#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>
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> &);
+};
+
+DEFINE_REF(eMP3ServiceOfflineOperations);
+
+eMP3ServiceOfflineOperations::eMP3ServiceOfflineOperations(const eServiceReference &ref): m_ref((const eServiceReference&)ref)
{
- ptr = 0;
- return -1;
+}
+
+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
eServiceMP3::eServiceMP3(const char *filename): m_filename(filename), m_pump(eApp, 1)
{
m_stream_tags = 0;
+ m_audioStreams.clear();
+ m_currentAudioStream = 0;
+ m_currentTrickRatio = 0;
+ CONNECT(m_seekTimeout.timeout, eServiceMP3::seekTimeoutCB);
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, *queue_audio = 0, *video = 0, *queue_video = 0, *mpegdemux = 0;
+ GstElement *audio = 0, *switch_audio = 0, *queue_audio = 0, *video = 0, *queue_video = 0, *videodemux = 0;
m_state = stIdle;
eDebug("SERVICEMP3 construct!");
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/hda/track-", 18) || strcasecmp(ext, ".wav"));
+ 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 all_ok = 0;
- m_gst_pipeline = gst_pipeline_new ("audio-player");
+ m_gst_pipeline = gst_pipeline_new ("mediaplayer");
if (!m_gst_pipeline)
eWarning("failed to create pipeline");
/* 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);
+ 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);
}
/* filesrc -> mpegdemux -> | queue_audio -> dvbaudiosink
| queue_video -> dvbvideosink */
- audio = gst_element_factory_make("dvbaudiosink", "audio");
+ audio = gst_element_factory_make("dvbaudiosink", "audiosink");
queue_audio = gst_element_factory_make("queue", "queue_audio");
- video = gst_element_factory_make("dvbvideosink", "video");
+ video = gst_element_factory_make("dvbvideosink", "videosink");
queue_video = gst_element_factory_make("queue", "queue_video");
if (is_mpeg_ps)
- mpegdemux = gst_element_factory_make("flupsdemux", "mpegdemux");
+ videodemux = gst_element_factory_make("flupsdemux", "videodemux");
else if (is_mpeg_ts)
- mpegdemux = gst_element_factory_make("flutsdemux", "mpegdemux");
+ videodemux = gst_element_factory_make("flutsdemux", "videodemux");
else if (is_matroska)
- mpegdemux = gst_element_factory_make("matroskademux", "mpegdemux");
+ videodemux = gst_element_factory_make("matroskademux", "videodemux");
else if (is_avi)
- mpegdemux = gst_element_factory_make("avidemux", "mpegdemux");
+ videodemux = gst_element_factory_make("avidemux", "videodemux");
- if (!mpegdemux)
+ if (!videodemux)
{
eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
- mpegdemux = gst_element_factory_make("mpegdemux", "mpegdemux");
+ videodemux = gst_element_factory_make("mpegdemux", "videodemux");
}
-
- eDebug("audio: %p, queue_audio %p, video %p, queue_video %p, mpegdemux %p", audio, queue_audio, video, queue_video, mpegdemux);
- if (audio && queue_audio && video && queue_video && mpegdemux)
+
+ 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);
/* 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);
- m_decoder = decoder;
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 */
- m_gst_audio = gst_bin_new ("audiobin");
+ audio = gst_bin_new ("audiobin");
GstPad *audiopad = gst_element_get_static_pad (conv, "sink");
- gst_bin_add_many(GST_BIN(m_gst_audio), conv, flt, sink, (char*)0);
+ 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(m_gst_audio, gst_ghost_pad_new ("sink", audiopad));
+ gst_element_add_pad(audio, gst_ghost_pad_new ("sink", audiopad));
gst_object_unref(audiopad);
- gst_bin_add (GST_BIN(m_gst_pipeline), m_gst_audio);
-
+ 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, m_gst_audio);
+ 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, mpegdemux, audio, queue_audio, video, queue_video, NULL);
- gst_element_link(source, mpegdemux);
+ 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);
-
- m_gst_audioqueue = queue_audio;
- m_gst_videoqueue = queue_video;
-
- g_signal_connect(mpegdemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
+ g_signal_connect(videodemux, "pad-added", G_CALLBACK (gstCBpadAdded), this);
}
} else
{
gst_object_unref(GST_OBJECT(video));
if (queue_video)
gst_object_unref(GST_OBJECT(queue_video));
- if (mpegdemux)
- gst_object_unref(GST_OBJECT(mpegdemux));
+ 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;
assert(m_state != stIdle);
if (m_state == stStopped)
return -1;
- printf("MP3: %s stop\n", m_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;
return 0;
RESULT eServiceMP3::setSlowMotion(int ratio)
{
+ /* we can't do slomo yet */
return -1;
}
RESULT eServiceMP3::setFastForward(int ratio)
{
- return -1;
+ 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()
{
if (!m_gst_pipeline)
return -1;
- gst_element_set_state(m_gst_pipeline, GST_STATE_PAUSED);
+ 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;
}
{
if (!m_gst_pipeline)
return -1;
- gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
+
+ GstStateChangeReturn res;
+ res = gst_element_set_state(m_gst_pipeline, GST_STATE_PLAYING);
return 0;
}
if (!m_gst_pipeline)
return -1;
- pause();
-
pts_t ppos;
getPlayPosition(ppos);
ppos += to * direction;
ppos = 0;
seekTo(ppos);
- unpause();
-
return 0;
}
RESULT eServiceMP3::setTrickmode(int trick)
{
- /* trickmode currently doesn't make any sense for us. */
+ /* trickmode is not yet supported by our dvbmediasinks. */
return -1;
}
case sComment:
case sTracknumber:
case sGenre:
+ case sVideoType:
return resIsString;
case sCurrentTitle:
tag = GST_TAG_TRACK_NUMBER;
case sGenre:
tag = GST_TAG_GENRE;
break;
+ case sVideoType:
+ tag = GST_TAG_VIDEO_CODEC;
+ break;
default:
return "";
}
return "";
}
+RESULT eServiceMP3::audioChannel(ePtr<iAudioChannelSelection> &ptr)
+{
+ ptr = this;
+ return 0;
+}
- void foreach(const GstTagList *list, const gchar *tag, gpointer user_data)
- {
- if (tag)
- eDebug("Tag: %c%c%c%c", tag[0], tag[1], tag[2], tag[3]);
-
- }
+RESULT eServiceMP3::audioTracks(ePtr<iAudioTrackSelection> &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 *selector = gst_bin_get_by_name(GST_BIN(m_gst_pipeline),"switch_audio");
+ if ( !selector)
+ {
+ eDebug("can't switch audio tracks! gst-plugin-selector needed");
+ return -1;
+ }
+ 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;
+}
+
+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 == audioStream::atMP2)
+ info.m_description = "MP2";
+ else if (m_audioStreams[i].type == audioStream::atMP3)
+ info.m_description = "MP3";
+ else if (m_audioStreams[i].type == audioStream::atAC3)
+ info.m_description = "AC3";
+ else if (m_audioStreams[i].type == audioStream::atAAC)
+ info.m_description = "AAC";
+ else if (m_audioStreams[i].type == audioStream::atDTS)
+ info.m_description = "DTS";
+ else if (m_audioStreams[i].type == audioStream::atPCM)
+ info.m_description = "PCM";
+ else if (m_audioStreams[i].type == audioStream::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)
+ if (!msg)
+ return;
+ gchar *sourceName;
+ GstObject *source;
+
+ source = GST_MESSAGE_SRC(msg);
+ sourceName = gst_object_get_name(source);
+
+ if (gst_message_get_structure(msg))
{
- if (gst_message_get_structure(msg))
- {
- gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
- eDebug("gst_message: %s", string);
- g_free(string);
- }
- else
- eDebug("gst_message: %s (without structure)", GST_MESSAGE_TYPE_NAME(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:
{
gchar *debug;
GError *err;
+
gst_message_parse_error (msg, &err, &debug);
g_free (debug);
- eWarning("Gstreamer error: %s", err->message);
+ 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;
gst_tag_list_free(m_stream_tags);
m_stream_tags = result;
}
- gst_tag_list_free(tags);
-
- m_event((iPlayableService*)this, evUpdatedInfo);
+ gchar *g_audiocodec;
+ if (gst_tag_list_get_string(tags, GST_TAG_AUDIO_CODEC, &g_audiocodec) && m_audioStreams.size())
+ {
+ 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);
+ }
break;
}
default:
break;
}
+ g_free (sourceName);
}
GstBusSyncReply eServiceMP3::gstBusSyncHandler(GstBus *bus, GstMessage *message, gpointer user_data)
void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
{
eServiceMP3 *_this = (eServiceMP3*)user_data;
+ GstBin *pipeline = GST_BIN(_this->m_gst_pipeline);
gchar *name;
name = gst_pad_get_name (pad);
- g_print ("A new pad %s was created\n", name);
- GstPad *sinkpad;
-
- if (g_strrstr(name,"audio")) // mpegdemux uses video_nn with n=0,1,.., flupsdemux uses stream id
- gst_pad_link(pad, gst_element_get_static_pad (_this->m_gst_audioqueue, "sink"));
+ 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 )
+ {
+ GstPadLinkReturn ret = gst_pad_link(pad, gst_element_get_request_pad (selector, "sink%d"));
+ 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 (selector), "select-all", FALSE, NULL);
+ }
+ else
+ gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_audio"), "sink"));
+ }
if (g_strrstr(name,"video"))
- gst_pad_link(pad, gst_element_get_static_pad (_this->m_gst_videoqueue, "sink"));
+ {
+ gst_pad_link(pad, gst_element_get_static_pad(gst_bin_get_by_name(pipeline,"queue_video"), "sink"));
+ }
g_free (name);
}
void eServiceMP3::gstCBfilterPadAdded(GstElement *filter, GstPad *pad, gpointer user_data)
{
eServiceMP3 *_this = (eServiceMP3*)user_data;
- gst_pad_link(pad, gst_element_get_static_pad (_this->m_decoder, "sink"));
+ 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)
GstPad *audiopad;
/* only link once */
- audiopad = gst_element_get_static_pad (_this->m_gst_audio, "sink");
+ 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);
void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
{
- eServiceMP3 *_this = (eServiceMP3*)user_data;
GstStructure *str;
/* check media type */