+#ifdef HAVE_GSTREAMER
+
+ /* 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/eerror.h>
#include <lib/base/object.h>
#include <lib/base/ebase.h>
#include <string>
#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 <gst/gst.h>
+#include <sys/stat.h>
+/* for subtitles */
+#include <lib/gui/esubtitle.h>
// eServiceFactoryMP3
{
ePtr<eServiceCenter> sc;
- eServiceCenter::getInstance(sc);
+ eServiceCenter::getPrivInstance(sc);
if (sc)
- sc->addServiceFactory(eServiceFactoryMP3::id, this);
+ {
+ std::list<std::string> extensions;
+ extensions.push_back("mp3");
+ extensions.push_back("ogg");
+ extensions.push_back("mpg");
+ extensions.push_back("vob");
+ extensions.push_back("wav");
+ extensions.push_back("wave");
+ extensions.push_back("mkv");
+ extensions.push_back("avi");
+ sc->addServiceFactory(eServiceFactoryMP3::id, this, extensions);
+ }
m_service_info = new eStaticServiceMP3Info();
}
{
ePtr<eServiceCenter> sc;
- eServiceCenter::getInstance(sc);
+ eServiceCenter::getPrivInstance(sc);
if (sc)
sc->removeServiceFactory(eServiceFactoryMP3::id);
}
return 0;
}
-RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &, ePtr<iServiceOfflineOperations> &ptr)
+class eMP3ServiceOfflineOperations: public iServiceOfflineOperations
{
- ptr = 0;
- return -1;
+ DECLARE_REF(eMP3ServiceOfflineOperations);
+ eServiceReference m_ref;
+public:
+ eMP3ServiceOfflineOperations(const eServiceReference &ref);
+
+ RESULT deleteFromDisk(int simulate);
+ RESULT getListOfFilenames(std::list<std::string> &);
+};
+
+DEFINE_REF(eMP3ServiceOfflineOperations);
+
+eMP3ServiceOfflineOperations::eMP3ServiceOfflineOperations(const eServiceReference &ref): m_ref((const eServiceReference&)ref)
+{
+}
+
+RESULT eMP3ServiceOfflineOperations::deleteFromDisk(int simulate)
+{
+ if (simulate)
+ return 0;
+ else
+ {
+ std::list<std::string> res;
+ if (getListOfFilenames(res))
+ return -1;
+
+ eBackgroundFileEraser *eraser = eBackgroundFileEraser::getInstance();
+ if (!eraser)
+ eDebug("FATAL !! can't get background file eraser");
+
+ for (std::list<std::string>::iterator i(res.begin()); i != res.end(); ++i)
+ {
+ eDebug("Removing %s...", i->c_str());
+ if (eraser)
+ eraser->erase(i->c_str());
+ else
+ ::unlink(i->c_str());
+ }
+
+ return 0;
+ }
+}
+
+RESULT eMP3ServiceOfflineOperations::getListOfFilenames(std::list<std::string> &res)
+{
+ res.clear();
+ res.push_back(m_ref.path);
+ return 0;
}
+RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &ref, ePtr<iServiceOfflineOperations> &ptr)
+{
+ ptr = new eMP3ServiceOfflineOperations(ref);
+ return 0;
+}
+
// eStaticServiceMP3Info
RESULT eStaticServiceMP3Info::getName(const eServiceReference &ref, std::string &name)
{
- name = "MP3 file: " + 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
-void eServiceMP3::test_end()
-{
- eDebug("end of mp3!");
- stop();
-}
-
-eServiceMP3::eServiceMP3(const char *filename): filename(filename), test(eApp)
+eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
{
+ 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_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_state = stIdle;
eDebug("SERVICEMP3 construct!");
+
+ /* FIXME: currently, decodebin isn't possible for
+ video streams. in that case, make a manual pipeline. */
+
+ const char *ext = strrchr(filename, '.');
+ if (!ext)
+ ext = filename;
+
+ int is_mpeg_ps = !(strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin") && strcasecmp(ext, ".dat"));
+ int is_mpeg_ts = !strcasecmp(ext, ".ts");
+ int is_matroska = !strcasecmp(ext, ".mkv");
+ int is_avi = !strcasecmp(ext, ".avi");
+ int is_mp3 = !strcasecmp(ext, ".mp3"); /* force mp3 instead of decodebin */
+ int is_video = is_mpeg_ps || is_mpeg_ts || is_matroska || is_avi;
+ int is_streaming = !strncmp(filename, "http://", 7);
+ int is_AudioCD = !(strncmp(filename, "/autofs/", 8) || strncmp(filename+strlen(filename)-13, "/track-", 7) || strcasecmp(ext, ".wav"));
+
+ eDebug("filename: %s, is_mpeg_ps: %d, is_mpeg_ts: %d, is_video: %d, is_streaming: %d, is_mp3: %d, is_matroska: %d, is_avi: %d, is_AudioCD: %d", filename, is_mpeg_ps, is_mpeg_ts, is_video, is_streaming, is_mp3, is_matroska, is_avi, is_AudioCD);
+
+ int is_audio = !is_video;
+
+ int all_ok = 0;
+
+ m_gst_pipeline = gst_pipeline_new ("mediaplayer");
+ if (!m_gst_pipeline)
+ eWarning("failed to create pipeline");
+
+ if (is_AudioCD)
+ {
+ source = gst_element_factory_make ("cdiocddasrc", "cda-source");
+ if (source)
+ g_object_set (G_OBJECT (source), "device", "/dev/cdroms/cdrom0", NULL);
+ else
+ is_AudioCD = 0;
+ }
+ if ( !is_streaming && !is_AudioCD )
+ source = gst_element_factory_make ("filesrc", "file-source");
+ else if ( is_streaming )
+ {
+ source = gst_element_factory_make ("neonhttpsrc", "http-source");
+ if (source)
+ g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
+ }
+
+ if (!source)
+ eWarning("failed to create %s", is_streaming ? "neonhttpsrc" : "filesrc");
+ /* configure source */
+ else if (!is_AudioCD)
+ g_object_set (G_OBJECT (source), "location", filename, NULL);
+ 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 (is_audio)
+ {
+ /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
+ const char *decodertype = "decodebin";
+
+ decoder = gst_element_factory_make (decodertype, "decoder");
+ if (!decoder)
+ eWarning("failed to create %s decoder", decodertype);
+
+ conv = gst_element_factory_make ("audioconvert", "converter");
+ if (!conv)
+ eWarning("failed to create audioconvert");
+
+ flt = gst_element_factory_make ("capsfilter", "flt");
+ if (!flt)
+ eWarning("failed to create capsfilter");
+
+ /* 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, */(char*)0);
+ g_object_set (G_OBJECT (flt), "caps", caps, (char*)0);
+ gst_caps_unref(caps);
+ }
+
+ sink = gst_element_factory_make ("alsasink", "alsa-output");
+ if (!sink)
+ eWarning("failed to create osssink");
+
+ if (source && decoder && conv && sink)
+ all_ok = 1;
+ } else /* is_video */
+ {
+ /* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
+ | queue_video -> dvbvideosink */
+
+ audio = gst_element_factory_make("dvbaudiosink", "audiosink");
+ queue_audio = gst_element_factory_make("queue", "queue_audio");
+
+ video = gst_element_factory_make("dvbvideosink", "videosink");
+ queue_video = gst_element_factory_make("queue", "queue_video");
+
+ if (is_mpeg_ps)
+ videodemux = gst_element_factory_make("flupsdemux", "videodemux");
+ else if (is_mpeg_ts)
+ videodemux = gst_element_factory_make("flutsdemux", "videodemux");
+ else if (is_matroska)
+ videodemux = gst_element_factory_make("matroskademux", "videodemux");
+ else if (is_avi)
+ videodemux = gst_element_factory_make("avidemux", "videodemux");
+
+ if (!videodemux)
+ {
+ eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
+ videodemux = gst_element_factory_make("mpegdemux", "videodemux");
+ }
+
+ eDebug("audio: %p, queue_audio %p, video %p, queue_video %p, videodemux %p", audio, queue_audio, video, queue_video, videodemux);
+ if (audio && queue_audio && video && queue_video && videodemux)
+ {
+ 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);
+ all_ok = 1;
+ }
+ }
+
+ if (m_gst_pipeline && all_ok)
+ {
+ gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
+
+ if (is_AudioCD)
+ {
+ 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 (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);
+
+ if (!is_mp3)
+ 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, (char*)0);
+ gst_element_link_many(conv, flt, sink, (char*)0);
+ gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));
+ gst_object_unref(audiopad);
+ gst_bin_add (GST_BIN(m_gst_pipeline), audio);
+ /* in mad's case, we can directly connect the decoder to the audiobin. otherwise, we do this in gstCBnewPad */
+ if (is_mp3)
+ gst_element_link(decoder, audio);
+
+ } else /* 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);
+ GstElement *parser = gst_element_factory_make("subparse", "parse_subtitles");
+ GstElement *switch_subtitles = gst_element_factory_make ("input-selector", "switch_subtitles");
+ GstElement *sink = gst_element_factory_make("fakesink", "sink_subtitles");
+ gst_bin_add_many(GST_BIN (m_gst_pipeline), subsource, switch_subtitles, parser, sink, NULL);
+ gst_element_link(subsource, switch_subtitles);
+ gst_element_link(switch_subtitles, parser);
+ gst_element_link(parser, sink);
+ g_object_set (G_OBJECT(switch_subtitles), "select-all", TRUE, NULL);
+ g_object_set (G_OBJECT(sink), "signal-handoffs", TRUE, NULL);
+ g_object_set (G_OBJECT(sink), "sync", TRUE, NULL);
+ g_signal_connect(sink, "handoff", G_CALLBACK(gstCBsubtitleAvail), this);
+ subtitleStream subs;
+ subs.language_code = std::string(".srt file");
+ m_subtitleStreams.push_back(subs);
+ }
+ gst_bin_add_many(GST_BIN(m_gst_pipeline), source, videodemux, audio, queue_audio, video, queue_video, NULL);
+ switch_audio = gst_element_factory_make ("input-selector", "switch_audio");
+ if (switch_audio)
+ {
+ g_object_set (G_OBJECT (switch_audio), "select-all", TRUE, NULL);
+ gst_bin_add(GST_BIN(m_gst_pipeline), switch_audio);
+ gst_element_link(switch_audio, queue_audio);
+ }
+ gst_element_link(source, videodemux);
+ gst_element_link(queue_audio, audio);
+ gst_element_link(queue_video, video);
+ g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+ }
+ } else
+ {
+ 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.");
+ m_gst_pipeline = 0;
+ }
+
+ gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
}
eServiceMP3::~eServiceMP3()
{
- eDebug("SERVICEMP3 destruct!");
+ delete m_subtitle_widget;
if (m_state == stRunning)
stop();
+
+ if (m_stream_tags)
+ gst_tag_list_free(m_stream_tags);
+
+ if (m_gst_pipeline)
+ {
+ gst_object_unref (GST_OBJECT (m_gst_pipeline));
+ eDebug("SERVICEMP3 destruct!");
+ }
}
DEFINE_REF(eServiceMP3);
assert(m_state == stIdle);
m_state = stRunning;
-
- printf("mp3 starts\n");
- printf("MP3: %s start\n", filename.c_str());
- test.start(1000, 1);
- CONNECT(test.timeout, eServiceMP3::test_end);
+ if (m_gst_pipeline)
+ {
+ eDebug("starting pipeline");
+ gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
+ }
m_event(this, evStart);
return 0;
}
assert(m_state != stIdle);
if (m_state == stStopped)
return -1;
- test.stop();
- printf("MP3: %s stop\n", filename.c_str());
+ eDebug("MP3: %s stop\n", m_filename.c_str());
+ gst_element_set_state(m_gst_pipeline, GST_STATE_NULL);
m_state = stStopped;
- m_event(this, evEnd);
return 0;
}
-RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr) { ptr=this; return 0; }
-RESULT eServiceMP3::seek(ePtr<iSeekableService> &ptr) { ptr = 0; return -1; }
+RESULT eServiceMP3::setTarget(int target)
+{
+ return -1;
+}
+
+RESULT eServiceMP3::pause(ePtr<iPauseableService> &ptr)
+{
+ ptr=this;
+ return 0;
+}
+
+RESULT eServiceMP3::setSlowMotion(int ratio)
+{
+ /* we can't do slomo yet */
+ return -1;
+}
+
+RESULT eServiceMP3::setFastForward(int ratio)
+{
+ m_currentTrickRatio = ratio;
+ if (ratio)
+ m_seekTimeout.start(1000, 0);
+ else
+ m_seekTimeout.stop();
+ return 0;
+}
+
+void eServiceMP3::seekTimeoutCB()
+{
+ pts_t ppos, len;
+ getPlayPosition(ppos);
+ getLength(len);
+ ppos += 90000*m_currentTrickRatio;
+
+ if (ppos < 0)
+ {
+ ppos = 0;
+ m_seekTimeout.stop();
+ }
+ if (ppos > len)
+ {
+ ppos = 0;
+ stop();
+ m_seekTimeout.stop();
+ return;
+ }
+ seekTo(ppos);
+}
// iPausableService
-RESULT eServiceMP3::pause() { printf("mp3 pauses!\n"); return 0; }
-RESULT eServiceMP3::unpause() { printf("mp3 unpauses!\n"); return 0; }
+RESULT eServiceMP3::pause()
+{
+ if (!m_gst_pipeline)
+ 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);
+ }
+ return 0;
+}
+
+RESULT eServiceMP3::unpause()
+{
+ if (!m_gst_pipeline)
+ return -1;
+
+ GstStateChangeReturn res;
+ res = gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
+ return 0;
+}
+
+ /* iSeekableService */
+RESULT eServiceMP3::seek(ePtr<iSeekableService> &ptr)
+{
+ ptr = this;
+ return 0;
+}
+
+RESULT eServiceMP3::getLength(pts_t &pts)
+{
+ if (!m_gst_pipeline)
+ 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))
+ return -1;
+
+ /* len is in nanoseconds. we have 90 000 pts per second. */
+
+ pts = len / 11111;
+ return 0;
+}
+
+RESULT eServiceMP3::seekTo(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,
+ GST_SEEK_TYPE_SET, time_nanoseconds,
+ GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE))
+ {
+ eDebug("SEEK failed");
+ return -1;
+ }
+ return 0;
+}
+
+RESULT eServiceMP3::seekRelative(int direction, pts_t to)
+{
+ if (!m_gst_pipeline)
+ return -1;
+
+ pts_t ppos;
+ getPlayPosition(ppos);
+ ppos += to * direction;
+ if (ppos < 0)
+ ppos = 0;
+ seekTo(ppos);
+
+ return 0;
+}
+
+RESULT eServiceMP3::getPlayPosition(pts_t &pts)
+{
+ if (!m_gst_pipeline)
+ 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))
+ return -1;
+
+ /* len is in nanoseconds. we have 90 000 pts per second. */
+ pts = len / 11111;
+ return 0;
+}
+
+RESULT eServiceMP3::setTrickmode(int trick)
+{
+ /* trickmode is not yet supported by our dvbmediasinks. */
+ return -1;
+}
+
+RESULT eServiceMP3::isCurrentlySeekable()
+{
+ return 1;
+}
-RESULT eServiceMP3::info(ePtr<iServiceInformation>&i) { i = this; return 0; }
+RESULT eServiceMP3::info(ePtr<iServiceInformation>&i)
+{
+ i = this;
+ return 0;
+}
RESULT eServiceMP3::getName(std::string &name)
{
- name = "MP3 File: " + filename;
+ name = m_filename;
+ size_t n = name.rfind('/');
+ if (n != std::string::npos)
+ name = name.substr(n + 1);
+ return 0;
+}
+
+int eServiceMP3::getInfo(int w)
+{
+ gchar *tag = 0;
+
+ switch (w)
+ {
+ case sTitle:
+ case sArtist:
+ case sAlbum:
+ case sComment:
+ case sTracknumber:
+ case sGenre:
+ case sVideoType:
+ return resIsString;
+ case sCurrentTitle:
+ tag = GST_TAG_TRACK_NUMBER;
+ break;
+ case sTotalTitles:
+ tag = GST_TAG_TRACK_COUNT;
+ break;
+ default:
+ return resNA;
+ }
+
+ if (!m_stream_tags || !tag)
+ return 0;
+
+ guint value;
+ if (gst_tag_list_get_uint(m_stream_tags, tag, &value))
+ return (int) value;
+
+ return 0;
+
+}
+
+std::string eServiceMP3::getInfoString(int w)
+{
+ gchar *tag = 0;
+ switch (w)
+ {
+ case sTitle:
+ tag = GST_TAG_TITLE;
+ break;
+ case sArtist:
+ tag = GST_TAG_ARTIST;
+ break;
+ case sAlbum:
+ tag = GST_TAG_ALBUM;
+ break;
+ case sComment:
+ tag = GST_TAG_COMMENT;
+ break;
+ case sTracknumber:
+ tag = GST_TAG_TRACK_NUMBER;
+ break;
+ case sGenre:
+ tag = GST_TAG_GENRE;
+ break;
+ case sVideoType:
+ tag = GST_TAG_VIDEO_CODEC;
+ break;
+ default:
+ return "";
+ }
+
+ if (!m_stream_tags || !tag)
+ return "";
+
+ gchar *value;
+
+ if (gst_tag_list_get_string(m_stream_tags, tag, &value))
+ {
+ std::string res = value;
+ g_free(value);
+ return res;
+ }
+
+ return "";
+}
+
+RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
+{
+ ptr = this;
+ return 0;
+}
+
+RESULT eServiceMP3::audioTracks(ePtr<iAudioTrackSelection> &ptr)
+{
+ ptr = this;
return 0;
}
+RESULT eServiceMP3::subtitle(ePtr<iSubtitleOutput> &ptr)
+{
+ ptr = this;
+ return 0;
+}
+
+int eServiceMP3::getNumberOfTracks()
+{
+ return m_audioStreams.size();
+}
+
+int eServiceMP3::getCurrentTrack()
+{
+ return m_currentAudioStream;
+}
+
+RESULT eServiceMP3::selectTrack(unsigned int i)
+{
+ int ret = selectAudioStream(i);
+ /* flush */
+ pts_t ppos;
+ getPlayPosition(ppos);
+ 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 )
+ {
+ eDebug("can't switch audio tracks! gst-plugin-selector needed");
+ return -1;
+ }
+ 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;
+}
+
+int eServiceMP3::getCurrentChannel()
+{
+ return STEREO;
+}
+
+RESULT eServiceMP3::selectChannel(int i)
+{
+ eDebug("eServiceMP3::selectChannel(%i)",i);
+ return 0;
+}
+
+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 = "MPEG";
+ else if (m_audioStreams[i].type == atMP3)
+ info.m_description = "MP3";
+ else if (m_audioStreams[i].type == atAC3)
+ info.m_description = "AC3";
+ else if (m_audioStreams[i].type == atAAC)
+ info.m_description = "AAC";
+ else if (m_audioStreams[i].type == atDTS)
+ info.m_description = "DTS";
+ else if (m_audioStreams[i].type == atPCM)
+ info.m_description = "PCM";
+ else if (m_audioStreams[i].type == atOGG)
+ info.m_description = "OGG";
+ else
+ info.m_description = "???";
+ if (info.m_language.empty())
+ info.m_language = m_audioStreams[i].language_code;
+ return 0;
+}
+
+void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
+{
+ if (!msg)
+ return;
+ gchar *sourceName;
+ GstObject *source;
+
+ source = GST_MESSAGE_SRC(msg);
+ sourceName = gst_object_get_name(source);
+
+ 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);
+ g_free(string);
+ }
+ else
+ eDebug("gst_message from %s: %s (without structure)", sourceName, GST_MESSAGE_TYPE_NAME(msg));
+
+ 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 )
+ {
+ if ( g_strrstr(sourceName, "videosink") )
+ m_event((iPlayableService*)this, evUser+11);
+ }
+ 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)
+ {
+ 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);
+ }
+ 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 )
+ {
+ 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) )
+ {
+ 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 )
+ {
+ 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) )
+ {
+ eDebug("found subtitle language %s",g_language);
+ IterSubtitleStream->language_code = std::string(g_language);
+ g_free (g_language);
+ }
+ }
+ }
+ }
+ default:
+ break;
+ }
+ g_free (sourceName);
+}
+
+GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
+{
+ eServiceMP3 *_this = (eServiceMP3*)user_data;
+ _this->m_pump.send(1);
+ /* wake */
+ return GST_BUS_PASS;
+}
+
+audiotype_t eServiceMP3::gstCheckAudioPad(GstStructure* structure)
+{
+ const gchar* type;
+ type = gst_structure_get_name(structure);
+
+ 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:
+ {
+ if ( layer == 3 )
+ return atMP3;
+ else
+ return atMPEG;
+ }
+ case 2:
+ return atMPEG;
+ 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;
+ }
+ return atUnknown;
+}
+
+void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
+{
+ 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"))
+ {
+ 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
+ {
+ gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink"));
+ _this->m_audioStreams.push_back(audio);
+ }
+ }
+ 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"))
+ {
+ GstElement *switch_subtitles = gst_bin_get_by_name(pipeline,"switch_subtitles");
+ if ( !switch_subtitles )
+ {
+ switch_subtitles = gst_element_factory_make ("input-selector", "switch_subtitles");
+ if ( !switch_subtitles )
+ return;
+ GstElement *parser = gst_element_factory_make("ssaparse", "parse_subtitles");
+ GstElement *sink = gst_element_factory_make("fakesink", "sink_subtitles");
+ gst_bin_add_many(pipeline, switch_subtitles, parser, sink, NULL);
+ gst_element_link(switch_subtitles, parser);
+ gst_element_link(parser, sink);
+ g_object_set (G_OBJECT(sink), "signal-handoffs", TRUE, NULL);
+ g_signal_connect(sink, "handoff", G_CALLBACK(gstCBsubtitleAvail), _this);
+ }
+ GstPad *sinkpad = gst_element_get_request_pad (switch_subtitles, "sink%d");
+ gst_pad_link(pad, sinkpad);
+ subtitleStream subs;
+ subs.pad = sinkpad;
+ _this->m_subtitleStreams.push_back(subs);
+ }
+}
+
+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"));
+}
+
+void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, 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);
+}
+
+void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
+{
+ 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)))
+ {
+ gstBusCall(bus, message);
+ gst_message_unref (message);
+ }
+}
+
eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
+
+void eServiceMP3::gstCBsubtitleAvail(GstElement *element, GstBuffer *buffer, GstPad *pad, gpointer user_data)
+{
+ const unsigned char *text = (unsigned char *)GST_BUFFER_DATA(buffer);
+ eDebug("gstCBsubtitleAvail: %s",text);
+ eServiceMP3 *_this = (eServiceMP3*)user_data;
+ if ( _this->m_subtitle_widget )
+ {
+ eDVBTeletextSubtitlePage page;
+ gRGB rgbcol(0xD0,0xD0,0xD0);
+ page.m_elements.push_back(eDVBTeletextSubtitlePageElement(rgbcol, (const char*)text));
+ (_this->m_subtitle_widget)->setPage(page);
+ }
+}
+
+RESULT eServiceMP3::enableSubtitles(eWidget *parent, ePyObject tuple)
+{
+ eDebug("eServiceMP3::enableSubtitles");
+
+ ePyObject entry;
+ int tuplesize = PyTuple_Size(tuple);
+ int pid;
+ gint nb_sources;
+ GstPad *active_pad;
+ GstElement *switch_subtitles = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_subtitles");
+
+ if (!PyTuple_Check(tuple))
+ goto error_out;
+ if (tuplesize < 1)
+ goto error_out;
+ entry = PyTuple_GET_ITEM(tuple, 1);
+ if (!PyInt_Check(entry))
+ goto error_out;
+ pid = PyInt_AsLong(entry);
+
+ m_subtitle_widget = new eSubtitleWidget(parent);
+ m_subtitle_widget->resize(parent->size()); /* full size */
+
+ if ( !switch_subtitles )
+ {
+ eDebug("can't switch subtitle tracks! gst-plugin-selector needed");
+ return -2;
+ }
+ g_object_get (G_OBJECT (switch_subtitles), "n-pads", &nb_sources, NULL);
+ if ( (unsigned int)pid >= m_subtitleStreams.size() || pid >= nb_sources || (unsigned int)m_currentSubtitleStream >= m_subtitleStreams.size() )
+ return -2;
+ char sinkpad[8];
+ sprintf(sinkpad, "sink%d", pid);
+ g_object_set (G_OBJECT (switch_subtitles), "active-pad", gst_element_get_pad (switch_subtitles, sinkpad), NULL);
+ g_object_get (G_OBJECT (switch_subtitles), "active-pad", &active_pad, NULL);
+ gchar *name;
+ name = gst_pad_get_name (active_pad);
+ eDebug ("switched subtitles to (%s)", name);
+ g_free(name);
+ m_currentSubtitleStream = pid;
+
+ return 0;
+error_out:
+ eDebug("enableSubtitles needs a tuple as 2nd argument!\n"
+ "for gst subtitles (2, subtitle_stream_count)");
+ return -1;
+}
+
+RESULT eServiceMP3::disableSubtitles(eWidget *parent)
+{
+ eDebug("eServiceMP3::disableSubtitles");
+ delete m_subtitle_widget;
+ m_subtitle_widget = 0;
+ return 0;
+}
+
+PyObject *eServiceMP3::getCachedSubtitle()
+{
+ eDebug("eServiceMP3::getCachedSubtitle");
+ Py_RETURN_NONE;
+}
+
+PyObject *eServiceMP3::getSubtitleList()
+{
+ eDebug("eServiceMP3::getSubtitleList");
+
+ ePyObject l = PyList_New(0);
+ int stream_count = 0;
+
+ for (std::vector<subtitleStream>::iterator IterSubtitleStream(m_subtitleStreams.begin()); IterSubtitleStream != m_subtitleStreams.end(); ++IterSubtitleStream)
+ {
+ ePyObject tuple = PyTuple_New(5);
+ PyTuple_SET_ITEM(tuple, 0, PyInt_FromLong(2));
+ PyTuple_SET_ITEM(tuple, 1, PyInt_FromLong(stream_count));
+ PyTuple_SET_ITEM(tuple, 2, PyInt_FromLong(0));
+ PyTuple_SET_ITEM(tuple, 3, PyInt_FromLong(0));
+ PyTuple_SET_ITEM(tuple, 4, PyString_FromString((IterSubtitleStream->language_code).c_str()));
+ PyList_Append(l, tuple);
+ Py_DECREF(tuple);
+ stream_count++;
+ }
+
+ return l;
+}
+
+#else
+#warning gstreamer not available, not building media player
+#endif
git.cweiske.de / enigma2.git / blobdiff