{
m_stream_tags = 0;
CONNECT(m_pump.recv_msg, eServiceMP3::gstPoll);
- GstElement *source, *decoder, *conv, *flt, *sink;
+ 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;
+
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"));
+ int is_mpeg_ts = !strcasecmp(ext, ".ts");
+ int is_mp3 = !strcasecmp(ext, ".mp3"); /* force mp3 instead of decodebin */
+ int is_video = is_mpeg_ps || is_mpeg_ts;
+ int is_streaming = !strncmp(filename, "http://", 7);
+
+ eDebug("filename: %s, is_mpeg_ps: %d, is_mpeg_ts: %d, is_video: %d, is_streaming: %d, is_mp3: %d", filename, is_mpeg_ps, is_mpeg_ts, is_video, is_streaming, is_mp3);
+
+ int is_audio = !is_video;
+
+ int all_ok = 0;
m_gst_pipeline = gst_pipeline_new ("audio-player");
if (!m_gst_pipeline)
eWarning("failed to create pipeline");
- source = gst_element_factory_make ("filesrc", "file-source");
- if (!source)
- eWarning("failed to create filesrc");
-
- decoder = gst_element_factory_make ("decodebin", "decoder");
- if (!decoder)
- eWarning("failed to create decodebin decoder");
-
- 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");
-
- /* workaround for [3des]' driver bugs: */
- if (flt)
+ if (!is_streaming)
+ source = gst_element_factory_make ("filesrc", "file-source");
+ else
{
- GstCaps *caps = gst_caps_new_simple("audio/x-raw-int", "endianness", G_TYPE_INT, 4321, 0);
- g_object_set (G_OBJECT (flt), "caps", caps, 0);
- gst_caps_unref(caps);
+ source = gst_element_factory_make ("neonhttpsrc", "http-source");
+ if (source)
+ g_object_set (G_OBJECT (source), "automatic-redirect", TRUE, NULL);
}
-
- sink = gst_element_factory_make ("alsasink", "alsa-output");
- if (!sink)
- eWarning("failed to create osssink");
-
- if (m_gst_pipeline && source && decoder && conv && sink)
- {
- gst_bus_set_sync_handler(gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline)), gstBusSyncHandler, this);
+ if (!source)
+ eWarning("failed to create %s", is_streaming ? "neonhttpsrc" : "filesrc");
+ else
+ /* configure source */
g_object_set (G_OBJECT (source), "location", filename, NULL);
- g_signal_connect (decoder, "new-decoded-pad", G_CALLBACK(gstCBnewPad), this);
- /* gst_bin will take the 'floating references' */
- gst_bin_add_many (GST_BIN (m_gst_pipeline),
- source, decoder, NULL);
-
- gst_element_link(source, decoder);
+ if (is_audio)
+ {
+ /* filesrc -> decodebin -> audioconvert -> capsfilter -> alsasink */
+ const char *decodertype = is_mp3 ? "mad" : "decodebin";
+
+ decoder = gst_element_factory_make (decodertype, "decoder");
+ if (!decoder)
+ eWarning("failed to create %s decoder", decodertype);
+
+ /* 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");
+ }
+
+ 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", "audio");
+ queue_audio = gst_element_factory_make("queue", "queue_audio");
- /* create audio bin */
- m_gst_audio = gst_bin_new ("audiobin");
- GstPad *audiopad = gst_element_get_pad (conv, "sink");
+ video = gst_element_factory_make("dvbvideosink", "video");
+ queue_video = gst_element_factory_make("queue", "queue_video");
- gst_bin_add_many(GST_BIN(m_gst_audio), conv, flt, sink, 0);
- gst_element_link_many(conv, flt, sink, 0);
- gst_element_add_pad(m_gst_audio, gst_ghost_pad_new ("sink", audiopad));
- gst_object_unref(audiopad);
+ if (is_mpeg_ps)
+ mpegdemux = gst_element_factory_make("flupsdemux", "mpegdemux");
+ else
+ mpegdemux = gst_element_factory_make("flutsdemux", "mpegdemux");
+
+ if (!mpegdemux)
+ {
+ eDebug("fluendo mpegdemux not available, falling back to mpegdemux\n");
+ mpegdemux = gst_element_factory_make("mpegdemux", "mpegdemux");
+ }
- gst_bin_add (GST_BIN(m_gst_pipeline), m_gst_audio);
+ 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)
+ {
+ 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-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_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);
+ }
+
+ /* gst_bin will take the 'floating references' */
+ gst_bin_add_many (GST_BIN (m_gst_pipeline),
+ source, 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);
+ 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(source, decoder);
+
+ /* create audio bin with the audioconverter, the capsfilter and the audiosink */
+ m_gst_audio = gst_bin_new ("audiobin");
+
+ GstPad *audiopad = gst_element_get_pad (conv, "sink");
+ gst_bin_add_many(GST_BIN(m_gst_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_object_unref(audiopad);
+ gst_bin_add (GST_BIN(m_gst_pipeline), m_gst_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);
+ } else
+ {
+ gst_bin_add_many(GST_BIN(m_gst_pipeline), source, mpegdemux, audio, queue_audio, video, queue_video, NULL);
+ gst_element_link(source, mpegdemux);
+ 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);
+ }
} else
{
if (m_gst_pipeline)
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 (mpegdemux)
+ gst_object_unref(GST_OBJECT(mpegdemux));
+
eDebug("sorry, can't play.");
+ m_gst_pipeline = 0;
}
gst_element_set_state (m_gst_pipeline, GST_STATE_PLAYING);
RESULT eServiceMP3::getName(std::string &name)
{
- name = "MP3 File: " + m_filename;
+ name = m_filename;
+ size_t n = name.rfind('/');
+ if (n != std::string::npos)
+ name = name.substr(n + 1);
return 0;
}
void eServiceMP3::gstBusCall(GstBus *bus, GstMessage *msg)
{
- gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
- eDebug("gst_message: %s", string);
- g_free(string);
-
+ if (msg)
+ {
+ gchar *string = gst_structure_to_string(gst_message_get_structure(msg));
+ eDebug("gst_message: %s", string);
+ g_free(string);
+ }
switch (GST_MESSAGE_TYPE (msg))
{
g_free (debug);
eWarning("Gstreamer error: %s", err->message);
g_error_free(err);
- exit(0);
+ /* TODO: signal error condition to user */
break;
}
case GST_MESSAGE_TAG:
m_stream_tags = result;
}
gst_tag_list_free(tags);
+
+ m_event((iPlayableService*)this, evUpdatedInfo);
break;
}
default:
return GST_BUS_PASS;
}
+void eServiceMP3::gstCBpadAdded(GstElement *decodebin, GstPad *pad, gpointer user_data)
+{
+ eServiceMP3 *_this = (eServiceMP3*)user_data;
+
+ gchar *name;
+ name = gst_pad_get_name (pad);
+ g_print ("A new pad %s was created\n", name);
+ if (!strncmp(name, "audio_", 6)) // mpegdemux uses video_nn with n=0,1,.., flupsdemux uses stream id
+ gst_pad_link(pad, gst_element_get_pad (_this->m_gst_audioqueue, "sink"));
+ if (!strncmp(name, "video_", 6))
+ gst_pad_link(pad, gst_element_get_pad (_this->m_gst_videoqueue, "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_pad (_this->m_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 */
audiopad = gst_element_get_pad (_this->m_gst_audio, "sink");
if (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);
gst_pad_link (pad, audiopad);
}
+void eServiceMP3::gstCBunknownType(GstElement *decodebin, GstPad *pad, GstCaps *caps, gpointer user_data)
+{
+ eServiceMP3 *_this = (eServiceMP3*)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&)
{
GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (m_gst_pipeline));
GstMessage *message;
- while (message = gst_bus_pop (bus))
+ while ((message = gst_bus_pop (bus)))
{
gstBusCall(bus, message);
gst_message_unref (message);
git.cweiske.de / enigma2.git / blobdiff