ eDebug("query position for decoder/pipeline check failed!");
+ eDebug("subtitle to late... drop");
+ m_subtitle_pages.pop_front();
+ }
+ else if ( diff_ms > 20 )
+ {
+// eDebug("start recheck timer");
+ m_subtitle_sync_timer->start(diff_ms > 1000 ? 1000 : diff_ms, true);
+ break;
+ }
+ else // immediate show
+ {
+ if (m_subtitle_widget)
+ m_subtitle_widget->setPage(page);
+ m_subtitle_pages.pop_front();
+ }
}
+ if (m_subtitle_pages.empty())
+ pullSubtitle();
}
RESULT eServiceMP3::enableSubtitles(eWidget *parent, ePyObject tuple)
{
ePyObject entry;
int tuplesize = PyTuple_Size(tuple);
- int pid;
- int type;
- gint nb_sources;
- GstPad *active_pad;
- GstElement *switch_substream = NULL;
- GstElement *switch_subparse = gst_bin_get_by_name (GST_BIN(m_gst_pipeline), "switch_subparse");
+ int pid, type;
+ gint text_pid = 0;
if (!PyTuple_Check(tuple))
goto error_out;
goto error_out;
type = PyInt_AsLong(entry);
- switch ((subtype_t)type)
+ if (m_currentSubtitleStream != pid)
{
- case stPlainText:
- switch_substream = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_substream_plain");
- break;
- case stSSA:
- switch_substream = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_substream_ssa");
- break;
- case stSRT:
- switch_substream = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_substream_srt");
- break;
- default:
- goto error_out;
+ eSingleLocker l(m_subs_to_pull_lock);
+ g_object_set (G_OBJECT (m_gst_playbin), "current-text", pid, NULL);
+ m_currentSubtitleStream = pid;
+ m_subs_to_pull = 0;
+ m_subtitle_pages.clear();
}
+ m_subtitle_widget = 0;
m_subtitle_widget = new eSubtitleWidget(parent);
m_subtitle_widget->resize(parent->size()); /* full size */
- if ( !switch_substream )
- {
- eDebug("can't switch subtitle tracks! gst-plugin-selector needed");
- return -2;
- }
- g_object_get (G_OBJECT (switch_substream), "n-pads", &nb_sources, NULL);
- if ( (unsigned int)pid >= m_subtitleStreams.size() || pid >= nb_sources || (unsigned int)m_currentSubtitleStream >= m_subtitleStreams.size() )
- return -2;
- g_object_get (G_OBJECT (switch_subparse), "n-pads", &nb_sources, NULL);
- if ( type < 0 || type >= nb_sources )
- return -2;
+ g_object_get (G_OBJECT (m_gst_playbin), "current-text", &text_pid, NULL);
- char sinkpad[6];
- sprintf(sinkpad, "sink%d", type);
- g_object_set (G_OBJECT (switch_subparse), "active-pad", gst_element_get_pad (switch_subparse, sinkpad), NULL);
- sprintf(sinkpad, "sink%d", pid);
- g_object_set (G_OBJECT (switch_substream), "active-pad", gst_element_get_pad (switch_substream, sinkpad), NULL);
- m_currentSubtitleStream = pid;
+ eDebug ("eServiceMP3::switched to subtitle stream %i", text_pid);
return 0;
+
error_out:
- eDebug("enableSubtitles needs a tuple as 2nd argument!\n"
+ eDebug("eServiceMP3::enableSubtitles needs a tuple as 2nd argument!\n"
"for gst subtitles (2, subtitle_stream_count, subtitle_type)");
return -1;
}
RESULT eServiceMP3::disableSubtitles(eWidget *parent)
{
eDebug("eServiceMP3::disableSubtitles");
+ m_subtitle_pages.clear();
delete m_subtitle_widget;
m_subtitle_widget = 0;
return 0;
PyObject *eServiceMP3::getCachedSubtitle()
{
- eDebug("eServiceMP3::getCachedSubtitle");
+// eDebug("eServiceMP3::getCachedSubtitle");
Py_RETURN_NONE;
}
return l;
}
+RESULT eServiceMP3::streamed(ePtr<iStreamedService> &ptr)
+{
+ ptr = this;
+ return 0;
+}
+
+PyObject *eServiceMP3::getBufferCharge()
+{
+ ePyObject tuple = PyTuple_New(5);
+ PyTuple_SET_ITEM(tuple, 0, PyInt_FromLong(m_bufferInfo.bufferPercent));
+ PyTuple_SET_ITEM(tuple, 1, PyInt_FromLong(m_bufferInfo.avgInRate));
+ PyTuple_SET_ITEM(tuple, 2, PyInt_FromLong(m_bufferInfo.avgOutRate));
+ PyTuple_SET_ITEM(tuple, 3, PyInt_FromLong(m_bufferInfo.bufferingLeft));
+ PyTuple_SET_ITEM(tuple, 4, PyInt_FromLong(m_buffer_size));
+ return tuple;
+}
+
+int eServiceMP3::setBufferSize(int size)
+{
+ m_buffer_size = size;
+ g_object_set (G_OBJECT (m_gst_playbin), "buffer-size", m_buffer_size, NULL);
+ return 0;
+}
+
+int eServiceMP3::getAC3Delay()
+{
+ return ac3_delay;
+}
+
+int eServiceMP3::getPCMDelay()
+{
+ return pcm_delay;
+}
+
+void eServiceMP3::setAC3Delay(int delay)
+{
+ ac3_delay = delay;
+ if (!m_gst_playbin || m_state != stRunning)
+ return;
+ else
+ {
+ GstElement *sink;
+ std::string config_delay;
+ int config_delay_int = delay;
+ if(ePythonConfigQuery::getConfigValue("config.av.generalAC3delay", config_delay) == 0)
+ config_delay_int += atoi(config_delay.c_str());
+
+ g_object_get (G_OBJECT (m_gst_playbin), "audio-sink", &sink, NULL);
+
+ if (!sink)
+ return;
+ else {
+ gchar *name = gst_element_get_name(sink);
+
+ if (strstr(name, "dvbaudiosink"))
+ eTSMPEGDecoder::setHwAC3Delay(config_delay_int);
+ g_free(name);
+ gst_object_unref(sink);
+ }
+ }
+}
+
+void eServiceMP3::setPCMDelay(int delay)
+{
+ pcm_delay = delay;
+ if (!m_gst_playbin || m_state != stRunning)
+ return;
+ else
+ {
+ GstElement *sink;
+ std::string config_delay;
+ int config_delay_int = delay;
+ if(ePythonConfigQuery::getConfigValue("config.av.generalPCMdelay", config_delay) == 0)
+ config_delay_int += atoi(config_delay.c_str());
+
+ g_object_get (G_OBJECT (m_gst_playbin), "audio-sink", &sink, NULL);
+
+ if (!sink)
+ return;
+ else {
+ gchar *name = gst_element_get_name(sink);
+
+ if (strstr(name, "dvbaudiosink"))
+ eTSMPEGDecoder::setHwPCMDelay(config_delay_int);
+ else {
+ // this is realy untested..and not used yet
+ gint64 offset = config_delay_int;
+ offset *= 1000000; // milli to nano
+ g_object_set (G_OBJECT (m_gst_playbin), "ts-offset", offset, NULL);
+ }
+ g_free(name);
+ gst_object_unref(sink);
+ }
+ }
+}
+
#else
#warning gstreamer not available, not building media player
#endif
git.cweiske.de / enigma2.git / blobdiff