#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 <gst/pbutils/missing-plugins.h>
+#include <sys/stat.h>
+/* for subtitles */
+#include <lib/gui/esubtitle.h>
+
+#ifndef GST_SEEK_FLAG_SKIP
+#warning Compiling for legacy gstreamer, things will break
+#define GST_SEEK_FLAG_SKIP 0
+#define GST_TAG_HOMEPAGE ""
+#endif
// 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");
+ extensions.push_back("m2ts");
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;
}
return 0;
}
-RESULT eServiceFactoryMP3::offlineOperations(const eServiceReference &, ePtr<iServiceOfflineOperations> &ptr)
+class eMP3ServiceOfflineOperations: public iServiceOfflineOperations
+{
+ DECLARE_REF(eMP3ServiceOfflineOperations);
+ eServiceReference m_ref;
+public:
+ eMP3ServiceOfflineOperations(const eServiceReference &ref);
+
+ RESULT deleteFromDisk(int simulate);
+ RESULT getListOfFilenames(std::list<std::string> &);
+ RESULT reindex();
+};
+
+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 eMP3ServiceOfflineOperations::reindex()
{
- ptr = 0;
return -1;
}
+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)
{
- 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
-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_audioStreams.clear();
m_currentAudioStream = 0;
+ m_currentSubtitleStream = 0;
+ m_subtitle_widget = 0;
m_currentTrickRatio = 0;
- CONNECT(m_seekTimeout.timeout, eServiceMP3::seekTimeoutCB);
+ 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 *filter = 0, *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;
- int is_mpeg_ps = !(strcasecmp(ext, ".mpeg") && strcasecmp(ext, ".mpg") && strcasecmp(ext, ".vob") && strcasecmp(ext, ".bin"));
- 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)
+ 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 )
{
- 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;
+ sourceinfo.containertype = ctMPEGPS;
+ sourceinfo.is_video = TRUE;
}
- if ( !is_streaming && !is_AudioCD )
- source = gst_element_factory_make ("filesrc", "file-source");
- else if ( is_streaming )
+ else if ( strcasecmp(ext, ".ts") == 0 )
{
- source = gst_element_factory_make ("neonhttpsrc", "http-source");
- if (source)
- g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
+ sourceinfo.containertype = ctMPEGTS;
+ sourceinfo.is_video = TRUE;
}
-
- 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);
+ else if ( strcasecmp(ext, ".mkv") == 0 )
+ {
+ sourceinfo.containertype = ctMKV;
+ sourceinfo.is_video = TRUE;
}
-
- if (is_audio)
+ else if ( strcasecmp(ext, ".avi") == 0 || strcasecmp(ext, ".divx") == 0)
{
- /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
- const char *decodertype = is_mp3 ? "mad" : "decodebin";
+ sourceinfo.containertype = ctAVI;
+ 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;
+ 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;
- decoder = gst_element_factory_make (decodertype, "decoder");
- if (!decoder)
- eWarning("failed to create %s decoder", decodertype);
+ gchar *uri;
- /* mp3 decoding needs id3demux to extract ID3 data. 'decodebin' would do that internally. */
- if (is_mp3)
- {
- filter = gst_element_factory_make ("id3demux", "filter");
- if (!filter)
- eWarning("failed to create id3demux");
- }
+ if ( sourceinfo.is_streaming )
+ {
+ uri = g_strdup_printf ("%s", filename);
+ }
+ else if ( sourceinfo.containertype == ctCDA )
+ {
+ int i_track = atoi(filename+18);
+ uri = g_strdup_printf ("cdda://%i", i_track);
+ }
+ else if ( sourceinfo.containertype == ctVCD )
+ {
+ int fd = open(filename,O_RDONLY);
+ char tmp[128*1024];
+ int ret = read(fd, tmp, 128*1024);
+ close(fd);
+ if ( ret == -1 ) // this is a "REAL" VCD
+ uri = g_strdup_printf ("vcd://");
+ else
+ uri = g_strdup_printf ("file://%s", filename);
+ }
+ else
- conv = gst_element_factory_make ("audioconvert", "converter");
- if (!conv)
- eWarning("failed to create audioconvert");
+ uri = g_strdup_printf ("file://%s", filename);
- flt = gst_element_factory_make ("capsfilter", "flt");
- if (!flt)
- eWarning("failed to create capsfilter");
+ eDebug("eServiceMP3::playbin2 uri=%s", uri);
- /* 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);
- }
+ m_gst_playbin = gst_element_factory_make("playbin2", "playbin");
+ if (!m_gst_playbin)
+ m_error_message = "failed to create GStreamer pipeline!\n";
- sink = gst_element_factory_make ("alsasink", "alsa-output");
- if (!sink)
- eWarning("failed to create osssink");
+ g_object_set (G_OBJECT (m_gst_playbin), "uri", uri, NULL);
- if (source && decoder && conv && sink)
- all_ok = 1;
- } else /* is_video */
- {
- /* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
- | queue_video -> dvbvideosink */
+ 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);
- 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");
- }
+ g_free(uri);
- 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)
+ 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);
+ g_signal_connect (subsink, "new-buffer", G_CALLBACK (gstCBsubtitleAvail), this);
+ g_object_set (G_OBJECT (m_gst_playbin), "text-sink", subsink, NULL);
+ }
- if (is_AudioCD)
+ 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_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");
-
- if (!is_mp3)
- {
- /* decodebin has dynamic pads. When they get created, we connect them to the audio bin */
- 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);
-
- if (filter)
- {
- /* id3demux also has dynamic pads, which need to be connected to the decoder (this is done in the 'gstCBfilterPadAdded' CB) */
- gst_bin_add(GST_BIN(m_gst_pipeline), filter);
- gst_element_link(source, filter);
- g_signal_connect (filter, "pad-added", G_CALLBACK(gstCBfilterPadAdded), this);
- } else
- /* 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);
- audioStream audioStreamElem;
- m_audioStreams.push_back(audioStreamElem);
- } else /* is_video */
- {
- 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);
+ 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
{
- 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;
+ m_event((iPlayableService*)this, evUser+12);
+
+ 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);
+
+ gst_element_set_state (m_gst_playbin, GST_STATE_PLAYING);
+ setBufferSize(m_buffer_size);
}
eServiceMP3::~eServiceMP3()
{
+ delete m_subtitle_widget;
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!");
}
}
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()
if (ppos < 0)
{
ppos = 0;
- m_seekTimeout.stop();
+ m_seekTimeout->stop();
}
if (ppos > len)
{
ppos = 0;
stop();
- m_seekTimeout.stop();
+ m_seekTimeout->stop();
return;
}
seekTo(ppos);
// 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);
+ GstStateChangeReturn res = gst_element_set_state(m_gst_playbin, GST_STATE_PAUSED);
if (res == GST_STATE_CHANGE_ASYNC)
{
pts_t ppos;
RESULT eServiceMP3::unpause()
{
- if (!m_gst_pipeline)
+ m_subtitle_pages.clear();
+ if (!m_gst_playbin || m_state != stRunning)
return -1;
GstStateChangeReturn res;
- res = gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
+ res = 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;
RESULT eServiceMP3::seekTo(pts_t to)
{
- if (!m_gst_pipeline)
+ if (!m_gst_playbin)
return -1;
+ eSingleLocker l(m_subs_to_pull_lock); // this is needed to dont handle incomming subtitles during seek!
+
/* 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;
+ }
+
+ m_subtitle_pages.clear();
+ m_subs_to_pull = 0;
+
+ return 0;
+}
+
+RESULT eServiceMP3::trickSeek(gdouble ratio)
+{
+ if (!m_gst_playbin)
+ return -1;
+ if (!ratio)
+ return seekRelative(0, 0);
+
+ GstEvent *s_event;
+ GstSeekFlags flags;
+ flags = GST_SEEK_FLAG_NONE;
+ flags |= GstSeekFlags (GST_SEEK_FLAG_FLUSH);
+// flags |= GstSeekFlags (GST_SEEK_FLAG_ACCURATE);
+ flags |= GstSeekFlags (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, 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, 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 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)
{
- gchar *tag = 0;
+ if ( !m_stream_tags && w < sUser && w > 26 )
+ return "";
+ 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;
+ case sTagTitleSortname:
+ tag = GST_TAG_TITLE_SORTNAME;
break;
- case sTracknumber:
- tag = GST_TAG_TRACK_NUMBER;
+ case sTagArtistSortname:
+ tag = GST_TAG_ARTIST_SORTNAME;
+ break;
+ case sTagAlbumSortname:
+ tag = GST_TAG_ALBUM_SORTNAME;
+ break;
+ 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-%M-%D", date);
+ return (std::string)res;
+ }
+ break;
+ case sTagComposer:
+ tag = GST_TAG_COMPOSER;
break;
- case sGenre:
+ case sTagGenre:
tag = GST_TAG_GENRE;
break;
- case sVideoType:
+ 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 "";
}
-
- if (!m_stream_tags || !tag)
+ if ( !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 "";
}
+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;
+ }
+ gdouble value;
+ if ( !tag || !m_stream_tags )
+ value = 0.0;
+ PyObject *pyValue;
+ 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);
+ pyValue = PyBuffer_FromMemory(GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer));
+ }
+ }
+ else
+ {
+ gst_tag_list_get_double(m_stream_tags, tag, &value);
+ pyValue = PyFloat_FromDouble(value);
+ }
+
+ return pyValue;
+}
+
RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
{
ptr = this;
return 0;
}
+RESULT eServiceMP3::subtitle(ePtr<iSubtitleOutput> &ptr)
+{
+ ptr = this;
+ return 0;
+}
+
int eServiceMP3::getNumberOfTracks()
{
return m_audioStreams.size();
int eServiceMP3::selectAudioStream(int i)
{
- gint nb_sources;
- GstPad *active_pad;
- GstElement *selector = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
- if ( !selector)
+ 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 (selector), "n-pads", &nb_sources, NULL);
- if ( i >= m_audioStreams.size() || i >= nb_sources || m_currentAudioStream >= m_audioStreams.size() )
- return -2;
- char sinkpad[8];
- sprintf(sinkpad, "sink%d", i);
- g_object_set (G_OBJECT (selector), "active-pad", gst_element_get_pad (selector, sinkpad), NULL);
- g_object_get (G_OBJECT (selector), "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 == audioStream::atMP2)
- info.m_description = "MP2";
- else if (m_audioStreams[i].type == audioStream::atMP3)
+ 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";
- else if (m_audioStreams[i].type == audioStream::atAC3)
+ else if (m_audioStreams[i].type == atAC3)
info.m_description = "AC3";
- else if (m_audioStreams[i].type == audioStream::atAAC)
+ else if (m_audioStreams[i].type == atAAC)
info.m_description = "AAC";
- else if (m_audioStreams[i].type == audioStream::atDTS)
+ else if (m_audioStreams[i].type == atDTS)
info.m_description = "DTS";
- else if (m_audioStreams[i].type == audioStream::atPCM)
+ else if (m_audioStreams[i].type == atPCM)
info.m_description = "PCM";
- else if (m_audioStreams[i].type == audioStream::atOGG)
+ 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;
GstObject *source;
source = GST_MESSAGE_SRC(msg);
- sourceName = gst_object_get_name(source);
-
+ sourceName = gst_object_get_name(source);
+#if 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);
+ }
+ } 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);
+ case GST_MESSAGE_INFO:
+ {
+ gchar *debug;
+ GError *inf;
+
+ gst_message_parse_info (msg, &inf, &debug);
+ g_free (debug);
+ if ( inf->domain == GST_STREAM_ERROR && inf->code == GST_STREAM_ERROR_DECODE )
+ {
+ if ( g_strrstr(sourceName, "videosink") )
+ m_event((iPlayableService*)this, evUser+14);
+ }
+ g_error_free(inf);
+ break;
+ }
+ case GST_MESSAGE_TAG:
+ {
+ GstTagList *tags, *result;
+ gst_message_parse_tag(msg, &tags);
+
+ result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_REPLACE);
+ if (result)
+ {
+ if (m_stream_tags)
+ gst_tag_list_free(m_stream_tags);
+ m_stream_tags = result;
+ }
+
+ const GValue *gv_image = gst_tag_list_get_value_index(tags, GST_TAG_IMAGE, 0);
+ if ( gv_image )
+ {
+ GstBuffer *buf_image;
+ buf_image = gst_value_get_buffer (gv_image);
+ int fd = open("/tmp/.id3coverart", O_CREAT|O_WRONLY|O_TRUNC, 0644);
+ int ret = write(fd, GST_BUFFER_DATA(buf_image), GST_BUFFER_SIZE(buf_image));
+ close(fd);
+ eDebug("eServiceMP3::/tmp/.id3coverart %d bytes written ", ret);
+ m_event((iPlayableService*)this, evUser+13);
+ }
+ gst_tag_list_free(tags);
+ m_event((iPlayableService*)this, evUpdatedInfo);
+ break;
+ }
+ case GST_MESSAGE_ASYNC_DONE:
+ {
+ if(GST_MESSAGE_SRC(msg) != GST_OBJECT(m_gst_playbin))
+ break;
+
+ GstTagList *tags;
+ gint i, active_idx, n_video = 0, n_audio = 0, n_text = 0;
+
+ g_object_get (m_gst_playbin, "n-video", &n_video, NULL);
+ g_object_get (m_gst_playbin, "n-audio", &n_audio, NULL);
+ g_object_get (m_gst_playbin, "n-text", &n_text, NULL);
+
+ eDebug("eServiceMP3::async-done - %d video, %d audio, %d subtitle", n_video, n_audio, n_text);
+
+ if ( n_video + n_audio <= 0 )
+ stop();
+
+ active_idx = 0;
+
+ m_audioStreams.clear();
+ m_subtitleStreams.clear();
- result = gst_tag_list_merge(m_stream_tags, tags, GST_TAG_MERGE_PREPEND);
- if (result)
+ for (i = 0; i < n_audio; i++)
+ {
+ audioStream audio;
+ gchar *g_codec, *g_lang;
+ GstPad* pad = 0;
+ g_signal_emit_by_name (m_gst_playbin, "get-audio-pad", i, &pad);
+ GstCaps* caps = gst_pad_get_negotiated_caps(pad);
+ if (!caps)
+ continue;
+ GstStructure* str = gst_caps_get_structure(caps, 0);
+ gchar *g_type;
+ g_type = gst_structure_get_name(str);
+ eDebug("AUDIO STRUCT=%s", g_type);
+ audio.type = gstCheckAudioPad(str);
+ g_codec = g_strdup(g_type);
+ g_lang = g_strdup_printf ("und");
+ g_signal_emit_by_name (m_gst_playbin, "get-audio-tags", i, &tags);
+ if ( tags && gst_is_tag_list(tags) )
+ {
+ gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_codec);
+ gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_lang);
+ gst_tag_list_free(tags);
+ }
+ audio.language_code = std::string(g_lang);
+ audio.codec = std::string(g_codec);
+ eDebug("eServiceMP3::audio stream=%i codec=%s language=%s", i, g_codec, g_lang);
+ m_audioStreams.push_back(audio);
+ g_free (g_lang);
+ g_free (g_codec);
+ gst_caps_unref(caps);
+ }
+
+ for (i = 0; i < n_text; i++)
+ {
+ gchar *g_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);
+ }
+ case GST_MESSAGE_ELEMENT:
{
- if (m_stream_tags)
- gst_tag_list_free(m_stream_tags);
- m_stream_tags = result;
+ if ( gst_is_missing_plugin_message(msg) )
+ {
+ gchar *description = gst_missing_plugin_message_get_description(msg);
+ if ( description )
+ {
+ m_error_message = "GStreamer plugin " + (std::string)description + " not available!\n";
+ g_free(description);
+ m_event((iPlayableService*)this, evUser+12);
+ }
+ }
+ else if (const GstStructure *msgstruct = gst_message_get_structure(msg))
+ {
+ const gchar *eventname = 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;
}
- gchar *g_audiocodec;
- if (gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec) && m_audioStreams.size())
+ case GST_MESSAGE_BUFFERING:
{
- std::vector<audioStream>::iterator IterAudioStream = m_audioStreams.begin();
- while ( IterAudioStream->language_code.length() && IterAudioStream != m_audioStreams.end())
- IterAudioStream++;
- if ( g_strrstr(g_audiocodec, "MPEG-1 layer 2") )
- IterAudioStream->type = audioStream::atMP2;
- else if ( g_strrstr(g_audiocodec, "MPEG-1 layer 3") )
- IterAudioStream->type = audioStream::atMP3;
- else if ( g_strrstr(g_audiocodec, "AC-3 audio") )
- IterAudioStream->type = audioStream::atAC3;
- else if ( g_strrstr(g_audiocodec, "Uncompressed 16-bit PCM audio") )
- IterAudioStream->type = audioStream::atPCM;
- gchar *g_language;
- if ( gst_tag_list_get_string(tags, GST_TAG_LANGUAGE_CODE, &g_language) )
- IterAudioStream->language_code = std::string(g_language);
- g_free (g_language);
- g_free (g_audiocodec);
+ GstBufferingMode mode;
+ gst_message_parse_buffering(msg, &(m_bufferInfo.bufferPercent));
+ gst_message_parse_buffering_stats(msg, &mode, &(m_bufferInfo.avgInRate), &(m_bufferInfo.avgOutRate), &(m_bufferInfo.bufferingLeft));
+ m_event((iPlayableService*)this, evBuffering);
}
- break;
- }
- default:
- break;
+ default:
+ break;
}
g_free (sourceName);
}
return GST_BUS_PASS;
}
-void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
+audiotype_t eServiceMP3::gstCheckAudioPad(GstStructure* 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;
+
+ switch (mpegversion) {
+ case 1:
+ {
+ gst_structure_get_int (structure, "layer", &layer);
+ if ( layer == 3 )
+ return atMP3;
+ else
+ return atMPEG;
+ break;
+ }
+ case 2:
+ return atAAC;
+ case 4:
+ return atAAC;
+ default:
+ return atUnknown;
+ }
+ }
+
+ 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::gstPoll(const int &msg)
+{
+ /* 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)
+ {
+ GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_playbin));
+ GstMessage *message;
+ usleep(1);
+ while ((message = gst_bus_pop (bus)))
+ {
+ gstBusCall(bus, message);
+ gst_message_unref (message);
+ }
+ }
+ else
+ pullSubtitle();
+}
+
+eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
+
+void eServiceMP3::gstCBsubtitleAvail(GstElement *appsink, gpointer user_data)
{
eServiceMP3 *_this = (eServiceMP3*)user_data;
- GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
- gchar *name;
- name = gst_pad_get_name (pad);
- eDebug ("A new pad %s was created", name);
- if (g_strrstr(name,"audio")) // mpegdemux, matroskademux, avidemux use video_nn with n=0,1,.., flupsdemux uses stream id
- {
- GstElement *selector = gst_bin_get_by_name(pipeline , "switch_audio" );
- audioStream audio;
- audio.pad = pad;
- _this->m_audioStreams.push_back(audio);
- if ( selector )
+ eSingleLocker l(_this->m_subs_to_pull_lock);
+ ++_this->m_subs_to_pull;
+ _this->m_pump.send(2);
+}
+
+void eServiceMP3::pullSubtitle()
+{
+ GstElement *sink;
+ g_object_get (G_OBJECT (m_gst_playbin), "text-sink", &sink, NULL);
+ if (sink)
+ {
+ 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!");
+}
+
+void eServiceMP3::pushSubtitles()
+{
+ ePangoSubtitlePage page;
+ pts_t running_pts;
+ while ( !m_subtitle_pages.empty() )
+ {
+ 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)
{
- GstPadLinkReturn ret = gst_pad_link(pad, gst_element_get_request_pad (selector, "sink%d"));
- if ( _this->m_audioStreams.size() == 1 )
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint64 now;
+ if (gst_element_query_position(m_gst_playbin, &fmt, &now) != -1)
{
- _this->selectAudioStream(0);
- gst_element_set_state (_this->m_gst_pipeline, GST_STATE_PLAYING);
+ 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
- g_object_set (G_OBJECT (selector), "select-all", FALSE, NULL);
+ 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();
}
- else
- gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink"));
}
- if (g_strrstr(name,"video"))
+ if (m_subtitle_pages.empty())
+ pullSubtitle();
+}
+
+RESULT eServiceMP3::enableSubtitles(eWidget *parent, ePyObject tuple)
+{
+ ePyObject entry;
+ int tuplesize = PyTuple_Size(tuple);
+ int pid, type;
+ gint text_pid = 0;
+
+ 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);
+ entry = PyTuple_GET_ITEM(tuple, 2);
+ if (!PyInt_Check(entry))
+ goto error_out;
+ type = PyInt_AsLong(entry);
+
+ if (m_currentSubtitleStream != pid)
{
- gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_video"), "sink"));
+ g_object_set (G_OBJECT (m_gst_playbin), "current-text", pid, NULL);
+ m_currentSubtitleStream = pid;
+ eSingleLocker l(m_subs_to_pull_lock);
+ m_subs_to_pull = 0;
+ m_subtitle_pages.clear();
}
- g_free (name);
+
+ m_subtitle_widget = 0;
+ m_subtitle_widget = new eSubtitleWidget(parent);
+ m_subtitle_widget->resize(parent->size()); /* full size */
+
+ g_object_get (G_OBJECT (m_gst_playbin), "current-text", &text_pid, NULL);
+
+ eDebug ("eServiceMP3::switched to subtitle stream %i", text_pid);
+
+
+ return 0;
+
+error_out:
+ eDebug("eServiceMP3::enableSubtitles needs a tuple as 2nd argument!\n"
+ "for gst subtitles (2, subtitle_stream_count, subtitle_type)");
+ return -1;
}
-void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
+RESULT eServiceMP3::disableSubtitles(eWidget *parent)
{
- 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"));
+ eDebug("eServiceMP3::disableSubtitles");
+ m_subtitle_pages.clear();
+ delete m_subtitle_widget;
+ m_subtitle_widget = 0;
+ return 0;
}
-void eServiceMP3::gstCBnewPad(GstElement *decodebin, GstPad *pad, gboolean last, gpointer user_data)
+PyObject *eServiceMP3::getCachedSubtitle()
{
- eServiceMP3 *_this = (eServiceMP3*)user_data;
- GstCaps *caps;
- GstStructure *str;
- GstPad *audiopad;
-
- /* only link once */
- GstElement *audio = gst_bin_get_by_name(GST_BIN(_this->m_gst_pipeline),"audiobin");
- audiopad = gst_element_get_static_pad (audio, "sink");
- if ( !audiopad || GST_PAD_IS_LINKED (audiopad)) {
- eDebug("audio already linked!");
- g_object_unref (audiopad);
- return;
- }
+// eDebug("eServiceMP3::getCachedSubtitle");
+ Py_RETURN_NONE;
+}
- /* 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;
+PyObject *eServiceMP3::getSubtitleList()
+{
+ eDebug("eServiceMP3::getSubtitleList");
+
+ ePyObject l = PyList_New(0);
+ int stream_count[sizeof(subtype_t)];
+ for ( unsigned int i = 0; i < sizeof(subtype_t); i++ )
+ stream_count[i] = 0;
+
+ for (std::vector<subtitleStream>::iterator IterSubtitleStream(m_subtitleStreams.begin()); IterSubtitleStream != m_subtitleStreams.end(); ++IterSubtitleStream)
+ {
+ subtype_t type = IterSubtitleStream->type;
+ ePyObject tuple = PyTuple_New(5);
+ PyTuple_SET_ITEM(tuple, 0, PyInt_FromLong(2));
+ PyTuple_SET_ITEM(tuple, 1, PyInt_FromLong(stream_count[type]));
+ PyTuple_SET_ITEM(tuple, 2, PyInt_FromLong(int(type)));
+ 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[type]++;
}
-
- gst_caps_unref (caps);
- gst_pad_link (pad, audiopad);
+ return l;
}
-void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
+RESULT eServiceMP3::streamed(ePtr<iStreamedService> &ptr)
{
- GstStructure *str;
+ ptr = this;
+ return 0;
+}
- /* 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);
+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;
}
-void eServiceMP3::gstPoll(const int&)
+int eServiceMP3::setBufferSize(int size)
{
- /* 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);
- }
+ m_buffer_size = size;
+ g_object_set (G_OBJECT (m_gst_playbin), "buffer-size", m_buffer_size, NULL);
+ return 0;
}
-eAutoInitPtr<eServiceFactoryMP3> init_eServiceFactoryMP3(eAutoInitNumbers::service+1, "eServiceFactoryMP3");
+
#else
#warning gstreamer not available, not building media player
#endif
git.cweiske.de / enigma2.git / blobdiff