From 92cfb335cde7386b53c76e030603a0897429a6ff Mon Sep 17 00:00:00 2001 From: Nicolas Baron Date: Fri, 9 Dec 2011 15:45:03 +0000 Subject: [PATCH 01/65] jack: add "client-name" property to jackaudiosink and jackaudiosrc https://bugzilla.gnome.org/show_bug.cgi?id=665872 --- ext/jack/gstjackaudiosink.c | 26 +++++++++++++++++++++++--- ext/jack/gstjackaudiosink.h | 1 + ext/jack/gstjackaudiosrc.c | 26 +++++++++++++++++++++++--- ext/jack/gstjackaudiosrc.h | 1 + 4 files changed, 48 insertions(+), 6 deletions(-) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index 58e2acf621..e1513b7e7e 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -329,7 +329,11 @@ gst_jack_ring_buffer_open_device (GstRingBuffer * buf) GST_DEBUG_OBJECT (sink, "open"); - name = g_get_application_name (); + if (sink->client_name) { + name = sink->client_name; + } else { + name = g_get_application_name (); + } if (!name) name = "GStreamer"; @@ -644,8 +648,9 @@ enum SIGNAL_LAST }; -#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO -#define DEFAULT_PROP_SERVER NULL +#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO +#define DEFAULT_PROP_SERVER NULL +#define DEFAULT_PROP_CLIENT_NAME NULL enum { @@ -653,6 +658,7 @@ enum PROP_CONNECT, PROP_SERVER, PROP_CLIENT, + PROP_CLIENT_NAME, PROP_LAST }; @@ -711,6 +717,12 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) "The Jack server to connect to (NULL = default)", DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_CLIENT_NAME, + g_param_spec_string ("client-name", "Client name", + "The client name of the Jack instance (NULL = default)", + DEFAULT_PROP_CLIENT_NAME, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_CLIENT, g_param_spec_boxed ("client", "JackClient", "Handle for jack client", GST_TYPE_JACK_CLIENT, @@ -738,6 +750,7 @@ gst_jack_audio_sink_init (GstJackAudioSink * sink, sink->jclient = NULL; sink->ports = NULL; sink->port_count = 0; + sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME); sink->buffers = NULL; } @@ -759,6 +772,10 @@ gst_jack_audio_sink_set_property (GObject * object, guint prop_id, sink = GST_JACK_AUDIO_SINK (object); switch (prop_id) { + case PROP_CLIENT_NAME: + g_free (sink->client_name); + sink->client_name = g_value_dup_string (value); + break; case PROP_CONNECT: sink->connect = g_value_get_enum (value); break; @@ -787,6 +804,9 @@ gst_jack_audio_sink_get_property (GObject * object, guint prop_id, sink = GST_JACK_AUDIO_SINK (object); switch (prop_id) { + case PROP_CLIENT_NAME: + g_value_set_string (value, sink->client_name); + break; case PROP_CONNECT: g_value_set_enum (value, sink->connect); break; diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h index 6f9c45c8eb..db5fc282f0 100644 --- a/ext/jack/gstjackaudiosink.h +++ b/ext/jack/gstjackaudiosink.h @@ -58,6 +58,7 @@ struct _GstJackAudioSink { GstJackConnect connect; gchar *server; jack_client_t *jclient; + gchar *client_name; /* our client */ GstJackAudioClient *client; diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 27faa82bdd..469737d9e8 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -336,7 +336,11 @@ gst_jack_ring_buffer_open_device (GstRingBuffer * buf) GST_DEBUG_OBJECT (src, "open"); - name = g_get_application_name (); + if (src->client_name) { + name = src->client_name; + } else { + name = g_get_application_name (); + } if (!name) name = "GStreamer"; @@ -647,8 +651,9 @@ enum LAST_SIGNAL }; -#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO -#define DEFAULT_PROP_SERVER NULL +#define DEFAULT_PROP_CONNECT GST_JACK_CONNECT_AUTO +#define DEFAULT_PROP_SERVER NULL +#define DEFAULT_PROP_CLIENT_NAME NULL enum { @@ -656,6 +661,7 @@ enum PROP_CONNECT, PROP_SERVER, PROP_CLIENT, + PROP_CLIENT_NAME, PROP_LAST }; @@ -731,6 +737,12 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) "The Jack server to connect to (NULL = default)", DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_CLIENT_NAME, + g_param_spec_string ("client-name", "Client name", + "The client name of the Jack instance (NULL = default)", + DEFAULT_PROP_CLIENT_NAME, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + g_object_class_install_property (gobject_class, PROP_CLIENT, g_param_spec_boxed ("client", "JackClient", "Handle for jack client", GST_TYPE_JACK_CLIENT, @@ -763,6 +775,7 @@ gst_jack_audio_src_init (GstJackAudioSrc * src, GstJackAudioSrcClass * gclass) src->ports = NULL; src->port_count = 0; src->buffers = NULL; + src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME); } static void @@ -781,6 +794,10 @@ gst_jack_audio_src_set_property (GObject * object, guint prop_id, GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); switch (prop_id) { + case PROP_CLIENT_NAME: + g_free (src->client_name); + src->client_name = g_value_dup_string (value); + break; case PROP_CONNECT: src->connect = g_value_get_enum (value); break; @@ -807,6 +824,9 @@ gst_jack_audio_src_get_property (GObject * object, guint prop_id, GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); switch (prop_id) { + case PROP_CLIENT_NAME: + g_value_set_string (value, src->client_name); + break; case PROP_CONNECT: g_value_set_enum (value, src->connect); break; diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h index 7e99b69df9..b69a4d6c49 100644 --- a/ext/jack/gstjackaudiosrc.h +++ b/ext/jack/gstjackaudiosrc.h @@ -75,6 +75,7 @@ struct _GstJackAudioSrc GstJackConnect connect; gchar *server; jack_client_t *jclient; + gchar *client_name; /* our client */ GstJackAudioClient *client; From 2e078fa556018c5c2f3c13d54aa6614622e5ecac Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Fri, 9 Dec 2011 15:50:28 +0000 Subject: [PATCH 02/65] jack: don't leak client name when freeing the element And add gtk-doc chunks for the new property. https://bugzilla.gnome.org/show_bug.cgi?id=665872 --- ext/jack/gstjackaudiosink.c | 13 +++++++++++++ ext/jack/gstjackaudiosrc.c | 13 +++++++++++++ 2 files changed, 26 insertions(+) diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index e1513b7e7e..078ca245d6 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -717,6 +717,13 @@ gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass) "The Jack server to connect to (NULL = default)", DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + /** + * GstJackAudioSink:client-name + * + * The client name to use. + * + * Since: 0.10.31 + */ g_object_class_install_property (gobject_class, PROP_CLIENT_NAME, g_param_spec_string ("client-name", "Client name", "The client name of the Jack instance (NULL = default)", @@ -760,6 +767,12 @@ gst_jack_audio_sink_dispose (GObject * object) GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object); gst_caps_replace (&sink->caps, NULL); + + if (sink->client_name != NULL) { + g_free (sink->client_name); + sink->client_name = NULL; + } + G_OBJECT_CLASS (parent_class)->dispose (object); } diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 469737d9e8..3c37970623 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -737,6 +737,13 @@ gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass) "The Jack server to connect to (NULL = default)", DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + /** + * GstJackAudioSrc:client-name + * + * The client name to use. + * + * Since: 0.10.31 + */ g_object_class_install_property (gobject_class, PROP_CLIENT_NAME, g_param_spec_string ("client-name", "Client name", "The client name of the Jack instance (NULL = default)", @@ -784,6 +791,12 @@ gst_jack_audio_src_dispose (GObject * object) GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object); gst_caps_replace (&src->caps, NULL); + + if (src->client_name != NULL) { + g_free (src->client_name); + src->client_name = NULL; + } + G_OBJECT_CLASS (parent_class)->dispose (object); } From d456fcc192f85d54bba7c891ab93a7b9edc64f3e Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 10 Dec 2011 02:21:02 +0000 Subject: [PATCH 03/65] po: update translations --- po/LINGUAS | 2 +- po/eo.po | 832 +++++++++++++++++++++++++++++++++++++++++++++++++++++ po/ja.po | 119 ++++---- po/lv.po | 33 ++- po/sr.po | 431 +++++++++++++-------------- 5 files changed, 1132 insertions(+), 285 deletions(-) create mode 100644 po/eo.po diff --git a/po/LINGUAS b/po/LINGUAS index 44f55d97ef..73d8b85b2d 100644 --- a/po/LINGUAS +++ b/po/LINGUAS @@ -1 +1 @@ -af az bg ca cs da de el en_GB es eu fi fr gl hu id it ja lt lv mt nb nl or pl pt_BR ro ru sk sl sq sr sv tr uk vi zh_CN zh_HK zh_TW +af az bg ca cs da de el en_GB eo es eu fi fr gl hu id it ja lt lv mt nb nl or pl pt_BR ro ru sk sl sq sr sv tr uk vi zh_CN zh_HK zh_TW diff --git a/po/eo.po b/po/eo.po new file mode 100644 index 0000000000..73f84a393c --- /dev/null +++ b/po/eo.po @@ -0,0 +1,832 @@ +# Esperanto translation for gst-plugins-good. +# Copyright (C) 2011 Free Software Foundation, Inc. +# This file is distributed under the same license as the gst-plugins-good package. +# Kristjan SCHMIDT , 2011. +# +msgid "" +msgstr "" +"Project-Id-Version: gst-plugins-good 0.10.28.2\n" +"Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" +"POT-Creation-Date: 2011-04-16 16:25+0100\n" +"PO-Revision-Date: 2011-06-04 21:48+0100\n" +"Last-Translator: Kristjan SCHMIDT \n" +"Language-Team: Esperanto \n" +"Language: eo\n" +"MIME-Version: 1.0\n" +"Content-Type: text/plain; charset=UTF-8\n" +"Content-Transfer-Encoding: 8bit\n" +"Plural-Forms: nplurals=2; plural=(n != 1);\n" + +#: ext/esd/esdsink.c:253 ext/esd/esdsink.c:358 +msgid "Could not establish connection to sound server" +msgstr "" + +#: ext/esd/esdsink.c:260 +msgid "Failed to query sound server capabilities" +msgstr "" + +#. TRANSLATORS: 'song title' by 'artist name' +#: ext/pulse/pulsesink.c:2686 +#, c-format +msgid "'%s' by '%s'" +msgstr "'%s' de '%s'" + +#: ext/flac/gstflacdec.c:1117 ext/libpng/gstpngdec.c:349 +#: ext/libpng/gstpngdec.c:360 ext/libpng/gstpngdec.c:559 +#: ext/wavpack/gstwavpackparse.c:1170 gst/avi/gstavidemux.c:5203 +msgid "Internal data stream error." +msgstr "Interna datumflu-eraro." + +#: ext/jpeg/gstjpegdec.c:286 +msgid "Failed to decode JPEG image" +msgstr "" + +#: ext/shout2/gstshout2.c:578 +msgid "Could not connect to server" +msgstr "Ne eblis konekti al servilo" + +#: ext/soup/gstsouphttpsrc.c:871 +msgid "Server does not support seeking." +msgstr "" + +#: ext/soup/gstsouphttpsrc.c:1078 +msgid "Could not resolve server name." +msgstr "" + +#: ext/soup/gstsouphttpsrc.c:1084 +msgid "Could not establish connection to server." +msgstr "" + +#: ext/soup/gstsouphttpsrc.c:1089 +msgid "Secure connection setup failed." +msgstr "" + +#: ext/soup/gstsouphttpsrc.c:1094 +msgid "A network error occured, or the server closed the connection unexpectedly." +msgstr "" + +#: ext/soup/gstsouphttpsrc.c:1100 +msgid "Server sent bad data." +msgstr "" + +#: ext/soup/gstsouphttpsrc.c:1249 +msgid "No URL set." +msgstr "" + +#: gst/avi/gstavimux.c:1810 +msgid "No or invalid input audio, AVI stream will be corrupt." +msgstr "" + +#: gst/quicktime/qtdemux.c:519 gst/quicktime/qtdemux.c:523 +msgid "This file contains no playable streams." +msgstr "" + +#: gst/quicktime/qtdemux.c:558 gst/quicktime/qtdemux.c:4009 +#: gst/quicktime/qtdemux.c:4071 gst/quicktime/qtdemux.c:4219 +msgid "This file is invalid and cannot be played." +msgstr "" + +#: gst/quicktime/qtdemux.c:2439 gst/quicktime/qtdemux.c:2515 +#: gst/quicktime/qtdemux.c:2558 gst/quicktime/qtdemux.c:4798 +#: gst/quicktime/qtdemux.c:4805 gst/quicktime/qtdemux.c:5391 +#: gst/quicktime/qtdemux.c:5817 gst/quicktime/qtdemux.c:5824 +#: gst/quicktime/qtdemux.c:7305 +msgid "This file is corrupt and cannot be played." +msgstr "" + +#: gst/quicktime/qtdemux.c:2647 +msgid "Invalid atom size." +msgstr "" + +#: gst/quicktime/qtdemux.c:2716 +msgid "This file is incomplete and cannot be played." +msgstr "" + +#: gst/quicktime/qtdemux.c:4994 +msgid "The video in this file might not play correctly." +msgstr "" + +#: gst/quicktime/qtdemux.c:7334 +#, c-format +msgid "This file contains too many streams. Only playing first %d" +msgstr "" + +#: gst/rtsp/gstrtspsrc.c:5187 +msgid "No supported stream was found. You might need to install a GStreamer RTSP extension plugin for Real media streams." +msgstr "" + +#: gst/rtsp/gstrtspsrc.c:5192 +msgid "No supported stream was found. You might need to allow more transport protocols or may otherwise be missing the right GStreamer RTSP extension plugin." +msgstr "" + +#: gst/wavparse/gstwavparse.c:2103 +msgid "Internal data flow error." +msgstr "" + +#: sys/oss/gstossmixertrack.c:98 sys/oss4/oss4-mixer.c:722 +#: sys/sunaudio/gstsunaudiomixertrack.c:69 +msgid "Volume" +msgstr "Laŭteco" + +#: sys/oss/gstossmixertrack.c:99 sys/oss4/oss4-mixer.c:735 +msgid "Bass" +msgstr "Baso" + +#: sys/oss/gstossmixertrack.c:100 sys/oss4/oss4-mixer.c:736 +msgid "Treble" +msgstr "" + +#: sys/oss/gstossmixertrack.c:101 +msgid "Synth" +msgstr "Sintezilo" + +#: sys/oss/gstossmixertrack.c:102 sys/oss4/oss4-mixer.c:750 +msgid "PCM" +msgstr "" + +#: sys/oss/gstossmixertrack.c:103 +msgid "Speaker" +msgstr "Parolilo" + +#: sys/oss/gstossmixertrack.c:104 +msgid "Line-in" +msgstr "" + +#: sys/oss/gstossmixertrack.c:105 sys/oss4/oss4-mixer.c:741 +msgid "Microphone" +msgstr "Mikrofono" + +#: sys/oss/gstossmixertrack.c:106 +msgid "CD" +msgstr "KD" + +#: sys/oss/gstossmixertrack.c:107 +msgid "Mixer" +msgstr "Miksilo" + +#: sys/oss/gstossmixertrack.c:108 +msgid "PCM-2" +msgstr "" + +#: sys/oss/gstossmixertrack.c:109 +msgid "Record" +msgstr "Registri" + +#: sys/oss/gstossmixertrack.c:110 +msgid "In-gain" +msgstr "" + +#: sys/oss/gstossmixertrack.c:111 +msgid "Out-gain" +msgstr "" + +#: sys/oss/gstossmixertrack.c:112 +msgid "Line-1" +msgstr "" + +#: sys/oss/gstossmixertrack.c:113 +msgid "Line-2" +msgstr "" + +#: sys/oss/gstossmixertrack.c:114 +msgid "Line-3" +msgstr "" + +#: sys/oss/gstossmixertrack.c:115 +msgid "Digital-1" +msgstr "" + +#: sys/oss/gstossmixertrack.c:116 +msgid "Digital-2" +msgstr "" + +#: sys/oss/gstossmixertrack.c:117 +msgid "Digital-3" +msgstr "" + +#: sys/oss/gstossmixertrack.c:118 +msgid "Phone-in" +msgstr "" + +#: sys/oss/gstossmixertrack.c:119 +msgid "Phone-out" +msgstr "" + +#: sys/oss/gstossmixertrack.c:120 +msgid "Video" +msgstr "Video" + +#: sys/oss/gstossmixertrack.c:121 +msgid "Radio" +msgstr "Radio" + +#: sys/oss/gstossmixertrack.c:122 sys/oss4/oss4-mixer.c:764 +#: sys/sunaudio/gstsunaudiomixertrack.c:71 +msgid "Monitor" +msgstr "Ekrano" + +#: sys/oss/gstosssink.c:399 sys/oss4/oss4-sink.c:494 +#: sys/oss4/oss4-source.c:361 +msgid "Could not open audio device for playback. Device is being used by another application." +msgstr "Ne eblis malfermi la sonaparaton por reproduktado. Ĝi estas uzate de alia aplikaĵo." + +#: sys/oss/gstosssink.c:406 sys/oss4/oss4-sink.c:504 +#: sys/oss4/oss4-source.c:371 +msgid "Could not open audio device for playback. You don't have permission to open the device." +msgstr "" + +#: sys/oss/gstosssink.c:414 sys/oss4/oss4-sink.c:515 +#: sys/oss4/oss4-source.c:382 +msgid "Could not open audio device for playback." +msgstr "Ne eblis malfermi la sonaparaton por reproduktado." + +#: sys/oss/gstosssrc.c:370 +msgid "Could not open audio device for recording. You don't have permission to open the device." +msgstr "" + +#: sys/oss/gstosssrc.c:378 +msgid "Could not open audio device for recording." +msgstr "Ne eblis malfermi sonaparaton por registrado." + +#: sys/oss4/oss4-mixer.c:302 +msgid "Could not open audio device for mixer control handling." +msgstr "" + +#: sys/oss4/oss4-mixer.c:316 +msgid "Could not open audio device for mixer control handling. This version of the Open Sound System is not supported by this element." +msgstr "" + +#: sys/oss4/oss4-mixer.c:723 +msgid "Master" +msgstr "Ĉefe" + +#: sys/oss4/oss4-mixer.c:724 +msgid "Front" +msgstr "Antaŭe" + +#: sys/oss4/oss4-mixer.c:725 +msgid "Rear" +msgstr "Malantaŭe" + +#: sys/oss4/oss4-mixer.c:726 +msgid "Headphones" +msgstr "Kaptelefono" + +#: sys/oss4/oss4-mixer.c:727 +msgid "Center" +msgstr "Centre" + +#: sys/oss4/oss4-mixer.c:728 +msgid "LFE" +msgstr "" + +#: sys/oss4/oss4-mixer.c:729 +msgid "Surround" +msgstr "Ĉirkaŭe" + +#: sys/oss4/oss4-mixer.c:730 +msgid "Side" +msgstr "Flanke" + +#: sys/oss4/oss4-mixer.c:731 sys/sunaudio/gstsunaudiomixertrack.c:72 +msgid "Built-in Speaker" +msgstr "" + +#: sys/oss4/oss4-mixer.c:732 sys/sunaudio/gstsunaudiomixertrack.c:76 +msgid "AUX 1 Out" +msgstr "" + +#: sys/oss4/oss4-mixer.c:733 sys/sunaudio/gstsunaudiomixertrack.c:77 +msgid "AUX 2 Out" +msgstr "" + +#: sys/oss4/oss4-mixer.c:734 +msgid "AUX Out" +msgstr "" + +#: sys/oss4/oss4-mixer.c:737 +msgid "3D Depth" +msgstr "" + +#: sys/oss4/oss4-mixer.c:738 +msgid "3D Center" +msgstr "" + +#: sys/oss4/oss4-mixer.c:739 +msgid "3D Enhance" +msgstr "" + +#: sys/oss4/oss4-mixer.c:740 +msgid "Telephone" +msgstr "" + +#: sys/oss4/oss4-mixer.c:742 sys/sunaudio/gstsunaudiomixertrack.c:74 +msgid "Line Out" +msgstr "" + +#: sys/oss4/oss4-mixer.c:743 sys/oss4/oss4-mixer.c:744 +msgid "Line In" +msgstr "" + +#: sys/oss4/oss4-mixer.c:745 +msgid "Internal CD" +msgstr "" + +#: sys/oss4/oss4-mixer.c:746 +msgid "Video In" +msgstr "" + +#: sys/oss4/oss4-mixer.c:747 +msgid "AUX 1 In" +msgstr "" + +#: sys/oss4/oss4-mixer.c:748 +msgid "AUX 2 In" +msgstr "" + +#: sys/oss4/oss4-mixer.c:749 +msgid "AUX In" +msgstr "" + +#: sys/oss4/oss4-mixer.c:751 sys/oss4/oss4-mixer.c:752 +msgid "Record Gain" +msgstr "" + +#: sys/oss4/oss4-mixer.c:753 +msgid "Output Gain" +msgstr "" + +#: sys/oss4/oss4-mixer.c:754 +msgid "Microphone Boost" +msgstr "" + +#: sys/oss4/oss4-mixer.c:755 +msgid "Loopback" +msgstr "" + +#: sys/oss4/oss4-mixer.c:756 +msgid "Diagnostic" +msgstr "" + +#: sys/oss4/oss4-mixer.c:757 +msgid "Bass Boost" +msgstr "" + +#: sys/oss4/oss4-mixer.c:758 +msgid "Playback Ports" +msgstr "" + +#: sys/oss4/oss4-mixer.c:759 +msgid "Input" +msgstr "Enigo" + +#: sys/oss4/oss4-mixer.c:760 sys/oss4/oss4-mixer.c:761 +msgid "Record Source" +msgstr "" + +#: sys/oss4/oss4-mixer.c:762 +msgid "Monitor Source" +msgstr "" + +#: sys/oss4/oss4-mixer.c:763 +msgid "Keyboard Beep" +msgstr "" + +#: sys/oss4/oss4-mixer.c:765 +msgid "Simulate Stereo" +msgstr "" + +#: sys/oss4/oss4-mixer.c:766 sys/oss4/oss4-mixer.c:786 +msgid "Stereo" +msgstr "Dukanale" + +#: sys/oss4/oss4-mixer.c:767 +msgid "Surround Sound" +msgstr "Ĉirkaŭa sono" + +#: sys/oss4/oss4-mixer.c:768 +msgid "Microphone Gain" +msgstr "" + +#: sys/oss4/oss4-mixer.c:769 +msgid "Speaker Source" +msgstr "" + +#: sys/oss4/oss4-mixer.c:770 +msgid "Microphone Source" +msgstr "" + +#: sys/oss4/oss4-mixer.c:771 +msgid "Jack" +msgstr "" + +#: sys/oss4/oss4-mixer.c:772 +msgid "Center / LFE" +msgstr "" + +#: sys/oss4/oss4-mixer.c:773 +msgid "Stereo Mix" +msgstr "" + +#: sys/oss4/oss4-mixer.c:774 +msgid "Mono Mix" +msgstr "" + +#: sys/oss4/oss4-mixer.c:775 +msgid "Input Mix" +msgstr "" + +#: sys/oss4/oss4-mixer.c:776 +msgid "SPDIF In" +msgstr "" + +#: sys/oss4/oss4-mixer.c:777 sys/sunaudio/gstsunaudiomixertrack.c:75 +msgid "SPDIF Out" +msgstr "" + +#: sys/oss4/oss4-mixer.c:778 +msgid "Microphone 1" +msgstr "Mikrofono 1" + +#: sys/oss4/oss4-mixer.c:779 +msgid "Microphone 2" +msgstr "Mikrofono 2" + +#: sys/oss4/oss4-mixer.c:780 +msgid "Digital Out" +msgstr "" + +#: sys/oss4/oss4-mixer.c:781 +msgid "Digital In" +msgstr "" + +#: sys/oss4/oss4-mixer.c:782 +msgid "HDMI" +msgstr "" + +#: sys/oss4/oss4-mixer.c:783 +msgid "Modem" +msgstr "Modemo" + +#: sys/oss4/oss4-mixer.c:784 +msgid "Handset" +msgstr "" + +#: sys/oss4/oss4-mixer.c:785 +msgid "Other" +msgstr "Alia" + +#: sys/oss4/oss4-mixer.c:787 +msgid "None" +msgstr "Neniu" + +#: sys/oss4/oss4-mixer.c:788 +msgid "On" +msgstr "Enŝaltite" + +#: sys/oss4/oss4-mixer.c:789 +msgid "Off" +msgstr "Elŝaltite" + +#: sys/oss4/oss4-mixer.c:790 +msgid "Mute" +msgstr "Silentigi" + +#: sys/oss4/oss4-mixer.c:791 +msgid "Fast" +msgstr "Rapide" + +#. TRANSLATORS: "Very Low" is a quality setting here +#: sys/oss4/oss4-mixer.c:793 +msgid "Very Low" +msgstr "Tre malalte" + +#. TRANSLATORS: "Low" is a quality setting here +#: sys/oss4/oss4-mixer.c:795 +msgid "Low" +msgstr "Malalte" + +#. TRANSLATORS: "Medium" is a quality setting here +#: sys/oss4/oss4-mixer.c:797 +msgid "Medium" +msgstr "Meze" + +#. TRANSLATORS: "High" is a quality setting here +#: sys/oss4/oss4-mixer.c:799 +msgid "High" +msgstr "Alte" + +#. TRANSLATORS: "Very High" is a quality setting here +#: sys/oss4/oss4-mixer.c:801 sys/oss4/oss4-mixer.c:802 +msgid "Very High" +msgstr "Tre alte" + +#. TRANSLATORS: "Production" is a quality setting here +#: sys/oss4/oss4-mixer.c:804 +msgid "Production" +msgstr "" + +#: sys/oss4/oss4-mixer.c:805 +msgid "Front Panel Microphone" +msgstr "" + +#: sys/oss4/oss4-mixer.c:806 +msgid "Front Panel Line In" +msgstr "" + +#: sys/oss4/oss4-mixer.c:807 +msgid "Front Panel Headphones" +msgstr "" + +#: sys/oss4/oss4-mixer.c:808 +msgid "Front Panel Line Out" +msgstr "" + +#: sys/oss4/oss4-mixer.c:809 +msgid "Green Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:810 +msgid "Pink Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:811 +msgid "Blue Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:812 +msgid "White Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:813 +msgid "Black Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:814 +msgid "Gray Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:815 +msgid "Orange Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:816 +msgid "Red Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:817 +msgid "Yellow Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:818 +msgid "Green Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:819 +msgid "Pink Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:820 +msgid "Blue Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:821 +msgid "White Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:822 +msgid "Black Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:823 +msgid "Gray Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:824 +msgid "Orange Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:825 +msgid "Red Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:826 +msgid "Yellow Front Panel Connector" +msgstr "" + +#: sys/oss4/oss4-mixer.c:827 +msgid "Spread Output" +msgstr "" + +#: sys/oss4/oss4-mixer.c:828 +msgid "Downmix" +msgstr "" + +#: sys/oss4/oss4-mixer.c:872 +msgid "Virtual Mixer Input" +msgstr "" + +#: sys/oss4/oss4-mixer.c:874 +msgid "Virtual Mixer Output" +msgstr "" + +#: sys/oss4/oss4-mixer.c:876 +msgid "Virtual Mixer Channels" +msgstr "" + +#. TRANSLATORS: name + number of a volume mixer control +#: sys/oss4/oss4-mixer.c:927 +#, c-format +msgid "%s %d Function" +msgstr "%s %d funkcio" + +#. TRANSLATORS: name of a volume mixer control +#: sys/oss4/oss4-mixer.c:934 +#, c-format +msgid "%s Function" +msgstr "%s funcio" + +#: sys/oss4/oss4-sink.c:524 sys/oss4/oss4-source.c:392 +msgid "Could not open audio device for playback. This version of the Open Sound System is not supported by this element." +msgstr "" + +#: sys/oss4/oss4-sink.c:640 +msgid "Playback is not supported by this audio device." +msgstr "" + +#: sys/oss4/oss4-sink.c:647 +msgid "Audio playback error." +msgstr "" + +#: sys/oss4/oss4-source.c:514 +msgid "Recording is not supported by this audio device." +msgstr "" + +#: sys/oss4/oss4-source.c:521 +msgid "Error recording from audio device." +msgstr "" + +#: sys/sunaudio/gstsunaudiomixertrack.c:70 +msgid "Gain" +msgstr "" + +#: sys/sunaudio/gstsunaudiomixertrack.c:73 +msgid "Headphone" +msgstr "" + +#: sys/v4l2/gstv4l2src.c:887 +#, c-format +msgid "Error reading %d bytes from device '%s'." +msgstr "" + +#: sys/v4l2/gstv4l2src.c:913 +#, c-format +msgid "Got unexpected frame size of %u instead of %u." +msgstr "" + +#: sys/v4l2/gstv4l2src.c:931 +#, c-format +msgid "Error reading %d bytes on device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:99 +#, c-format +msgid "Error getting capabilities for device '%s': It isn't a v4l2 driver. Check if it is a v4l1 driver." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:139 +#, c-format +msgid "Failed to query attributes of input %d in device %s" +msgstr "" + +#: sys/v4l2/v4l2_calls.c:169 +#, c-format +msgid "Failed to get setting of tuner %d on device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:213 +#, c-format +msgid "Failed to query norm on device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:264 sys/v4l2/v4l2_calls.c:347 +#, c-format +msgid "Failed getting controls attributes on device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:492 +#, c-format +msgid "Cannot identify device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:499 +#, c-format +msgid "This isn't a device '%s'." +msgstr "Tio ne estas '%s'-aparato." + +#: sys/v4l2/v4l2_calls.c:506 +#, c-format +msgid "Could not open device '%s' for reading and writing." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:513 +#, c-format +msgid "Device '%s' is not a capture device." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:522 +#, c-format +msgid "Device '%s' is not a output device." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:622 +#, c-format +msgid "Failed to set norm for device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:660 +#, c-format +msgid "Failed to get current tuner frequency for device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:702 +#, c-format +msgid "Failed to set current tuner frequency for device '%s' to %lu Hz." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:736 +#, c-format +msgid "Failed to get signal strength for device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:772 +#, c-format +msgid "Failed to get value for control %d on device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:807 +#, c-format +msgid "Failed to set value %d for control %d on device '%s'." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:839 +#, c-format +msgid "Failed to get current input on device '%s'. May be it is a radio device" +msgstr "" + +#: sys/v4l2/v4l2_calls.c:864 +#, c-format +msgid "Failed to set input %d on device %s." +msgstr "" + +#: sys/v4l2/v4l2_calls.c:896 +#, c-format +msgid "Failed to get current output on device '%s'. May be it is a radio device" +msgstr "" + +#: sys/v4l2/v4l2_calls.c:921 +#, c-format +msgid "Failed to set output %d on device %s." +msgstr "" + +#: sys/v4l2/v4l2src_calls.c:81 +#, c-format +msgid "Could not enqueue buffers in device '%s'." +msgstr "" + +#: sys/v4l2/v4l2src_calls.c:197 +#, c-format +msgid "Failed trying to get video frames from device '%s'." +msgstr "" + +#: sys/v4l2/v4l2src_calls.c:199 +#, c-format +msgid "Failed after %d tries. device %s. system error: %s" +msgstr "" + +#: sys/v4l2/v4l2src_calls.c:239 +#, c-format +msgid "Could not get parameters on device '%s'" +msgstr "" + +#: sys/v4l2/v4l2src_calls.c:267 +msgid "Video input device did not accept new frame rate setting." +msgstr "" + +#: sys/v4l2/v4l2src_calls.c:339 +#, c-format +msgid "Could not map buffers from device '%s'" +msgstr "" + +#: sys/v4l2/v4l2src_calls.c:347 +#, c-format +msgid "The driver of device '%s' does not support any known capture method." +msgstr "" + +#: sys/ximage/gstximagesrc.c:719 +msgid "Changing resolution at runtime is not yet supported." +msgstr "" + +#: sys/ximage/gstximagesrc.c:733 +msgid "Cannot operate without a clock" +msgstr "" diff --git a/po/ja.po b/po/ja.po index 18c09ad94c..1a5899ed4c 100644 --- a/po/ja.po +++ b/po/ja.po @@ -3,13 +3,14 @@ # # Takao Fujiwara , 2006. # Makoto Kato , 2009-2011. +# Takeshi Hamasaki , 2011 msgid "" msgstr "" "Project-Id-Version: gst-plugins-good 0.10.28.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-05-10 09:38+0100\n" -"PO-Revision-Date: 2011-04-26 20:15+0900\n" -"Last-Translator: Makoto Kato \n" +"POT-Creation-Date: 2011-12-10 02:18+0000\n" +"PO-Revision-Date: 2011-08-28 23:59+0900\n" +"Last-Translator: Takeshi Hamasaki \n" "Language-Team: Japanese \n" "Language: ja\n" "MIME-Version: 1.0\n" @@ -25,9 +26,9 @@ msgid "Failed to query sound server capabilities" msgstr "サウンドサーバーのケイパビリティのクエリーに失敗しました" #. TRANSLATORS: 'song title' by 'artist name' -#, c-format +#, fuzzy, c-format msgid "'%s' by '%s'" -msgstr "" +msgstr "対" msgid "Internal data stream error." msgstr "内部データストリームエラー" @@ -54,8 +55,9 @@ msgid "" "A network error occured, or the server closed the connection unexpectedly." msgstr "ネットワークエラーが発生したか、サーバーが予期せず接続を閉じました。" +#, fuzzy msgid "Server sent bad data." -msgstr "" +msgstr "データストリーム (サーバープッシュ型)" msgid "No URL set." msgstr "URLが指定されていません。" @@ -95,14 +97,14 @@ msgstr "" "サポートしているストリームが見つかりません。Real メディアストリームのための" "GStreamer RTSP 拡張プラグインをインストールする必要があるかもしれません。" -#, fuzzy msgid "" "No supported stream was found. You might need to allow more transport " "protocols or may otherwise be missing the right GStreamer RTSP extension " "plugin." msgstr "" -"サポートしているストリームが見つかりません。正しい GStreamer RTSP 拡張プラグ" -"インをインストールする必要があるかもしれません。" +"サポートしているストリームが見つかりません。別の転送プロトコルをインストール" +"する必要があるかもしれません。または、正しい GStreamer RTSP 拡張プラグインが" +"ないのかもしれません。" msgid "Internal data flow error." msgstr "内部データフローエラー。" @@ -282,14 +284,16 @@ msgstr "AUX 2出力" msgid "AUX Out" msgstr "AUX出力" +#, fuzzy msgid "3D Depth" -msgstr "" +msgstr "最大深度: " msgid "3D Center" msgstr "3Dセンター" +#, fuzzy msgid "3D Enhance" -msgstr "" +msgstr "色を強調しています" msgid "Telephone" msgstr "電話" @@ -328,7 +332,7 @@ msgid "Loopback" msgstr "ループバック" msgid "Diagnostic" -msgstr "" +msgstr "診断" msgid "Bass Boost" msgstr "低音ブースト" @@ -348,8 +352,9 @@ msgstr "モニターソース" msgid "Keyboard Beep" msgstr "キーボードビープ音" +#, fuzzy msgid "Simulate Stereo" -msgstr "" +msgstr "彫金を表現します" msgid "Stereo" msgstr "ステレオ" @@ -423,32 +428,39 @@ msgstr "オフ" msgid "Mute" msgstr "ミュート" +#, fuzzy msgid "Fast" -msgstr "" +msgstr "速い" #. TRANSLATORS: "Very Low" is a quality setting here +#, fuzzy msgid "Very Low" -msgstr "" +msgstr "低品位" #. TRANSLATORS: "Low" is a quality setting here +#, fuzzy msgid "Low" -msgstr "" +msgstr "低音" #. TRANSLATORS: "Medium" is a quality setting here +#, fuzzy msgid "Medium" -msgstr "" +msgstr "中音" #. TRANSLATORS: "High" is a quality setting here +#, fuzzy msgid "High" -msgstr "" +msgstr "高音" #. TRANSLATORS: "Very High" is a quality setting here +#, fuzzy msgid "Very High" -msgstr "" +msgstr "高品位" #. TRANSLATORS: "Production" is a quality setting here +#, fuzzy msgid "Production" -msgstr "" +msgstr "プロダクション" msgid "Front Panel Microphone" msgstr "フロントパネルのマイクロフォン" @@ -516,8 +528,9 @@ msgstr "" msgid "Yellow Front Panel Connector" msgstr "" +#, fuzzy msgid "Spread Output" -msgstr "" +msgstr "情報出力:\n" msgid "Downmix" msgstr "ダウンミックス" @@ -532,35 +545,33 @@ msgid "Virtual Mixer Channels" msgstr "仮想ミキサーチャンネル" #. TRANSLATORS: name + number of a volume mixer control -#, c-format +#, fuzzy, c-format msgid "%s %d Function" -msgstr "" +msgstr "機能" #. TRANSLATORS: name of a volume mixer control -#, c-format +#, fuzzy, c-format msgid "%s Function" -msgstr "" +msgstr "機能" -#, fuzzy msgid "" "Could not open audio device for playback. This version of the Open Sound " "System is not supported by this element." msgstr "" -"再生用にオーディオデバイスを開くことができませんでした。デバイスを開くための" -"権限がありません。" +"再生用にオーディオデバイスを開くことができませんでした。このバージョンの " +"Open Sound System は、このエレメントによってサポートされていません。" msgid "Playback is not supported by this audio device." -msgstr "" +msgstr "再生はこのオーディオデバイスではサポートされていません。" msgid "Audio playback error." -msgstr "" +msgstr "オーディオ再生エラーです。" msgid "Recording is not supported by this audio device." msgstr "このオーディオデバイスによって録音はサポートされていません。" -#, fuzzy msgid "Error recording from audio device." -msgstr "デバイス %2$s から　%1$d バイト読み込み時にエラーが発生しました。" +msgstr "オーディオデバイスからの録音時にエラーが発生しました。" # SUN REVIEWED msgid "Gain" @@ -579,15 +590,15 @@ msgstr "予期しないフレームサイズ (%2$u ではなく %1$u) を取得 #, c-format msgid "Error reading %d bytes on device '%s'." -msgstr "デバイス '%2$s' 上で %1$d バイトの読み込み時ににエラーが発生しました。" +msgstr "デバイス '%2$s' 上で %1$d バイトの読み込み時にエラーが発生しました。" #, c-format msgid "" "Error getting capabilities for device '%s': It isn't a v4l2 driver. Check if " "it is a v4l1 driver." msgstr "" -"デバイス '%s' ケイパビリティの取得時にエラーが発生しました: v412 ドライバでは" -"ありません。もし v411 ドライバであれば、チェックしてください。" +"デバイス '%s' ケイパビリティの取得時にエラーが発生しました: v4l2 ドライバでは" +"ありません。v4l1 ドライバでないか、チェックしてください。" #, c-format msgid "Failed to query attributes of input %d in device %s" @@ -597,17 +608,17 @@ msgstr "デバイス %2$s 内の入力 %1$d の属性のクエリーに失敗し msgid "Failed to get setting of tuner %d on device '%s'." msgstr "デバイス '%2$s' 上のチューナー %1$d の設定の取得に失敗しました。" -#, c-format +#, fuzzy, c-format msgid "Failed to query norm on device '%s'." -msgstr "" +msgstr "デバイス '%2$s' 上のチューナー %1$d の設定の取得に失敗しました。" #, c-format msgid "Failed getting controls attributes on device '%s'." msgstr "デバイス '%s' のコントロール属性の取得に失敗しました" -#, c-format +#, fuzzy, c-format msgid "Cannot identify device '%s'." -msgstr "" +msgstr "ミキサーデバイス'%s'を開けません" #, c-format msgid "This isn't a device '%s'." @@ -626,9 +637,11 @@ msgstr "デバイス '%s' はキャプチャデバイスではありません" msgid "Device '%s' is not a output device." msgstr "デバイス '%s' は出力デバイスではありません" -#, c-format +#, fuzzy, c-format msgid "Failed to set norm for device '%s'." msgstr "" +"デバイス '%3$s' 上のコントロール %2$d の値を %1$d へ設定することに失敗しまし" +"た。" #, c-format msgid "Failed to get current tuner frequency for device '%s'." @@ -638,9 +651,9 @@ msgstr "デバイス '%s' の現在のチューナーの周波数の取得に失 msgid "Failed to set current tuner frequency for device '%s' to %lu Hz." msgstr "デバイス '%s' のチューナーの周波数の %lu Hz への設定に失敗しました" -#, c-format +#, fuzzy, c-format msgid "Failed to get signal strength for device '%s'." -msgstr "" +msgstr "デバイス '%s' の現在のチューナーの周波数の取得に失敗しました" #, c-format msgid "Failed to get value for control %d on device '%s'." @@ -658,24 +671,26 @@ msgstr "" "デバイス '%s' 上の現在の入力の取得に失敗しました。ラジオデバイスかもしれませ" "ん" -#, c-format +#, fuzzy, c-format msgid "Failed to set input %d on device %s." msgstr "" +"デバイス '%3$s' 上のコントロール %2$d の値を %1$d へ設定することに失敗しまし" +"た。" -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" -"デバイス '%s' 上の現在の入力の取得に失敗しました。ラジオデバイスかもしれませ" +"デバイス '%s' 上の現在の出力の取得に失敗しました。ラジオデバイスかもしれませ" "ん" -#, fuzzy, c-format -msgid "Failed to set output %d on device %s." -msgstr "デバイス '%2$s' 上のチューナー %1$d の設定の取得に失敗しました。" - #, c-format +msgid "Failed to set output %d on device %s." +msgstr "出力%dをデバイス %s に設定できませんでした。" + +#, fuzzy, c-format msgid "Could not enqueue buffers in device '%s'." -msgstr "" +msgstr "デバイス '%s' からバッファをマップできません。" #, c-format msgid "Failed trying to get video frames from device '%s'." @@ -704,10 +719,10 @@ msgstr "" "ません" msgid "Changing resolution at runtime is not yet supported." -msgstr "" +msgstr "実行時の解像度変更はサポートされていません。" msgid "Cannot operate without a clock" -msgstr "" +msgstr "クロックなしでは動作できません。" #~ msgid "" #~ "Failed trying to get video frames from device '%s'. Not enough memory." diff --git a/po/lv.po b/po/lv.po index ef270e41c5..15affb3721 100644 --- a/po/lv.po +++ b/po/lv.po @@ -3,12 +3,13 @@ # # Arvis Lācis , 2009. # Rihards Prieditis , 2010. +# Rihards Prieditis , 2011. msgid "" msgstr "" -"Project-Id-Version: gst-plugins-good 0.10.23.2\n" +"Project-Id-Version: gst-plugins-good 0.10.28.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-05-10 09:38+0100\n" -"PO-Revision-Date: 2010-07-07 11:53+0100\n" +"POT-Creation-Date: 2011-12-10 02:18+0000\n" +"PO-Revision-Date: 2011-09-02 11:23-0000\n" "Last-Translator: Rihards Priedītis \n" "Language-Team: Latvian \n" "Language: lv\n" @@ -40,28 +41,26 @@ msgid "Could not connect to server" msgstr "Nevar savienoties ar serveri" msgid "Server does not support seeking." -msgstr "" +msgstr "Serveris neatbalsta meklēšanu." -#, fuzzy msgid "Could not resolve server name." -msgstr "Nevar savienoties ar serveri" +msgstr "Nevar atpazīt servera nosaukumu." -#, fuzzy msgid "Could not establish connection to server." -msgstr "Nevar izveidot savienojumu ar skaņas serveri" +msgstr "Nevar izveidot savienojumu ar serveri." msgid "Secure connection setup failed." -msgstr "" +msgstr "Drošā savienojuma izveidošana neizdevās." msgid "" "A network error occured, or the server closed the connection unexpectedly." -msgstr "" +msgstr "Notika tīkla kļūda, vai serveris negaidīti aizvēra savienojumu." msgid "Server sent bad data." -msgstr "" +msgstr "Serveris nosūtija sliktus datus." msgid "No URL set." -msgstr "" +msgstr "URL nav uzstādīts." msgid "No or invalid input audio, AVI stream will be corrupt." msgstr "Nav neviena vai nederīgs ievades audio, AVI straume tiks bojāta." @@ -76,7 +75,7 @@ msgid "This file is corrupt and cannot be played." msgstr "Šis fails ir bojāts un nevar tikt atskaņots." msgid "Invalid atom size." -msgstr "" +msgstr "Nederīgs atoma izmērs." msgid "This file is incomplete and cannot be played." msgstr "Šis pails ir nepabeigts un nevar tikt atskaņots." @@ -629,16 +628,16 @@ msgstr "" msgid "Failed to set input %d on device %s." msgstr "Neizdevās uzstādīt ievadi %d uz ierīces %s." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" msgstr "" -"Neizdevās saņemt pašreizējo ievadi no ierīces \"%s\". Iespējams tā ir radio " +"Neizdevās saņemt pašreizējo izvadu no ierīces \"%s\". Iespējams tā ir radio " "ierīce" -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Neizdevās uzstādīt ievadi %d uz ierīces %s." +msgstr "Neizdevās uzstādīt izvadu %d uz ierīces %s." #, c-format msgid "Could not enqueue buffers in device '%s'." diff --git a/po/sr.po b/po/sr.po index 2aec60b304..141138d2f9 100644 --- a/po/sr.po +++ b/po/sr.po @@ -1,105 +1,110 @@ # Serbian translation of gst-plugins # Copyright (C) 2004 Free Software Foundation, Inc. +# This file is distributed under the same license as the gst-plugins-good package. # Danilo Segan , 2004. -# +# Мирослав Николић , 2011. msgid "" msgstr "" -"Project-Id-Version: gst-plugins 0.7.6\n" +"Project-Id-Version: gst-plugins-good-0.10.28.2\n" "Report-Msgid-Bugs-To: http://bugzilla.gnome.org/\n" -"POT-Creation-Date: 2011-05-10 09:38+0100\n" -"PO-Revision-Date: 2004-03-13 00:18+0100\n" -"Last-Translator: Danilo Segan \n" +"POT-Creation-Date: 2011-12-10 02:18+0000\n" +"PO-Revision-Date: 2011-12-04 16:44+0200\n" +"Last-Translator: Мирослав Николић \n" "Language-Team: Serbian \n" "Language: sr\n" "MIME-Version: 1.0\n" "Content-Type: text/plain; charset=UTF-8\n" "Content-Transfer-Encoding: 8bit\n" -"Plural-Forms: nplurals=3; plural=n%10==1 && n%100!=11 ? 0 : (n%10>=2 && n" +"Plural-Forms: nplurals=3; plural=(n%10==1 && n%100!=11 ? 0 : n%10>=2 && n" "%10<=4 && (n%100<10 || n%100>=20) ? 1 : 2);\n" +"X-Generator: Virtaal 0.7.0\n" +"X-Project-Style: gnome\n" msgid "Could not establish connection to sound server" -msgstr "" +msgstr "Не могу да успоставим везу са сервером звука" msgid "Failed to query sound server capabilities" -msgstr "" +msgstr "Нисам успео да испитам могућности сервера звука" #. TRANSLATORS: 'song title' by 'artist name' #, c-format msgid "'%s' by '%s'" -msgstr "" +msgstr "„%s“ изводи „%s“" msgid "Internal data stream error." -msgstr "" +msgstr "Унутрашња грешка тока података." msgid "Failed to decode JPEG image" -msgstr "" +msgstr "Нисам успео да декодирам ЈПЕГ слику" -#, fuzzy msgid "Could not connect to server" -msgstr "Не могу да затворим управљачки уређај „%s“." +msgstr "Не могу да се повежем са сервером" msgid "Server does not support seeking." -msgstr "" +msgstr "Сервер не подржава позиционирање." -#, fuzzy msgid "Could not resolve server name." -msgstr "Не могу да затворим управљачки уређај „%s“." +msgstr "Не могу да решим назив сервера." -#, fuzzy msgid "Could not establish connection to server." -msgstr "Не могу да затворим управљачки уређај „%s“." +msgstr "Не могу да успоставим везу са сервером." msgid "Secure connection setup failed." -msgstr "" +msgstr "Подешавање безбедне везе није успело." msgid "" "A network error occured, or the server closed the connection unexpectedly." -msgstr "" +msgstr "Дошло је до грешке на мрежи, или је сервер неочекивано затворио везу." msgid "Server sent bad data." -msgstr "" +msgstr "Сервер је послао лоше податке." msgid "No URL set." -msgstr "" +msgstr "Није подешена адреса." msgid "No or invalid input audio, AVI stream will be corrupt." -msgstr "Улазног звука нема или је неисправан, АВИ ток ће бити искварен." +msgstr "Улазног звука нема или је неисправан, АВИ ток ће бити оштећен." msgid "This file contains no playable streams." -msgstr "" +msgstr "Ова датотека не садржи токове за пуштање." msgid "This file is invalid and cannot be played." -msgstr "" +msgstr "Ова датотека је неисправна и не може бити пуштена." msgid "This file is corrupt and cannot be played." -msgstr "" +msgstr "Ова датотека је оштећена и не може бити пуштена." msgid "Invalid atom size." -msgstr "" +msgstr "Неисправна величина атома." msgid "This file is incomplete and cannot be played." -msgstr "" +msgstr "Ова датотека је непотпуна и не може бити пуштена." msgid "The video in this file might not play correctly." -msgstr "" +msgstr "Видео у овој датотеци можда неће бити пуштен исправно." #, c-format msgid "This file contains too many streams. Only playing first %d" -msgstr "" +msgstr "Ова датотека садржи превише токова. Пуштам само први %d" msgid "" "No supported stream was found. You might need to install a GStreamer RTSP " "extension plugin for Real media streams." msgstr "" +"Није пронађен ниједан подржани ток. Можда ћете морати да инсталирате " +"прикључак РТСП проширења Гстримера за токове Стварног медија." msgid "" "No supported stream was found. You might need to allow more transport " "protocols or may otherwise be missing the right GStreamer RTSP extension " "plugin." msgstr "" +"Није пронађен ниједан подржани ток. Можда ћете морати да омогућите више " +"протокола преноса или можда на неки други начин недостаје прави прикључак " +"РТСП проширења ГСтримера." msgid "Internal data flow error." -msgstr "" +msgstr "Унутрaшња грешка протока података." msgid "Volume" msgstr "Јачина звука" @@ -108,10 +113,10 @@ msgid "Bass" msgstr "Бас" msgid "Treble" -msgstr "Шум" +msgstr "Високотонац" msgid "Synth" -msgstr "Синт." +msgstr "Синтисајзер" msgid "PCM" msgstr "ПЦМ" @@ -120,7 +125,7 @@ msgid "Speaker" msgstr "Звучник" msgid "Line-in" -msgstr "Ул.лин." +msgstr "Линијски улаз" msgid "Microphone" msgstr "Микрофон" @@ -129,7 +134,7 @@ msgid "CD" msgstr "ЦД" msgid "Mixer" -msgstr "Миксер" +msgstr "Мешач" msgid "PCM-2" msgstr "ПЦМ-2" @@ -138,34 +143,34 @@ msgid "Record" msgstr "Снимање" msgid "In-gain" -msgstr "Ул. пој." +msgstr "Улазно појачање" msgid "Out-gain" -msgstr "Из. пој." +msgstr "Излазно појачање" msgid "Line-1" -msgstr "Лин. 1" +msgstr "Линија 1" msgid "Line-2" -msgstr "Лин. 2" +msgstr "Линија 2" msgid "Line-3" -msgstr "Лин. 3" +msgstr "Линија 3" msgid "Digital-1" -msgstr "Диг. 1" +msgstr "Дигитални 1" msgid "Digital-2" -msgstr "Диг. 2" +msgstr "Дигитални 2" msgid "Digital-3" -msgstr "Диг. 3" +msgstr "Дигитални 3" msgid "Phone-in" -msgstr "Тел. ул." +msgstr "Телефонски улаз" msgid "Phone-out" -msgstr "Тел. из." +msgstr "Телефонски излаз" msgid "Video" msgstr "Видео" @@ -180,497 +185,493 @@ msgid "" "Could not open audio device for playback. Device is being used by another " "application." msgstr "" +"Не могу да покренем звучни уређај ради пуштања. Уређај користи нека друга " +"апликација." msgid "" "Could not open audio device for playback. You don't have permission to open " "the device." msgstr "" +"Не могу да покренем аудио уређај ради пуштања. Немате овлашћење за покретање " +"уређаја." -#, fuzzy msgid "Could not open audio device for playback." -msgstr "Не могу да отворим звучни уређај „%s“ ради уписа." +msgstr "Не могу да покренем звучни уређај ради пуштања." msgid "" "Could not open audio device for recording. You don't have permission to open " "the device." msgstr "" +"Не могу да покренем аудио уређај ради снимања. Немате овлашћење за покретање " +"уређаја." -#, fuzzy msgid "Could not open audio device for recording." -msgstr "Не могу да отворим ЦД уређај ради читања." +msgstr "Не могу да покренем звучни уређај ради снимања." -#, fuzzy msgid "Could not open audio device for mixer control handling." -msgstr "Не могу да отворим звучни уређај „%s“ ради уписа." +msgstr "Не могу да покренем звучни уређај за руковање управљањем мешача." msgid "" "Could not open audio device for mixer control handling. This version of the " "Open Sound System is not supported by this element." msgstr "" +"Не могу да покренем звучни уређај за руковање управљањем мешача. Ово издање " +"система отвореног звука није подржано овим елементом." msgid "Master" -msgstr "" +msgstr "Главни" msgid "Front" -msgstr "" +msgstr "Предњи" -#, fuzzy msgid "Rear" -msgstr "Снимање" +msgstr "Задњи" msgid "Headphones" -msgstr "" +msgstr "Слушалице" msgid "Center" -msgstr "" +msgstr "Средињи" msgid "LFE" -msgstr "" +msgstr "ЛФЕ" msgid "Surround" -msgstr "" +msgstr "Окружење" -#, fuzzy msgid "Side" -msgstr "Видео" +msgstr "Са стране" -#, fuzzy msgid "Built-in Speaker" -msgstr "Звучник" +msgstr "Уграђени звучник" msgid "AUX 1 Out" -msgstr "" +msgstr "АУХ 1 излаз" msgid "AUX 2 Out" -msgstr "" +msgstr "АУХ 2 излаз" msgid "AUX Out" -msgstr "" +msgstr "АУХ излаз" msgid "3D Depth" -msgstr "" +msgstr "3Д дубина" msgid "3D Center" -msgstr "" +msgstr "3Д средиште" msgid "3D Enhance" -msgstr "" +msgstr "3Д побољшање" msgid "Telephone" -msgstr "" +msgstr "Телефон" msgid "Line Out" -msgstr "" +msgstr "Линијски излаз" -#, fuzzy msgid "Line In" -msgstr "Ул.лин." +msgstr "Линијски улаз" msgid "Internal CD" -msgstr "" +msgstr "Унутрашњи ЦД" -#, fuzzy msgid "Video In" -msgstr "Видео" +msgstr "Видео улаз" msgid "AUX 1 In" -msgstr "" +msgstr "АУХ 1 улаз" msgid "AUX 2 In" -msgstr "" +msgstr "АУХ 2 улаз" msgid "AUX In" -msgstr "" +msgstr "АУХ улаз" -#, fuzzy msgid "Record Gain" -msgstr "Снимање" +msgstr "Појачање снимања" -#, fuzzy msgid "Output Gain" -msgstr "Из. пој." +msgstr "Појачање излаза" -#, fuzzy msgid "Microphone Boost" -msgstr "Микрофон" +msgstr "Појачање микрофона" msgid "Loopback" -msgstr "" +msgstr "Повратна петља" msgid "Diagnostic" -msgstr "" +msgstr "Дијагностика" msgid "Bass Boost" -msgstr "" +msgstr "Појачање баса" msgid "Playback Ports" -msgstr "" +msgstr "Портови пуштања" msgid "Input" -msgstr "" +msgstr "Улаз" -#, fuzzy msgid "Record Source" -msgstr "Снимање" +msgstr "Извор снимања" -#, fuzzy msgid "Monitor Source" -msgstr "Праћење" +msgstr "Извор праћења" msgid "Keyboard Beep" -msgstr "" +msgstr "Звук тастатуре" msgid "Simulate Stereo" -msgstr "" +msgstr "Лажни стерео" msgid "Stereo" -msgstr "" +msgstr "Стерео" msgid "Surround Sound" -msgstr "" +msgstr "Звук окружења" -#, fuzzy msgid "Microphone Gain" -msgstr "Микрофон" +msgstr "Појачање микрофона" -#, fuzzy msgid "Speaker Source" -msgstr "Звучник" +msgstr "Извор звучника" -#, fuzzy msgid "Microphone Source" -msgstr "Микрофон" +msgstr "Извор микрофона" msgid "Jack" -msgstr "" +msgstr "Утичница" msgid "Center / LFE" -msgstr "" +msgstr "Средиште / ЛФЕ" msgid "Stereo Mix" -msgstr "" +msgstr "Стерео микс" msgid "Mono Mix" -msgstr "" +msgstr "Моно микс" msgid "Input Mix" -msgstr "" +msgstr "Улазни микс" msgid "SPDIF In" -msgstr "" +msgstr "СПДИФ улаз" msgid "SPDIF Out" -msgstr "" +msgstr "СПДИФ излаз" -#, fuzzy msgid "Microphone 1" -msgstr "Микрофон" +msgstr "Микрофон 1" -#, fuzzy msgid "Microphone 2" -msgstr "Микрофон" +msgstr "Микрофон 2" -#, fuzzy msgid "Digital Out" -msgstr "Диг. 1" +msgstr "Дигитални излаз" -#, fuzzy msgid "Digital In" -msgstr "Диг. 1" +msgstr "Дигитални улаз" msgid "HDMI" -msgstr "" +msgstr "ХДМИ" msgid "Modem" -msgstr "" +msgstr "Модем" msgid "Handset" -msgstr "" +msgstr "Слушалица" msgid "Other" -msgstr "" +msgstr "Остало" msgid "None" -msgstr "" +msgstr "Ништа" msgid "On" -msgstr "" +msgstr "Укљ." msgid "Off" -msgstr "" +msgstr "Искљ." msgid "Mute" -msgstr "" +msgstr "Без звука" msgid "Fast" -msgstr "" +msgstr "Брзо" #. TRANSLATORS: "Very Low" is a quality setting here msgid "Very Low" -msgstr "" +msgstr "Врло низак" #. TRANSLATORS: "Low" is a quality setting here msgid "Low" -msgstr "" +msgstr "Низак" #. TRANSLATORS: "Medium" is a quality setting here msgid "Medium" -msgstr "" +msgstr "Средњи" #. TRANSLATORS: "High" is a quality setting here msgid "High" -msgstr "" +msgstr "Висок" #. TRANSLATORS: "Very High" is a quality setting here msgid "Very High" -msgstr "" +msgstr "Врло висок" #. TRANSLATORS: "Production" is a quality setting here msgid "Production" -msgstr "" +msgstr "Производни" -#, fuzzy msgid "Front Panel Microphone" -msgstr "Микрофон" +msgstr "Микрофон предње површи" msgid "Front Panel Line In" -msgstr "" +msgstr "Линијски улаз предње површи" msgid "Front Panel Headphones" -msgstr "" +msgstr "Слушалице предње површи" msgid "Front Panel Line Out" -msgstr "" +msgstr "Линијски излаз предње површи" msgid "Green Connector" -msgstr "" +msgstr "Зелени прикључак" msgid "Pink Connector" -msgstr "" +msgstr "Ружичаст прикључак" msgid "Blue Connector" -msgstr "" +msgstr "Плави прикључак" msgid "White Connector" -msgstr "" +msgstr "Бели прикључак" msgid "Black Connector" -msgstr "" +msgstr "Црни прикључак" msgid "Gray Connector" -msgstr "" +msgstr "Сиви прикључак" msgid "Orange Connector" -msgstr "" +msgstr "Наранџасти прикључак" msgid "Red Connector" -msgstr "" +msgstr "Црвени прикључак" msgid "Yellow Connector" -msgstr "" +msgstr "Жути прикључак" msgid "Green Front Panel Connector" -msgstr "" +msgstr "Зелени прикључак на предњој површи" msgid "Pink Front Panel Connector" -msgstr "" +msgstr "Ружичасти прикључак на предњој површи" msgid "Blue Front Panel Connector" -msgstr "" +msgstr "Плави прикључак на предњој површи" msgid "White Front Panel Connector" -msgstr "" +msgstr "Бели прикључак на предњој површи" msgid "Black Front Panel Connector" -msgstr "" +msgstr "Црни прикључак на предњој површи" msgid "Gray Front Panel Connector" -msgstr "" +msgstr "Сиви прикључак на предњој површи" msgid "Orange Front Panel Connector" -msgstr "" +msgstr "Наранџасти прикључак на предњој површи" msgid "Red Front Panel Connector" -msgstr "" +msgstr "Црвени прикључак на предњој површи" msgid "Yellow Front Panel Connector" -msgstr "" +msgstr "Жути прикључак на предњој површи" msgid "Spread Output" -msgstr "" +msgstr "Излаз ширења" msgid "Downmix" -msgstr "" +msgstr "Сабирни мешач" msgid "Virtual Mixer Input" -msgstr "" +msgstr "Улаз виртуелног мешача" msgid "Virtual Mixer Output" -msgstr "" +msgstr "Излаз виртуелног мешача" msgid "Virtual Mixer Channels" -msgstr "" +msgstr "Канали виртуелног мешача" #. TRANSLATORS: name + number of a volume mixer control #, c-format msgid "%s %d Function" -msgstr "" +msgstr "%s %d функција" #. TRANSLATORS: name of a volume mixer control #, c-format msgid "%s Function" -msgstr "" +msgstr "%s функција" msgid "" "Could not open audio device for playback. This version of the Open Sound " "System is not supported by this element." msgstr "" +"Не могу да покренем звучни уређај ради пуштања. Ово издање система отвореног " +"звука није подржано овим елементом." msgid "Playback is not supported by this audio device." -msgstr "" +msgstr "Пуштање није подржано од стране овог звучног уређаја." msgid "Audio playback error." -msgstr "" +msgstr "Грешка приликом пуштања звука." msgid "Recording is not supported by this audio device." -msgstr "" +msgstr "Снимање није подржано од стране овог звучног уређаја." msgid "Error recording from audio device." -msgstr "" +msgstr "Грешка приликом снимања са звучног уређаја." msgid "Gain" -msgstr "" +msgstr "Појачање" msgid "Headphone" -msgstr "" +msgstr "Слушалице" -#, fuzzy, c-format +#, c-format msgid "Error reading %d bytes from device '%s'." -msgstr "Не могу да примим бафере са уређаја „%s“." +msgstr "Грешка приликом читања %d бајтова са уређаја „%s“." #, c-format msgid "Got unexpected frame size of %u instead of %u." -msgstr "" +msgstr "Добих неочекивану величину кадра, %u уместо %u." #, c-format msgid "Error reading %d bytes on device '%s'." -msgstr "" +msgstr "Грешка приликом читања %d бајтова на уређају „%s“." #, c-format msgid "" "Error getting capabilities for device '%s': It isn't a v4l2 driver. Check if " "it is a v4l1 driver." msgstr "" +"Грешка приликом добављања могућности за уређај „%s“: Није в4л2 управљачки " +"програм. Проверите да ли је то в4л1 управљачки програм." #, c-format msgid "Failed to query attributes of input %d in device %s" -msgstr "" +msgstr "Нисам успео да пропитам својства уноса %d у уређају %s" -#, fuzzy, c-format +#, c-format msgid "Failed to get setting of tuner %d on device '%s'." -msgstr "Не могу да примим довољно бафера са уређаја „%s“." +msgstr "Нисам успео да добавим подешавања тјунера %d на уређају „%s“." #, c-format msgid "Failed to query norm on device '%s'." -msgstr "" +msgstr "Нисам успео да пропитам норму на уређају „%s“." #, c-format msgid "Failed getting controls attributes on device '%s'." -msgstr "" +msgstr "Нисам успео да добавим својства контрола на уређају „%s“." -#, fuzzy, c-format +#, c-format msgid "Cannot identify device '%s'." -msgstr "Не могу да пишем на видео уређај „%s“." +msgstr "Не могу да распознам уређај „%s“." #, c-format msgid "This isn't a device '%s'." -msgstr "" +msgstr "Ово није уређај „%s“." -#, fuzzy, c-format +#, c-format msgid "Could not open device '%s' for reading and writing." -msgstr "Не могу да отворим уређај „%s“ ради читања и уписа." +msgstr "Не могу да покренем уређај „%s“ ради читања и уписа." -#, fuzzy, c-format +#, c-format msgid "Device '%s' is not a capture device." -msgstr "Уређај „%s“ не представља уређај за снимање." +msgstr "Уређај „%s“ није уређај за снимање." -#, fuzzy, c-format +#, c-format msgid "Device '%s' is not a output device." -msgstr "Уређај „%s“ не представља уређај за снимање." +msgstr "Уређај „%s“ није излазни уређај." -#, fuzzy, c-format +#, c-format msgid "Failed to set norm for device '%s'." -msgstr "Не могу да примим бафере са уређаја „%s“." +msgstr "Нисам успео да подесим норму за уређај „%s“." -#, fuzzy, c-format +#, c-format msgid "Failed to get current tuner frequency for device '%s'." -msgstr "Не могу да примим довољно бафера са уређаја „%s“." +msgstr "Нисам успео да добавим текућу учестаност тјунера за уређај „%s“." #, c-format msgid "Failed to set current tuner frequency for device '%s' to %lu Hz." msgstr "" +"Нисам успео да подесим текућу учестаност тјунера за уређај „%s“ на %lu Hz." #, c-format msgid "Failed to get signal strength for device '%s'." -msgstr "" +msgstr "Нисам успео да добавим јачину сигнала за уређај „%s“." -#, fuzzy, c-format +#, c-format msgid "Failed to get value for control %d on device '%s'." -msgstr "Не могу да затворим управљачки уређај „%s“." +msgstr "Нисам успео да добавим вредност за контролу %d на уређају „%s“." -#, fuzzy, c-format +#, c-format msgid "Failed to set value %d for control %d on device '%s'." -msgstr "Не могу да затворим управљачки уређај „%s“." +msgstr "Нисам успео да подесим вредност %d за контролу %d на уређају „%s“." #, c-format msgid "Failed to get current input on device '%s'. May be it is a radio device" msgstr "" +"Нисам успео да добавим текући улаз на уређају „%s“. Можда је то радио уређај." -#, fuzzy, c-format +#, c-format msgid "Failed to set input %d on device %s." -msgstr "Не могу да отворим радио уређај '%s'" +msgstr "Нисам успео да подесим улаз %d на уређају %s." -#, fuzzy, c-format +#, c-format msgid "" "Failed to get current output on device '%s'. May be it is a radio device" -msgstr "Не могу да примим довољно бафера са уређаја „%s“." +msgstr "" +"Нисам успео да добавим текући излаз на уређају „%s“. Можда је то радио " +"уређај." -#, fuzzy, c-format +#, c-format msgid "Failed to set output %d on device %s." -msgstr "Не могу да отворим радио уређај '%s'" +msgstr "Нисам успео да подесим излаз %d на уређају %s." -#, fuzzy, c-format +#, c-format msgid "Could not enqueue buffers in device '%s'." -msgstr "Не могу да примим бафере са уређаја „%s“." +msgstr "Не могу да додам помоћне меморије у уређају „%s“." -#, fuzzy, c-format +#, c-format msgid "Failed trying to get video frames from device '%s'." -msgstr "Не могу да примим довољно бафера са уређаја „%s“." +msgstr "Нисам успео да добавим видео кадрове са уређаја „%s“." #, c-format msgid "Failed after %d tries. device %s. system error: %s" -msgstr "" +msgstr "Нисам успео након %d покушаја. уређај %s. системска грешка: %s" -#, fuzzy, c-format +#, c-format msgid "Could not get parameters on device '%s'" -msgstr "Не могу да примим бафере са уређаја „%s“." +msgstr "Не могу да добавим параметре на уређају „%s“" msgid "Video input device did not accept new frame rate setting." -msgstr "" +msgstr "Улазни видео уређај не прихвата нове поставке протока кадрова." -#, fuzzy, c-format +#, c-format msgid "Could not map buffers from device '%s'" -msgstr "Не могу да примим бафере са уређаја „%s“." +msgstr "Не могу да мапирам помоћне меморије са уређаја „%s“" #, c-format msgid "The driver of device '%s' does not support any known capture method." msgstr "" +"Управљачки програм уређаја „%s“ не подржава ниједан познати начин снимања." msgid "Changing resolution at runtime is not yet supported." -msgstr "" +msgstr "Промена резолуције приликом извршавања још увек није подржана." msgid "Cannot operate without a clock" -msgstr "" +msgstr "Не могу да радим без сата" #~ msgid "Could not open file \"%s\" for writing." #~ msgstr "Не могу да отворим датотеку „%s“ ради уписа." From 60e53346af0c23588630879eac3de14f1e1fe8fa Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 10 Dec 2011 12:32:32 +0000 Subject: [PATCH 04/65] tests: fix up rgvolume test for basetransform event caching Some tests assumed that tag events would always pushed through immediately, which isn't the case any longer, so push a newsegment event and an empty buffer first. --- tests/check/elements/rgvolume.c | 45 +++++++++++++++++++++++++++++++++ 1 file changed, 45 insertions(+) diff --git a/tests/check/elements/rgvolume.c b/tests/check/elements/rgvolume.c index 3add27a722..52262590f2 100644 --- a/tests/check/elements/rgvolume.c +++ b/tests/check/elements/rgvolume.c @@ -49,6 +49,8 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_STATIC_CAPS (RG_VOLUME_CAPS_TEMPLATE_STRING) ); +static GstBuffer *test_buffer_new (gfloat value); + /* gstcheck sets up a chain function that appends buffers to a global list. * This is our equivalent of that for event handling. */ static gboolean @@ -78,6 +80,35 @@ setup_rgvolume (void) return element; } +static void +send_newsegment_and_empty_buffer (void) +{ + GstBuffer *buf; + GstEvent *ev; + + fail_unless (g_list_length (events) == 0); + + ev = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0); + fail_unless (gst_pad_push_event (mysrcpad, ev), + "Pushing newsegment event failed"); + + buf = test_buffer_new (0.0); + GST_BUFFER_SIZE (buf) = 0; + GST_BUFFER_DURATION (buf) = 0; + GST_BUFFER_OFFSET_END (buf) = GST_BUFFER_OFFSET (buf); + fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK); + + fail_unless (g_list_length (events) == 1); + fail_unless (events->data == ev); + gst_mini_object_unref ((GstMiniObject *) events->data); + events = g_list_remove (events, ev); + + fail_unless (g_list_length (buffers) == 1); + fail_unless (buffers->data == buf); + gst_mini_object_unref ((GstMiniObject *) buffers->data); + buffers = g_list_remove (buffers, buf); +} + static void cleanup_rgvolume (GstElement * element) { @@ -288,6 +319,8 @@ GST_START_TEST (test_events) set_playing_state (element); + send_newsegment_and_empty_buffer (); + tag_list = gst_tag_list_new (); gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_GAIN, +4.95, GST_TAG_TRACK_PEAK, 0.59463, @@ -336,6 +369,8 @@ GST_START_TEST (test_simple) "pre-amp", -6.00, "fallback-gain", +1.23, NULL); set_playing_state (element); + send_newsegment_and_empty_buffer (); + tag_list = gst_tag_list_new (); gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_GAIN, -3.45, GST_TAG_TRACK_PEAK, 1.0, @@ -376,6 +411,8 @@ GST_START_TEST (test_fallback_gain) "pre-amp", -6.00, "fallback-gain", -3.00, NULL); set_playing_state (element); + send_newsegment_and_empty_buffer (); + tag_list = gst_tag_list_new (); gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_GAIN, +3.5, GST_TAG_TRACK_PEAK, 1.0, @@ -419,6 +456,8 @@ GST_START_TEST (test_fallback_track) "pre-amp", -6.00, "fallback-gain", +1.23, NULL); set_playing_state (element); + send_newsegment_and_empty_buffer (); + tag_list = gst_tag_list_new (); gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_GAIN, +2.11, GST_TAG_TRACK_PEAK, 1.0, NULL); @@ -444,6 +483,8 @@ GST_START_TEST (test_fallback_album) "pre-amp", -6.00, "fallback-gain", +1.23, NULL); set_playing_state (element); + send_newsegment_and_empty_buffer (); + tag_list = gst_tag_list_new (); gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE, GST_TAG_ALBUM_GAIN, +3.73, GST_TAG_ALBUM_PEAK, 1.0, NULL); @@ -466,6 +507,8 @@ GST_START_TEST (test_headroom) "pre-amp", +0.00, "fallback-gain", +1.23, NULL); set_playing_state (element); + send_newsegment_and_empty_buffer (); + tag_list = gst_tag_list_new (); gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_GAIN, +3.50, GST_TAG_TRACK_PEAK, 1.0, NULL); @@ -508,6 +551,8 @@ GST_START_TEST (test_reference_level) "headroom", +0.00, "pre-amp", +0.00, "fallback-gain", +1.23, NULL); set_playing_state (element); + send_newsegment_and_empty_buffer (); + tag_list = gst_tag_list_new (); gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE, GST_TAG_TRACK_GAIN, 0.00, GST_TAG_TRACK_PEAK, 0.2, From a65091802fc1458c372687195d779ddba682d376 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 10 Dec 2011 13:35:08 +0000 Subject: [PATCH 05/65] tests: dist test file used in matroskaparse unit test --- tests/files/Makefile.am | 1 + 1 file changed, 1 insertion(+) diff --git a/tests/files/Makefile.am b/tests/files/Makefile.am index a0a2829378..ae08e1a0fc 100644 --- a/tests/files/Makefile.am +++ b/tests/files/Makefile.am @@ -7,6 +7,7 @@ EXTRA_DIST = \ id3-577468-unsynced-tag.tag \ id3-588148-unsynced-v24.tag \ pcm16sine.flv \ + pinknoise-vorbis.mkv \ test-cert.pem \ test-key.pem From 5bb4dcd89c73296062d9de49d1493fb3b5439154 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 10 Dec 2011 14:48:57 +0000 Subject: [PATCH 06/65] soup: fix start/stop race in souphttpclientsink Fix crash or hang in generic/states unit test when doing stop() right after start(). Create main loop in the start function already and not just in the thread function, so that stop() always has a valid main loop to quit on. Also, calling g_main_loop_quit() before g_main_loop_run() won't work and result in the stop function waiting for the thread to join forever. Therefore, wait for the thread to be ready and get the main loop running in the start() function, to be sure stop() always works. --- ext/soup/gstsouphttpclientsink.c | 32 +++++++++++++++++++++++++++++++- 1 file changed, 31 insertions(+), 1 deletion(-) diff --git a/ext/soup/gstsouphttpclientsink.c b/ext/soup/gstsouphttpclientsink.c index 494dd8d699..13cc805763 100644 --- a/ext/soup/gstsouphttpclientsink.c +++ b/ext/soup/gstsouphttpclientsink.c @@ -450,6 +450,20 @@ gst_soup_http_client_sink_get_times (GstBaseSink * sink, GstBuffer * buffer, } +static gboolean +thread_ready_idle_cb (gpointer data) +{ + GstSoupHttpClientSink *souphttpsink = GST_SOUP_HTTP_CLIENT_SINK (data); + + GST_LOG_OBJECT (souphttpsink, "thread ready"); + + g_mutex_lock (souphttpsink->mutex); + g_cond_signal (souphttpsink->cond); + g_mutex_unlock (souphttpsink->mutex); + + return FALSE; /* only run once */ +} + static gpointer thread_func (gpointer ptr) { @@ -457,7 +471,6 @@ thread_func (gpointer ptr) GST_DEBUG ("thread start"); - souphttpsink->loop = g_main_loop_new (souphttpsink->context, TRUE); g_main_loop_run (souphttpsink->loop); GST_DEBUG ("thread quit"); @@ -473,13 +486,30 @@ gst_soup_http_client_sink_start (GstBaseSink * sink) if (souphttpsink->prop_session) { souphttpsink->session = souphttpsink->prop_session; } else { + GSource *source; GError *error = NULL; souphttpsink->context = g_main_context_new (); + /* set up idle source to signal when the main loop is running and + * it's safe for ::stop() to call g_main_loop_quit() */ + source = g_idle_source_new (); + g_source_set_callback (source, thread_ready_idle_cb, sink, NULL); + g_source_attach (source, souphttpsink->context); + g_source_unref (source); + + souphttpsink->loop = g_main_loop_new (souphttpsink->context, TRUE); + + g_mutex_lock (souphttpsink->mutex); + souphttpsink->thread = g_thread_create (thread_func, souphttpsink, TRUE, &error); + GST_LOG_OBJECT (souphttpsink, "waiting for main loop thread to start up"); + g_cond_wait (souphttpsink->cond, souphttpsink->mutex); + g_mutex_unlock (souphttpsink->mutex); + GST_LOG_OBJECT (souphttpsink, "main loop thread running"); + souphttpsink->session = soup_session_async_new_with_options (SOUP_SESSION_ASYNC_CONTEXT, souphttpsink->context, SOUP_SESSION_USER_AGENT, From 9c1095f474a9b17ead0d092fbf760b22b391bd5a Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 11 Dec 2011 17:55:14 +0000 Subject: [PATCH 07/65] pulseaudiosink: don't leak pad template --- ext/pulse/pulseaudiosink.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/ext/pulse/pulseaudiosink.c b/ext/pulse/pulseaudiosink.c index 389ce46743..15f3bf1eeb 100644 --- a/ext/pulse/pulseaudiosink.c +++ b/ext/pulse/pulseaudiosink.c @@ -140,8 +140,7 @@ gst_pulse_audio_sink_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - gst_element_class_add_static_pad_template (element_class, - &sink_template); + gst_element_class_add_static_pad_template (element_class, &sink_template); gst_element_class_set_details_simple (element_class, "Bin wrapping pulsesink", "Sink/Audio/Bin", @@ -356,6 +355,7 @@ static void gst_pulse_audio_sink_init (GstPulseAudioSink * pbin, GstPulseAudioSinkClass * klass) { + GstPadTemplate *template; GstPad *pad = NULL; GParamSpec **specs; GString *prop; @@ -375,8 +375,9 @@ gst_pulse_audio_sink_init (GstPulseAudioSink * pbin, } pad = gst_element_get_static_pad (GST_ELEMENT (pbin->psink), "sink"); - pbin->sinkpad = gst_ghost_pad_new_from_template ("sink", pad, - gst_static_pad_template_get (&sink_template)); + template = gst_static_pad_template_get (&sink_template); + pbin->sinkpad = gst_ghost_pad_new_from_template ("sink", pad, template); + gst_object_unref (template); pbin->sinkpad_old_eventfunc = GST_PAD_EVENTFUNC (pbin->sinkpad); gst_pad_set_event_function (pbin->sinkpad, From 8aebe194aa6d9bb01aeb53538c4ada54d3c19f1d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 11 Dec 2011 18:40:31 +0000 Subject: [PATCH 08/65] gdkpixbufsink: fix inverted pixel-aspect-ratio Spotted by Mike Morrison. https://bugzilla.gnome.org/show_bug.cgi?id=665882 --- ext/gdk_pixbuf/gstgdkpixbufsink.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/ext/gdk_pixbuf/gstgdkpixbufsink.c b/ext/gdk_pixbuf/gstgdkpixbufsink.c index 8a9677a2ce..2c2714ef19 100644 --- a/ext/gdk_pixbuf/gstgdkpixbufsink.c +++ b/ext/gdk_pixbuf/gstgdkpixbufsink.c @@ -271,7 +271,7 @@ gst_gdk_pixbuf_sink_set_caps (GstBaseSink * basesink, GstCaps * caps) GST_INFO_OBJECT (sink, "format : %d", fmt); GST_INFO_OBJECT (sink, "width x height : %d x %d", w, h); - GST_INFO_OBJECT (sink, "pixel-aspect-ratio : %d/%d", par_d, par_n); + GST_INFO_OBJECT (sink, "pixel-aspect-ratio : %d/%d", par_n, par_d); return TRUE; } @@ -344,7 +344,7 @@ gst_gdk_pixbuf_sink_handle_buffer (GstBaseSink * basesink, GstBuffer * buf, * The structure will take its own ref to the pixbuf. */ s = gst_structure_new (msg_name, "pixbuf", GDK_TYPE_PIXBUF, pixbuf, - "pixel-aspect-ratio", GST_TYPE_FRACTION, sink->par_d, sink->par_n, + "pixel-aspect-ratio", GST_TYPE_FRACTION, sink->par_n, sink->par_d, NULL); msg = gst_message_new_element (GST_OBJECT_CAST (sink), s); From e94186241b04b9208f41dd804662781c06e005f5 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Mon, 12 Dec 2011 10:24:45 +0100 Subject: [PATCH 09/65] configure: Require GLib >= 2.24 All other modules require this already and nobody is testing with older versions anyway. --- configure.ac | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/configure.ac b/configure.ac index f8a3f93fec..70fc566770 100644 --- a/configure.ac +++ b/configure.ac @@ -207,7 +207,7 @@ AC_CHECK_TYPE([struct ip_mreqn], [ dnl *** checks for dependency libraries *** dnl GLib is required -AG_GST_GLIB_CHECK([2.20]) +AG_GST_GLIB_CHECK([2.24]) PKG_CHECK_MODULES(GIO, [ gio-2.0 >= 2.20 ], , AC_MSG_ERROR([gio is required])) dnl Orc From 66f6e1288845ab21787b1fcd4605a90a5b552bb8 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 12 Dec 2011 02:30:45 +0000 Subject: [PATCH 10/65] Work around deprecated thread API in glib master Add private replacements for deprecated functions such as g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly to avoid the deprecation warnings. We'll change these over to the new API once we depend on glib >= 2.32. --- Makefile.am | 1 + ext/jack/gstjackaudioclient.c | 2 + ext/pulse/pulseaudiosink.c | 1 + ext/pulse/pulsesink.c | 2 + ext/soup/gstsouphttpclientsink.c | 2 + gst-libs/gst/glib-compat-private.h | 120 ++++++++++++++++++++++++++++ gst/audiofx/audiochebband.c | 2 + gst/audiofx/audiocheblimit.c | 2 + gst/audiofx/audiofirfilter.c | 2 + gst/audiofx/audioiirfilter.c | 2 + gst/audiofx/audiowsincband.c | 2 + gst/audiofx/audiowsinclimit.c | 2 + gst/equalizer/gstiirequalizer.c | 2 + gst/imagefreeze/gstimagefreeze.c | 2 + gst/rtpmanager/gstrtpbin.c | 2 + gst/rtpmanager/gstrtpjitterbuffer.c | 2 + gst/rtpmanager/gstrtpsession.c | 2 + gst/rtpmanager/rtpsession.c | 2 + gst/shapewipe/gstshapewipe.c | 1 + gst/udp/gstmultiudpsink.c | 5 +- gst/videobox/gstvideobox.c | 2 + gst/videocrop/gstaspectratiocrop.c | 5 +- gst/videomixer/videomixer.c | 2 + gst/videomixer/videomixer2.c | 2 + sys/oss4/oss4-mixer.c | 1 + sys/v4l2/gstv4l2bufferpool.c | 1 + sys/v4l2/gstv4l2xoverlay.c | 1 + sys/ximage/gstximagesrc.c | 2 + 28 files changed, 170 insertions(+), 4 deletions(-) create mode 100644 gst-libs/gst/glib-compat-private.h diff --git a/Makefile.am b/Makefile.am index fb78f5d65d..862126dd13 100644 --- a/Makefile.am +++ b/Makefile.am @@ -38,6 +38,7 @@ DISTCLEANFILES = _stdint.h noinst_HEADERS = \ gst-libs/gst/gettext.h \ gst-libs/gst/gst-i18n-plugin.h + gst-libs/gst/glib-compat-private.h ACLOCAL_AMFLAGS = -I m4 -I common/m4 diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c index 1789edb60d..2bb355529b 100644 --- a/ext/jack/gstjackaudioclient.c +++ b/ext/jack/gstjackaudioclient.c @@ -23,6 +23,8 @@ #include "gstjackaudioclient.h" +#include + GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug); #define GST_CAT_DEFAULT gst_jack_audio_client_debug diff --git a/ext/pulse/pulseaudiosink.c b/ext/pulse/pulseaudiosink.c index 15f3bf1eeb..cd9f7d914b 100644 --- a/ext/pulse/pulseaudiosink.c +++ b/ext/pulse/pulseaudiosink.c @@ -53,6 +53,7 @@ #include #include +#include #include #include "pulsesink.h" diff --git a/ext/pulse/pulsesink.c b/ext/pulse/pulsesink.c index cfab338b5d..3c6177089d 100644 --- a/ext/pulse/pulsesink.c +++ b/ext/pulse/pulsesink.c @@ -58,6 +58,8 @@ #include /* only used for GST_PLUGINS_BASE_VERSION_* */ +#include + #include "pulsesink.h" #include "pulseutil.h" diff --git a/ext/soup/gstsouphttpclientsink.c b/ext/soup/gstsouphttpclientsink.c index 13cc805763..5f3d47e1e3 100644 --- a/ext/soup/gstsouphttpclientsink.c +++ b/ext/soup/gstsouphttpclientsink.c @@ -42,6 +42,8 @@ #include #include "gstsouphttpclientsink.h" +#include + GST_DEBUG_CATEGORY_STATIC (souphttpclientsink_dbg); #define GST_CAT_DEFAULT souphttpclientsink_dbg diff --git a/gst-libs/gst/glib-compat-private.h b/gst-libs/gst/glib-compat-private.h new file mode 100644 index 0000000000..d143d3257d --- /dev/null +++ b/gst-libs/gst/glib-compat-private.h @@ -0,0 +1,120 @@ +/* + * glib-compat.c + * Functions copied from glib 2.10 + * + * Copyright 2005 David Schleef + */ + +#ifndef __GLIB_COMPAT_PRIVATE_H__ +#define __GLIB_COMPAT_PRIVATE_H__ + +#include + +G_BEGIN_DECLS + +#if !GLIB_CHECK_VERSION(2,25,0) + +#if defined (_MSC_VER) && !defined(_WIN64) +typedef struct _stat32 GStatBuf; +#else +typedef struct stat GStatBuf; +#endif + +#endif + +#if GLIB_CHECK_VERSION(2,26,0) +#define GLIB_HAS_GDATETIME +#endif + +/* See bug #651514 */ +#if GLIB_CHECK_VERSION(2,29,5) +#define G_ATOMIC_POINTER_COMPARE_AND_EXCHANGE(a,b,c) \ + g_atomic_pointer_compare_and_exchange ((a),(b),(c)) +#define G_ATOMIC_INT_COMPARE_AND_EXCHANGE(a,b,c) \ + g_atomic_int_compare_and_exchange ((a),(b),(c)) +#else +#define G_ATOMIC_POINTER_COMPARE_AND_EXCHANGE(a,b,c) \ + g_atomic_pointer_compare_and_exchange ((volatile gpointer *)(a),(b),(c)) +#define G_ATOMIC_INT_COMPARE_AND_EXCHANGE(a,b,c) \ + g_atomic_int_compare_and_exchange ((volatile int *)(a),(b),(c)) +#endif + +/* See bug #651514 */ +#if GLIB_CHECK_VERSION(2,29,5) +#define G_ATOMIC_INT_ADD(a,b) g_atomic_int_add ((a),(b)) +#else +#define G_ATOMIC_INT_ADD(a,b) g_atomic_int_exchange_and_add ((a),(b)) +#endif + +/* copies */ + +#if GLIB_CHECK_VERSION (2, 31, 0) +#define g_mutex_new gst_g_mutex_new +static inline GMutex * +gst_g_mutex_new (void) +{ + GMutex *mutex = g_slice_new (GMutex); + g_mutex_init (mutex); + return mutex; +} +#define g_mutex_free gst_g_mutex_free +static inline void +gst_g_mutex_free (GMutex *mutex) +{ + g_mutex_clear (mutex); + g_slice_free (GMutex, mutex); +} +#define g_static_rec_mutex_init gst_g_static_rec_mutex_init +static inline void +gst_g_static_rec_mutex_init (GStaticRecMutex *mutex) +{ + static const GStaticRecMutex init_mutex = G_STATIC_REC_MUTEX_INIT; + + *mutex = init_mutex; +} +#define g_cond_new gst_g_cond_new +static inline GCond * +gst_g_cond_new (void) +{ + GCond *cond = g_slice_new (GCond); + g_cond_init (cond); + return cond; +} +#define g_cond_free gst_g_cond_free +static inline void +gst_g_cond_free (GCond *cond) +{ + g_cond_clear (cond); + g_slice_free (GCond, cond); +} +#define g_cond_timed_wait gst_g_cond_timed_wait +static inline gboolean +gst_g_cond_timed_wait (GCond *cond, GMutex *mutex, GTimeVal *abs_time) +{ + gint64 end_time; + + if (abs_time == NULL) { + g_cond_wait (cond, mutex); + return TRUE; + } + + end_time = abs_time->tv_sec; + end_time *= 1000000; + end_time += abs_time->tv_usec; + + /* would be nice if we had clock_rtoffset, but that didn't seem to + * make it into the kernel yet... + */ + /* if CLOCK_MONOTONIC is not defined then g_get_montonic_time() and + * g_get_real_time() are returning the same clock and we'd add ~0 + */ + end_time += g_get_monotonic_time () - g_get_real_time (); + return g_cond_wait_until (cond, mutex, end_time); +} +#endif /* GLIB_CHECK_VERSION (2, 31, 0) */ + +/* adaptations */ + +G_END_DECLS + +#endif diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c index 89a950c23e..08c1acf34c 100644 --- a/gst/audiofx/audiochebband.c +++ b/gst/audiofx/audiochebband.c @@ -82,6 +82,8 @@ #include "audiochebband.h" +#include "gst/glib-compat-private.h" + #define GST_CAT_DEFAULT gst_audio_cheb_band_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c index 8d51fbbf66..b7bb90bbc2 100644 --- a/gst/audiofx/audiocheblimit.c +++ b/gst/audiofx/audiocheblimit.c @@ -78,6 +78,8 @@ #include "audiocheblimit.h" +#include "gst/glib-compat-private.h" + #define GST_CAT_DEFAULT gst_audio_cheb_limit_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); diff --git a/gst/audiofx/audiofirfilter.c b/gst/audiofx/audiofirfilter.c index b6102ae78f..bd5a51be4f 100644 --- a/gst/audiofx/audiofirfilter.c +++ b/gst/audiofx/audiofirfilter.c @@ -57,6 +57,8 @@ #include "audiofirfilter.h" +#include "gst/glib-compat-private.h" + #define GST_CAT_DEFAULT gst_audio_fir_filter_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); diff --git a/gst/audiofx/audioiirfilter.c b/gst/audiofx/audioiirfilter.c index bac16b2952..ff5f9caec4 100644 --- a/gst/audiofx/audioiirfilter.c +++ b/gst/audiofx/audioiirfilter.c @@ -53,6 +53,8 @@ #include "audioiirfilter.h" +#include "gst/glib-compat-private.h" + #define GST_CAT_DEFAULT gst_audio_iir_filter_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); diff --git a/gst/audiofx/audiowsincband.c b/gst/audiofx/audiowsincband.c index 07356e2403..3f94988466 100644 --- a/gst/audiofx/audiowsincband.c +++ b/gst/audiofx/audiowsincband.c @@ -64,6 +64,8 @@ #include "audiowsincband.h" +#include "gst/glib-compat-private.h" + #define GST_CAT_DEFAULT gst_gst_audio_wsincband_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); diff --git a/gst/audiofx/audiowsinclimit.c b/gst/audiofx/audiowsinclimit.c index aa0cf4b0f4..ad605697d6 100644 --- a/gst/audiofx/audiowsinclimit.c +++ b/gst/audiofx/audiowsinclimit.c @@ -64,6 +64,8 @@ #include "audiowsinclimit.h" +#include "gst/glib-compat-private.h" + #define GST_CAT_DEFAULT gst_audio_wsinclimit_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); diff --git a/gst/equalizer/gstiirequalizer.c b/gst/equalizer/gstiirequalizer.c index c640124de5..d66db1a4d0 100644 --- a/gst/equalizer/gstiirequalizer.c +++ b/gst/equalizer/gstiirequalizer.c @@ -31,6 +31,8 @@ #include "gstiirequalizer3bands.h" #include "gstiirequalizer10bands.h" +#include "gst/glib-compat-private.h" + GST_DEBUG_CATEGORY (equalizer_debug); #define GST_CAT_DEFAULT equalizer_debug diff --git a/gst/imagefreeze/gstimagefreeze.c b/gst/imagefreeze/gstimagefreeze.c index 77785caaef..312f4aa155 100644 --- a/gst/imagefreeze/gstimagefreeze.c +++ b/gst/imagefreeze/gstimagefreeze.c @@ -41,6 +41,8 @@ #include "config.h" #endif +#include + #include "gstimagefreeze.h" static void gst_image_freeze_finalize (GObject * object); diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index b322f8ecc4..ef00a8519f 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -129,6 +129,8 @@ #include "gstrtpsession.h" #include "gstrtpjitterbuffer.h" +#include + GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug); #define GST_CAT_DEFAULT gst_rtp_bin_debug diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index c69d47772e..4c72fb6265 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -67,6 +67,8 @@ #include "rtpjitterbuffer.h" #include "rtpstats.h" +#include + GST_DEBUG_CATEGORY (rtpjitterbuffer_debug); #define GST_CAT_DEFAULT (rtpjitterbuffer_debug) diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 68284383c3..31f9a73771 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -114,6 +114,8 @@ #include +#include + #include "gstrtpbin-marshal.h" #include "gstrtpsession.h" #include "rtpsession.h" diff --git a/gst/rtpmanager/rtpsession.c b/gst/rtpmanager/rtpsession.c index f499106f26..eed96ac5b7 100644 --- a/gst/rtpmanager/rtpsession.c +++ b/gst/rtpmanager/rtpsession.c @@ -23,6 +23,8 @@ #include #include +#include + #include "gstrtpbin-marshal.h" #include "rtpsession.h" diff --git a/gst/shapewipe/gstshapewipe.c b/gst/shapewipe/gstshapewipe.c index 3397dda8f6..b8c91b527c 100644 --- a/gst/shapewipe/gstshapewipe.c +++ b/gst/shapewipe/gstshapewipe.c @@ -46,6 +46,7 @@ #include #include +#include #include "gstshapewipe.h" diff --git a/gst/udp/gstmultiudpsink.c b/gst/udp/gstmultiudpsink.c index a2d74a71f0..7a35aaecad 100644 --- a/gst/udp/gstmultiudpsink.c +++ b/gst/udp/gstmultiudpsink.c @@ -41,6 +41,8 @@ #include #include +#include "gst/glib-compat-private.h" + GST_DEBUG_CATEGORY_STATIC (multiudpsink_debug); #define GST_CAT_DEFAULT (multiudpsink_debug) @@ -172,8 +174,7 @@ gst_multiudpsink_base_init (gpointer g_class) { GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - gst_element_class_add_static_pad_template (element_class, - &sink_template); + gst_element_class_add_static_pad_template (element_class, &sink_template); gst_element_class_set_details_simple (element_class, "UDP packet sender", "Sink/Network", diff --git a/gst/videobox/gstvideobox.c b/gst/videobox/gstvideobox.c index 691c5f1238..9d311840b2 100644 --- a/gst/videobox/gstvideobox.c +++ b/gst/videobox/gstvideobox.c @@ -66,6 +66,8 @@ #include +#include "gst/glib-compat-private.h" + GST_DEBUG_CATEGORY_STATIC (videobox_debug); #define GST_CAT_DEFAULT videobox_debug diff --git a/gst/videocrop/gstaspectratiocrop.c b/gst/videocrop/gstaspectratiocrop.c index b37217aed4..150df37f49 100644 --- a/gst/videocrop/gstaspectratiocrop.c +++ b/gst/videocrop/gstaspectratiocrop.c @@ -43,6 +43,8 @@ #include "gstaspectratiocrop.h" +#include "gst/glib-compat-private.h" + GST_DEBUG_CATEGORY_STATIC (aspect_ratio_crop_debug); #define GST_CAT_DEFAULT aspect_ratio_crop_debug @@ -169,8 +171,7 @@ gst_aspect_ratio_crop_base_init (gpointer g_class) "Crops video into a user-defined aspect-ratio", "Thijs Vermeir "); - gst_element_class_add_static_pad_template (element_class, - &sink_template); + gst_element_class_add_static_pad_template (element_class, &sink_template); gst_element_class_add_static_pad_template (element_class, &src_template); } diff --git a/gst/videomixer/videomixer.c b/gst/videomixer/videomixer.c index 2c78c618b9..5a49b3912b 100644 --- a/gst/videomixer/videomixer.c +++ b/gst/videomixer/videomixer.c @@ -89,6 +89,8 @@ #include "videomixer.h" #include "videomixer2.h" +#include "gst/glib-compat-private.h" + #ifdef DISABLE_ORC #define orc_memset memset #else diff --git a/gst/videomixer/videomixer2.c b/gst/videomixer/videomixer2.c index fde327a717..b876c22fd1 100644 --- a/gst/videomixer/videomixer2.c +++ b/gst/videomixer/videomixer2.c @@ -93,6 +93,8 @@ #include +#include "gst/glib-compat-private.h" + #ifdef DISABLE_ORC #define orc_memset memset #else diff --git a/sys/oss4/oss4-mixer.c b/sys/oss4/oss4-mixer.c index fa81eccca7..87f7ca61e5 100644 --- a/sys/oss4/oss4-mixer.c +++ b/sys/oss4/oss4-mixer.c @@ -54,6 +54,7 @@ #include #include +#include "gst/glib-compat-private.h" #include diff --git a/sys/v4l2/gstv4l2bufferpool.c b/sys/v4l2/gstv4l2bufferpool.c index e976ec5079..b81c6a4118 100644 --- a/sys/v4l2/gstv4l2bufferpool.c +++ b/sys/v4l2/gstv4l2bufferpool.c @@ -39,6 +39,7 @@ #endif #include "v4l2_calls.h" #include "gst/gst-i18n-plugin.h" +#include /* videodev2.h is not versioned and we can't easily check for the presence * of enum values at compile time, but the V4L2_CAP_VIDEO_OUTPUT_OVERLAY define diff --git a/sys/v4l2/gstv4l2xoverlay.c b/sys/v4l2/gstv4l2xoverlay.c index 80f114d202..60d0919067 100644 --- a/sys/v4l2/gstv4l2xoverlay.c +++ b/sys/v4l2/gstv4l2xoverlay.c @@ -40,6 +40,7 @@ #include "v4l2_calls.h" #include "gst/gst-i18n-plugin.h" +#include struct _GstV4l2Xv { diff --git a/sys/ximage/gstximagesrc.c b/sys/ximage/gstximagesrc.c index b0d99ad6ab..36c0b44e78 100644 --- a/sys/ximage/gstximagesrc.c +++ b/sys/ximage/gstximagesrc.c @@ -49,6 +49,8 @@ #include #include +#include "gst/glib-compat-private.h" + GST_DEBUG_CATEGORY_STATIC (gst_debug_ximage_src); #define GST_CAT_DEFAULT gst_debug_ximage_src From f60eb9ae2c2ebb402fafbfa314970890e9f3342b Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 12 Dec 2011 02:31:36 +0000 Subject: [PATCH 11/65] alpha: use new glib API for static mutex if available --- gst/alpha/gstalpha.c | 22 ++++++++++++++++++++++ gst/alpha/gstalpha.h | 4 ++++ 2 files changed, 26 insertions(+) diff --git a/gst/alpha/gstalpha.c b/gst/alpha/gstalpha.c index 65aae69b29..f4d0cce482 100644 --- a/gst/alpha/gstalpha.c +++ b/gst/alpha/gstalpha.c @@ -161,6 +161,8 @@ static GstStaticCaps gst_alpha_alpha_caps = ";" GST_VIDEO_CAPS_ARGB ";" GST_VIDEO_CAPS_BGRA ";" GST_VIDEO_CAPS_ABGR ";" GST_VIDEO_CAPS_RGBA); +/* FIXME: why do we need our own lock for this? */ +#if !GLIB_CHECK_VERSION (2, 31, 0) #define GST_ALPHA_LOCK(alpha) G_STMT_START { \ GST_LOG_OBJECT (alpha, "Locking alpha from thread %p", g_thread_self ()); \ g_static_mutex_lock (&alpha->lock); \ @@ -171,6 +173,18 @@ static GstStaticCaps gst_alpha_alpha_caps = GST_LOG_OBJECT (alpha, "Unlocking alpha from thread %p", g_thread_self ()); \ g_static_mutex_unlock (&alpha->lock); \ } G_STMT_END +#else +#define GST_ALPHA_LOCK(alpha) G_STMT_START { \ + GST_LOG_OBJECT (alpha, "Locking alpha from thread %p", g_thread_self ()); \ + g_mutex_lock (&alpha->lock); \ + GST_LOG_OBJECT (alpha, "Locked alpha from thread %p", g_thread_self ()); \ +} G_STMT_END + +#define GST_ALPHA_UNLOCK(alpha) G_STMT_START { \ + GST_LOG_OBJECT (alpha, "Unlocking alpha from thread %p", g_thread_self ()); \ + g_mutex_unlock (&alpha->lock); \ +} G_STMT_END +#endif static gboolean gst_alpha_start (GstBaseTransform * trans); static gboolean gst_alpha_get_unit_size (GstBaseTransform * btrans, @@ -312,7 +326,11 @@ gst_alpha_init (GstAlpha * alpha, GstAlphaClass * klass) alpha->black_sensitivity = DEFAULT_BLACK_SENSITIVITY; alpha->white_sensitivity = DEFAULT_WHITE_SENSITIVITY; +#if !GLIB_CHECK_VERSION (2, 31, 0) g_static_mutex_init (&alpha->lock); +#else + g_mutex_init (&alpha->lock); +#endif } static void @@ -320,7 +338,11 @@ gst_alpha_finalize (GObject * object) { GstAlpha *alpha = GST_ALPHA (object); +#if !GLIB_CHECK_VERSION (2, 31, 0) g_static_mutex_free (&alpha->lock); +#else + g_mutex_clear (&alpha->lock); +#endif G_OBJECT_CLASS (parent_class)->finalize (object); } diff --git a/gst/alpha/gstalpha.h b/gst/alpha/gstalpha.h index 44025572cd..5de966e09d 100644 --- a/gst/alpha/gstalpha.h +++ b/gst/alpha/gstalpha.h @@ -70,7 +70,11 @@ struct _GstAlpha /* */ /* caps */ +#if !GLIB_CHECK_VERSION (2, 31, 0) GStaticMutex lock; +#else + GMutex lock; +#endif GstVideoFormat in_format, out_format; gint width, height; From 330d9842884e701822fd2644f179c0e4113074f5 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 12 Dec 2011 02:38:37 +0000 Subject: [PATCH 12/65] Use g_thread_try_new() instead of g_thread_crate() with newer glib versions --- ext/soup/gstsouphttpclientsink.c | 6 ++++++ gst/rtpmanager/gstrtpsession.c | 5 +++++ sys/oss4/oss4-mixer.c | 5 +++++ tests/icles/v4l2src-test.c | 8 +++++++- 4 files changed, 23 insertions(+), 1 deletion(-) diff --git a/ext/soup/gstsouphttpclientsink.c b/ext/soup/gstsouphttpclientsink.c index 5f3d47e1e3..dc9e9415c3 100644 --- a/ext/soup/gstsouphttpclientsink.c +++ b/ext/soup/gstsouphttpclientsink.c @@ -504,8 +504,14 @@ gst_soup_http_client_sink_start (GstBaseSink * sink) g_mutex_lock (souphttpsink->mutex); + /* FIXME: error handling */ +#if !GLIB_CHECK_VERSION (2, 31, 0) souphttpsink->thread = g_thread_create (thread_func, souphttpsink, TRUE, &error); +#else + souphttpsink->thread = g_thread_try_new ("souphttpclientsink-thread", + thread_func, souphttpsink, &error); +#endif GST_LOG_OBJECT (souphttpsink, "waiting for main loop thread to start up"); g_cond_wait (souphttpsink->cond, souphttpsink->mutex); diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 31f9a73771..cf263e3cc0 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -910,8 +910,13 @@ start_rtcp_thread (GstRtpSession * rtpsession) g_thread_join (rtpsession->priv->thread); /* only create a new thread if the old one was stopped. Otherwise we can * just reuse the currently running one. */ +#if !GLIB_CHECK_VERSION (2, 31, 0) rtpsession->priv->thread = g_thread_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &error); +#else + rtpsession->priv->thread = g_thread_try_new ("rtpsession-rtcp-thread", + (GThreadFunc) rtcp_thread, rtpsession, &error); +#endif rtpsession->priv->thread_stopped = FALSE; } GST_RTP_SESSION_UNLOCK (rtpsession); diff --git a/sys/oss4/oss4-mixer.c b/sys/oss4/oss4-mixer.c index 87f7ca61e5..03e2d9d729 100644 --- a/sys/oss4/oss4-mixer.c +++ b/sys/oss4/oss4-mixer.c @@ -542,8 +542,13 @@ gst_oss4_mixer_start_watch_task (GstOss4Mixer * mixer) mixer->watch_cond = g_cond_new (); mixer->watch_shutdown = FALSE; +#if !GLIB_CHECK_VERSION (2, 31, 0) mixer->watch_thread = g_thread_create (gst_oss4_mixer_watch_thread, gst_object_ref (mixer), TRUE, &err); +#else + mixer->watch_thread = g_thread_try_new ("oss4-mixer-thread", + gst_oss4_mixer_watch_thread, gst_object_ref (mixer), &err); +#endif if (mixer->watch_thread == NULL) { GST_ERROR_OBJECT (mixer, "Could not create watch thread: %s", err->message); diff --git a/tests/icles/v4l2src-test.c b/tests/icles/v4l2src-test.c index 4fcf48b726..dbcaef014d 100644 --- a/tests/icles/v4l2src-test.c +++ b/tests/icles/v4l2src-test.c @@ -491,7 +491,13 @@ main (int argc, char *argv[]) gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING); loop = g_main_loop_new (NULL, FALSE); - if (!(input_thread = g_thread_create (read_user, source, TRUE, NULL))) { +#if !GLIB_CHECK_VERSION (2, 31, 0) + input_thread = g_thread_create (read_user, source, TRUE, NULL); +#else + input_thread = g_thread_try_new ("v4l2src-test", read_user, source, NULL); +#endif + + if (input_thread == NULL) { fprintf (stderr, "error: g_thread_create return NULL"); return -1; } From 72fc3d1bd92d62aff11e9ad5a6b7e89139a0f4c9 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 12 Dec 2011 02:41:37 +0000 Subject: [PATCH 13/65] tests: g_thread_init() is deprecated in glib master It's not needed any longer. --- tests/check/elements/souphttpsrc.c | 3 +++ tests/icles/equalizer-test.c | 2 ++ tests/icles/gdkpixbufsink-test.c | 2 ++ tests/icles/test-oss4.c | 2 ++ tests/icles/videocrop-test.c | 2 ++ 5 files changed, 11 insertions(+) diff --git a/tests/check/elements/souphttpsrc.c b/tests/check/elements/souphttpsrc.c index ba1574594c..7a1b677714 100644 --- a/tests/check/elements/souphttpsrc.c +++ b/tests/check/elements/souphttpsrc.c @@ -442,8 +442,11 @@ souphttpsrc_suite (void) TCase *tc_chain, *tc_internet; g_type_init (); + +#if !GLIB_CHECK_VERSION (2, 31, 0) if (!g_thread_supported ()) g_thread_init (NULL); +#endif s = suite_create ("souphttpsrc"); tc_chain = tcase_create ("general"); diff --git a/tests/icles/equalizer-test.c b/tests/icles/equalizer-test.c index e8126fad3a..fc6d5273ea 100644 --- a/tests/icles/equalizer-test.c +++ b/tests/icles/equalizer-test.c @@ -179,8 +179,10 @@ main (int argc, char **argv) GstPad *eq_sinkpad; gchar *uri; +#if !GLIB_CHECK_VERSION (2, 31, 0) if (!g_thread_supported ()) g_thread_init (NULL); +#endif /* command line option parsing */ ctx = g_option_context_new ("FILENAME"); diff --git a/tests/icles/gdkpixbufsink-test.c b/tests/icles/gdkpixbufsink-test.c index b09fe09a2d..425e47053c 100644 --- a/tests/icles/gdkpixbufsink-test.c +++ b/tests/icles/gdkpixbufsink-test.c @@ -329,8 +329,10 @@ main (int argc, char **argv) GOptionContext *ctx; GError *opt_err = NULL; +#if !GLIB_CHECK_VERSION (2, 31, 0) if (!g_thread_supported ()) g_thread_init (NULL); +#endif gtk_init (&argc, &argv); diff --git a/tests/icles/test-oss4.c b/tests/icles/test-oss4.c index f15875d7bc..233e891fd9 100644 --- a/tests/icles/test-oss4.c +++ b/tests/icles/test-oss4.c @@ -233,8 +233,10 @@ main (int argc, char **argv) GOptionContext *ctx; GError *err = NULL; +#if !GLIB_CHECK_VERSION (2, 31, 0) if (!g_thread_supported ()) g_thread_init (NULL); +#endif ctx = g_option_context_new (""); g_option_context_add_main_entries (ctx, options, NULL); diff --git a/tests/icles/videocrop-test.c b/tests/icles/videocrop-test.c index a4874e160a..cc8f8d04cc 100644 --- a/tests/icles/videocrop-test.c +++ b/tests/icles/videocrop-test.c @@ -193,8 +193,10 @@ main (int argc, char **argv) GstCaps *filter_caps = NULL; GList *caps_list, *l; +#if !GLIB_CHECK_VERSION (2, 31, 0) if (!g_thread_supported ()) g_thread_init (NULL); +#endif /* command line option parsing */ ctx = g_option_context_new (""); From b8b8454bcb2aba9a1e5477763b7841705df55052 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 12 Dec 2011 02:52:13 +0000 Subject: [PATCH 14/65] Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly GStaticRecMutex is part of our API/ABI, not much we can do here in 0.10 for most of these. --- ext/dv/gstdvdemux.c | 5 +++++ ext/flac/gstflacdec.c | 5 +++++ ext/wavpack/gstwavpackparse.c | 5 +++++ gst/avi/gstavidemux.c | 4 ++++ gst/flv/gstflvdemux.c | 4 ++++ gst/imagefreeze/gstimagefreeze.c | 4 ++++ gst/isomp4/gstqtmoovrecover.c | 4 ++++ gst/isomp4/qtdemux.c | 4 ++++ gst/matroska/matroska-demux.c | 4 ++++ gst/rtpmanager/gstrtpssrcdemux.c | 4 ++++ gst/rtsp/gstrtspsrc.c | 4 ++++ gst/videomixer/videomixer2.c | 4 ++++ gst/wavparse/gstwavparse.c | 5 +++++ 13 files changed, 56 insertions(+) diff --git a/ext/dv/gstdvdemux.c b/ext/dv/gstdvdemux.c index 7563203e9b..467ebe5870 100644 --- a/ext/dv/gstdvdemux.c +++ b/ext/dv/gstdvdemux.c @@ -21,6 +21,11 @@ #ifdef HAVE_CONFIG_H #include "config.h" #endif + +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/ext/flac/gstflacdec.c b/ext/flac/gstflacdec.c index f6c0d813c9..65aa927f87 100644 --- a/ext/flac/gstflacdec.c +++ b/ext/flac/gstflacdec.c @@ -44,6 +44,11 @@ #ifdef HAVE_CONFIG_H #include "config.h" #endif + +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include "gstflacdec.h" diff --git a/ext/wavpack/gstwavpackparse.c b/ext/wavpack/gstwavpackparse.c index 8734e625db..e1ff785b91 100644 --- a/ext/wavpack/gstwavpackparse.c +++ b/ext/wavpack/gstwavpackparse.c @@ -41,6 +41,11 @@ #ifdef HAVE_CONFIG_H #include "config.h" #endif + +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 83a0547999..b6c065f509 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -45,6 +45,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/gst/flv/gstflvdemux.c b/gst/flv/gstflvdemux.c index 00a68e47b6..640301bcea 100644 --- a/gst/flv/gstflvdemux.c +++ b/gst/flv/gstflvdemux.c @@ -34,6 +34,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include "gstflvdemux.h" #include "gstflvmux.h" diff --git a/gst/imagefreeze/gstimagefreeze.c b/gst/imagefreeze/gstimagefreeze.c index 312f4aa155..6456040ae6 100644 --- a/gst/imagefreeze/gstimagefreeze.c +++ b/gst/imagefreeze/gstimagefreeze.c @@ -41,6 +41,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include "gstimagefreeze.h" diff --git a/gst/isomp4/gstqtmoovrecover.c b/gst/isomp4/gstqtmoovrecover.c index 80799724c8..0b832d32ba 100644 --- a/gst/isomp4/gstqtmoovrecover.c +++ b/gst/isomp4/gstqtmoovrecover.c @@ -65,6 +65,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c index b91b4b3ba7..037f29eb4e 100644 --- a/gst/isomp4/qtdemux.c +++ b/gst/isomp4/qtdemux.c @@ -47,6 +47,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include "gst/gst-i18n-plugin.h" #include diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 3d5a5694c9..eac7d65199 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -49,6 +49,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include #include diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c index 294b31cc5b..e4dd5697b5 100644 --- a/gst/rtpmanager/gstrtpssrcdemux.c +++ b/gst/rtpmanager/gstrtpssrcdemux.c @@ -44,6 +44,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include #include diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index b7b790f37e..cfdbea49f5 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -81,6 +81,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #ifdef HAVE_UNISTD_H #include #endif /* HAVE_UNISTD_H */ diff --git a/gst/videomixer/videomixer2.c b/gst/videomixer/videomixer2.c index b876c22fd1..585f3aafbe 100644 --- a/gst/videomixer/videomixer2.c +++ b/gst/videomixer/videomixer2.c @@ -86,6 +86,10 @@ #include "config.h" #endif +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include "videomixer2.h" diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index ef222e58e4..6c5e77b969 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -49,6 +49,11 @@ #ifdef HAVE_CONFIG_H #include "config.h" #endif + +/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex + * with newer GLib versions (>= 2.31.0) */ +#define GLIB_DISABLE_DEPRECATION_WARNINGS + #include #include From abd27e2caa247721897032864e0c0b28e21fb798 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 12 Dec 2011 10:18:14 +0000 Subject: [PATCH 15/65] tests: use atexit, g_atexit has been deprecated in glib master --- tests/check/elements/souphttpsrc.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/tests/check/elements/souphttpsrc.c b/tests/check/elements/souphttpsrc.c index 7a1b677714..9fc8f7cf96 100644 --- a/tests/check/elements/souphttpsrc.c +++ b/tests/check/elements/souphttpsrc.c @@ -23,6 +23,8 @@ # include "config.h" #endif +#include + #include #include #include @@ -454,7 +456,7 @@ souphttpsrc_suite (void) suite_add_tcase (s, tc_chain); run_server (&http_port, &https_port); - g_atexit (stop_server); + atexit (stop_server); tcase_add_test (tc_chain, test_first_buffer_has_offset); tcase_add_test (tc_chain, test_redirect_yes); tcase_add_test (tc_chain, test_redirect_no); From 271a89a30384620191f9d69235011df277195430 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Mon, 12 Dec 2011 10:38:20 +0000 Subject: [PATCH 16/65] build: dist glib-compat-private.h properly Add missing slash. --- Makefile.am | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Makefile.am b/Makefile.am index 862126dd13..fd94a81c3d 100644 --- a/Makefile.am +++ b/Makefile.am @@ -37,7 +37,7 @@ DISTCLEANFILES = _stdint.h noinst_HEADERS = \ gst-libs/gst/gettext.h \ - gst-libs/gst/gst-i18n-plugin.h + gst-libs/gst/gst-i18n-plugin.h \ gst-libs/gst/glib-compat-private.h ACLOCAL_AMFLAGS = -I m4 -I common/m4 From ed3f89fad57dc43f225a4fe973e0e28d27c03538 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 9 Feb 2011 15:31:22 +0100 Subject: [PATCH 17/65] qtdemux: increase parse tolerance for fuzzy file cases --- gst/isomp4/qtdemux.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c index 037f29eb4e..14d5413e45 100644 --- a/gst/isomp4/qtdemux.c +++ b/gst/isomp4/qtdemux.c @@ -5306,8 +5306,13 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl) GST_LOG_OBJECT (qtdemux, "%u timestamp blocks", stream->n_sample_times); /* make sure there's enough data */ - if (!qt_atom_parser_has_chunks (&stream->stts, stream->n_sample_times, 2 * 4)) - goto corrupt_file; + if (!qt_atom_parser_has_chunks (&stream->stts, stream->n_sample_times, 8)) { + stream->n_sample_times = gst_byte_reader_get_remaining (&stream->stts) / 8; + GST_LOG_OBJECT (qtdemux, "overriding to %u timestamp blocks", + stream->n_sample_times); + if (!stream->n_sample_times) + goto corrupt_file; + } /* sync sample atom */ stream->stps_present = FALSE; From 581ca6ce4eb5fd878879d85a660ead21438eeed8 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 12 Dec 2011 15:15:46 +0100 Subject: [PATCH 18/65] matroskademux: mind (un)signed in some timestamp arithmetic ... to avoid ending up with invalid (negative) duration. --- gst/matroska/matroska-demux.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index eac7d65199..8247d760ee 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -3485,8 +3485,8 @@ gst_matroska_demux_parse_blockgroup_or_simpleblock (GstMatroskaDemux * demux, GST_OBJECT_LOCK (demux); if (demux->common.segment.duration == -1 || - demux->common.segment.duration < - lace_time - demux->stream_start_time) { + demux->stream_start_time + demux->common.segment.duration < + last_stop_end) { gst_segment_set_duration (&demux->common.segment, GST_FORMAT_TIME, last_stop_end - demux->stream_start_time); GST_OBJECT_UNLOCK (demux); From 42310421b332db94dc5df6b9e1a072bdd8c7102d Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Mon, 12 Dec 2011 11:54:56 +0100 Subject: [PATCH 19/65] glib-compat: Add license boilerplate for LGPL --- gst-libs/gst/glib-compat-private.h | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/gst-libs/gst/glib-compat-private.h b/gst-libs/gst/glib-compat-private.h index d143d3257d..b9248e6861 100644 --- a/gst-libs/gst/glib-compat-private.h +++ b/gst-libs/gst/glib-compat-private.h @@ -3,6 +3,21 @@ * Functions copied from glib 2.10 * * Copyright 2005 David Schleef + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. */ #ifndef __GLIB_COMPAT_PRIVATE_H__ From 734e352b73dda701905058f9cc0e375f9f2e8363 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Mon, 12 Dec 2011 16:49:19 +0000 Subject: [PATCH 20/65] wavparse: add a ignore-length property This allows playing broken streams which write an incorrect length in their data chunks (such as, at least, one streaming camera). https://bugzilla.gnome.org/show_bug.cgi?id=665911 --- gst/wavparse/gstwavparse.c | 86 +++++++++++++++++++++++++++++++++++++- gst/wavparse/gstwavparse.h | 2 + 2 files changed, 86 insertions(+), 2 deletions(-) diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 6c5e77b969..6f91e6b3ef 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -87,6 +87,19 @@ static gboolean gst_wavparse_sink_event (GstPad * pad, GstEvent * event); static void gst_wavparse_loop (GstPad * pad); static gboolean gst_wavparse_srcpad_event (GstPad * pad, GstEvent * event); +static void gst_wavparse_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_wavparse_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +#define DEFAULT_IGNORE_LENGTH FALSE + +enum +{ + PROP_0, + PROP_IGNORE_LENGTH, +}; + static GstStaticPadTemplate sink_template_factory = GST_STATIC_PAD_TEMPLATE ("wavparse_sink", GST_PAD_SINK, @@ -134,6 +147,27 @@ gst_wavparse_class_init (GstWavParseClass * klass) object_class->dispose = gst_wavparse_dispose; + object_class->set_property = gst_wavparse_set_property; + object_class->get_property = gst_wavparse_get_property; + + /** + * GstWavParse:ignore-length + * + * This selects whether the length found in a data chunk + * should be ignored. This may be useful for streamed audio + * where the length is unknown until the end of streaming, + * and various software/hardware just puts some random value + * in there and hopes it doesn't break too much. + * + * Since: 0.10.36 + */ + g_object_class_install_property (object_class, PROP_IGNORE_LENGTH, + g_param_spec_boolean ("ignore-length", + "Ignore length", + "Ignore length from the Wave header", + DEFAULT_IGNORE_LENGTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS) + ); + gstelement_class->change_state = gst_wavparse_change_state; gstelement_class->send_event = gst_wavparse_send_event; } @@ -1348,6 +1382,10 @@ gst_wavparse_stream_headers (GstWavParse * wav) switch (tag) { case GST_RIFF_TAG_data:{ GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size); + if (wav->ignore_length) { + GST_DEBUG_OBJECT (wav, "Ignoring length"); + size = 0; + } if (wav->streaming) { gst_adapter_flush (wav->adapter, 8); gotdata = TRUE; @@ -1566,11 +1604,13 @@ gst_wavparse_stream_headers (GstWavParse * wav) if (gst_wavparse_calculate_duration (wav)) { gst_segment_init (&wav->segment, GST_FORMAT_TIME); - gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, wav->duration); + if (!wav->ignore_length) + gst_segment_set_duration (&wav->segment, GST_FORMAT_TIME, wav->duration); } else { /* no bitrate, let downstream peer do the math, we'll feed it bytes. */ gst_segment_init (&wav->segment, GST_FORMAT_BYTES); - gst_segment_set_duration (&wav->segment, GST_FORMAT_BYTES, wav->datasize); + if (!wav->ignore_length) + gst_segment_set_duration (&wav->segment, GST_FORMAT_BYTES, wav->datasize); } /* now we have all the info to perform a pending seek if any, if no @@ -2489,6 +2529,11 @@ gst_wavparse_pad_query (GstPad * pad, GstQuery * query) gint64 duration = 0; GstFormat format; + if (wav->ignore_length) { + res = FALSE; + break; + } + gst_query_parse_duration (query, &format, NULL); switch (format) { @@ -2657,6 +2702,43 @@ gst_wavparse_change_state (GstElement * element, GstStateChange transition) return ret; } +static void +gst_wavparse_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstWavParse *self; + + g_return_if_fail (GST_IS_WAVPARSE (object)); + self = GST_WAVPARSE (object); + + switch (prop_id) { + case PROP_IGNORE_LENGTH: + self->ignore_length = g_value_get_boolean (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec); + } + +} + +static void +gst_wavparse_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstWavParse *self; + + g_return_if_fail (GST_IS_WAVPARSE (object)); + self = GST_WAVPARSE (object); + + switch (prop_id) { + case PROP_IGNORE_LENGTH: + g_value_set_boolean (value, self->ignore_length); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (self, prop_id, pspec); + } +} + static gboolean plugin_init (GstPlugin * plugin) { diff --git a/gst/wavparse/gstwavparse.h b/gst/wavparse/gstwavparse.h index ecac96881e..972463f669 100644 --- a/gst/wavparse/gstwavparse.h +++ b/gst/wavparse/gstwavparse.h @@ -118,6 +118,8 @@ struct _GstWavParse { gboolean first; /* discont after seek */ gboolean discont; + + gboolean ignore_length; }; struct _GstWavParseClass { From b612550379b1a01257741b6b96df576b9f43f376 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Tue, 13 Dec 2011 11:42:40 +0000 Subject: [PATCH 21/65] wavparse: fix format specifier signedness Use unsigned specifiers for all unsigned values. A lot of the values used here are unsigned, and some can take high enough values that their signed counterpart will be negative. https://bugzilla.gnome.org/show_bug.cgi?id=665911 --- gst/wavparse/gstwavparse.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 6f91e6b3ef..3ef323bcf3 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -599,7 +599,7 @@ gst_wavparse_fmt (GstWavParse * wav) gst_element_add_pad (GST_ELEMENT_CAST (wav), wav->srcpad); gst_element_no_more_pads (GST_ELEMENT_CAST (wav)); - GST_DEBUG ("frequency %d, channels %d", wav->rate, wav->channels); + GST_DEBUG ("frequency %u, channels %u", wav->rate, wav->channels); return TRUE; @@ -640,8 +640,8 @@ gst_wavparse_other (GstWavParse * wav) GST_WARNING_OBJECT (wav, "could not peek head"); return FALSE; } - GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %d", tag, - (gchar *) & tag, length); + GST_DEBUG_OBJECT (wav, "got tag (%08x) %4.4s, length %u", tag, + (const gchar *) &tag, length); switch (tag) { case GST_RIFF_TAG_LIST: @@ -1094,7 +1094,7 @@ gst_wavparse_peek_chunk_info (GstWavParse * wav, guint32 * tag, guint32 * size) *tag = GST_READ_UINT32_LE (data); *size = GST_READ_UINT32_LE (data + 4); - GST_DEBUG ("Next chunk size is %d bytes, type %" GST_FOURCC_FORMAT, *size, + GST_DEBUG ("Next chunk size is %u bytes, type %" GST_FOURCC_FORMAT, *size, GST_FOURCC_ARGS (*tag)); return TRUE; @@ -1124,7 +1124,7 @@ gst_wavparse_peek_chunk (GstWavParse * wav, guint32 * tag, guint32 * size) * so we throw poor man's exception, which can be caught if caller really * wants to handle 0 size chunk */ if (!(*size) || (*size) >= (1 << 30)) { - GST_INFO ("Invalid/unexpected chunk size %d for tag %" GST_FOURCC_FORMAT, + GST_INFO ("Invalid/unexpected chunk size %u for tag %" GST_FOURCC_FORMAT, *size, GST_FOURCC_ARGS (*tag)); /* chain should give up */ wav->abort_buffering = TRUE; @@ -1297,7 +1297,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) { /* Note: workaround for mp2/mp3 embedded in wav, that relies on the * bitrate inside the mpeg stream */ - GST_INFO ("resetting bps from %d to 0 for mp2/3", wav->av_bps); + GST_INFO ("resetting bps from %u to 0 for mp2/3", wav->av_bps); wav->bps = 0; break; } @@ -1381,7 +1381,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) */ switch (tag) { case GST_RIFF_TAG_data:{ - GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %d", size); + GST_DEBUG_OBJECT (wav, "Got 'data' TAG, size : %u", size); if (wav->ignore_length) { GST_DEBUG_OBJECT (wav, "Ignoring length"); size = 0; @@ -1410,7 +1410,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) /* We will continue parsing tags 'till end */ wav->offset += size; } - GST_DEBUG_OBJECT (wav, "datasize = %d", size); + GST_DEBUG_OBJECT (wav, "datasize = %u", size); break; } case GST_RIFF_TAG_fact:{ @@ -1424,7 +1424,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) /* need more data */ goto exit; } - GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk", + GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk", data_size, size); break; } @@ -1469,7 +1469,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) /* need more data */ goto exit; } - GST_DEBUG_OBJECT (wav, "need %d, available %d; ignoring chunk", + GST_DEBUG_OBJECT (wav, "need %u, available %u; ignoring chunk", data_size, size); break; } @@ -1597,7 +1597,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) wav->bps = (guint32) gst_util_uint64_scale ((guint64) wav->rate, wav->datasize, (guint64) wav->fact); - GST_INFO_OBJECT (wav, "calculated bps : %d, enabling VBR", wav->bps); + GST_INFO_OBJECT (wav, "calculated bps : %u, enabling VBR", wav->bps); #endif wav->vbr = TRUE; } @@ -1637,7 +1637,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) if (wav->blockalign > 0) wav->max_buf_size -= (wav->max_buf_size % wav->blockalign); - GST_DEBUG_OBJECT (wav, "max buffer size %d", wav->max_buf_size); + GST_DEBUG_OBJECT (wav, "max buffer size %u", wav->max_buf_size); return GST_FLOW_OK; @@ -1705,7 +1705,7 @@ no_bytes_per_sample: unknown_format: { GST_ELEMENT_ERROR (wav, STREAM, TYPE_NOT_FOUND, (NULL), - ("No caps found for format 0x%x, %d channels, %d Hz", + ("No caps found for format 0x%x, %u channels, %u Hz", wav->format, wav->channels, wav->rate)); goto fail; } @@ -1908,13 +1908,13 @@ iterate_adapter: if (G_UNLIKELY (extra)) { extra = wav->bytes_per_sample - extra; if (extra <= avail) { - GST_DEBUG_OBJECT (wav, "flushing %d bytes to sample boundary", extra); + GST_DEBUG_OBJECT (wav, "flushing %u bytes to sample boundary", extra); gst_adapter_flush (wav->adapter, extra); wav->offset += extra; wav->dataleft -= extra; goto iterate_adapter; } else { - GST_DEBUG_OBJECT (wav, "flushing %d bytes", avail); + GST_DEBUG_OBJECT (wav, "flushing %u bytes", avail); gst_adapter_clear (wav->adapter); wav->offset += avail; wav->dataleft -= avail; @@ -1923,7 +1923,7 @@ iterate_adapter: } if (avail < desired) { - GST_LOG_OBJECT (wav, "Got only %d bytes of data from the sinkpad", avail); + GST_LOG_OBJECT (wav, "Got only %u bytes of data from the sinkpad", avail); return GST_FLOW_OK; } From ecec3859d838d334bda4c48bdcfd901f63b6bc2e Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Tue, 13 Dec 2011 11:46:43 +0000 Subject: [PATCH 22/65] wavparse: avoid using floating point unnecessarily https://bugzilla.gnome.org/show_bug.cgi?id=665911 --- gst/wavparse/gstwavparse.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 3ef323bcf3..b4e5df4ecd 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -1302,7 +1302,7 @@ gst_wavparse_stream_headers (GstWavParse * wav) break; } case GST_RIFF_WAVE_FORMAT_PCM: - if (wav->blockalign > wav->channels * (guint) ceil (wav->depth / 8.0)) + if (wav->blockalign > wav->channels * ((wav->depth + 7) / 8)) goto invalid_blockalign; /* fall through */ default: @@ -1686,7 +1686,7 @@ invalid_blockalign: { GST_ELEMENT_ERROR (wav, STREAM, FAILED, (NULL), ("Stream claims blockalign = %u, which is more than %u - invalid data", - wav->blockalign, wav->channels * (guint) ceil (wav->depth / 8.0))); + wav->blockalign, wav->channels * ((wav->depth + 7) / 8))); goto fail; } invalid_bps: From 4a5364d7f6db3fad2811d25b8c1cfae61fbec3c9 Mon Sep 17 00:00:00 2001 From: Thiago Santos Date: Mon, 12 Dec 2011 11:51:06 -0300 Subject: [PATCH 23/65] qtmux: make debug message more useful Add information about the taglist and which pad received the tag event on the debug logging. --- gst/isomp4/gstqtmux.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index a83835c183..ed32e4954f 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -3273,8 +3273,9 @@ gst_qt_mux_sink_event (GstPad * pad, GstEvent * event) GST_OBJECT_LOCK (qtmux); mode = gst_tag_setter_get_tag_merge_mode (setter); - GST_DEBUG_OBJECT (qtmux, "received tag event"); gst_event_parse_tag (event, &list); + GST_DEBUG_OBJECT (qtmux, "received tag event on pad %s:%s : %" + GST_PTR_FORMAT, GST_DEBUG_PAD_NAME (pad), list); gst_tag_setter_merge_tags (setter, list, mode); GST_OBJECT_UNLOCK (qtmux); From fd880965773834e167841f54166e8c5e761c1e5b Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 13 Dec 2011 17:02:01 +0100 Subject: [PATCH 24/65] matroskademux: tweak final closing segment sending ... to avoid it interfering with (sparse) stream syncing. --- gst/matroska/matroska-demux.c | 12 ++++++++---- 1 file changed, 8 insertions(+), 4 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 8247d760ee..00162e2108 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -4432,13 +4432,17 @@ pause: /* Close the segment, i.e. update segment stop with the duration * if no stop was set */ if (GST_CLOCK_TIME_IS_VALID (demux->last_stop_end) && - !GST_CLOCK_TIME_IS_VALID (demux->common.segment.stop)) { + !GST_CLOCK_TIME_IS_VALID (demux->common.segment.stop) && + GST_CLOCK_TIME_IS_VALID (demux->common.segment.start) && + demux->last_stop_end > demux->common.segment.start) { + /* arrange to accumulate duration downstream, but avoid sending + * newsegment with decreasing start (w.r.t. sync newsegment events) */ GstEvent *event = gst_event_new_new_segment_full (TRUE, demux->common.segment.rate, demux->common.segment.applied_rate, demux->common.segment.format, - demux->common.segment.start, - MAX (demux->last_stop_end, demux->common.segment.start), - demux->common.segment.time); + demux->last_stop_end, demux->last_stop_end, + demux->common.segment.time + (demux->last_stop_end - + demux->common.segment.start)); gst_matroska_demux_send_event (demux, event); } From 00ed34d2eb67be63a89e8fb8735b8501dccd4e8d Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 13 Dec 2011 18:15:18 +0100 Subject: [PATCH 25/65] matroskademux: cater for safer arithmetic with global start time --- gst/matroska/matroska-demux.c | 24 ++++++++++++++++-------- 1 file changed, 16 insertions(+), 8 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 00162e2108..81f83f0150 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -1373,10 +1373,12 @@ gst_matroska_demux_query (GstMatroskaDemux * demux, GstPad * pad, GST_OBJECT_LOCK (demux); if (context) gst_query_set_position (query, GST_FORMAT_TIME, - context->pos - demux->stream_start_time); + MAX (context->pos, demux->stream_start_time) - + demux->stream_start_time); else gst_query_set_position (query, GST_FORMAT_TIME, - demux->common.segment.last_stop - demux->stream_start_time); + MAX (demux->common.segment.last_stop, demux->stream_start_time) - + demux->stream_start_time); GST_OBJECT_UNLOCK (demux); } else if (format == GST_FORMAT_DEFAULT && context && context->default_duration) { @@ -1745,8 +1747,11 @@ gst_matroska_demux_search_pos (GstMatroskaDemux * demux, GstClockTime time) otime = demux->common.segment.last_stop; GST_OBJECT_UNLOCK (demux); + /* sanitize */ + time = MAX (time, demux->stream_start_time); + /* avoid division by zero in first estimation below */ - if (otime == 0) + if (otime <= demux->stream_start_time) otime = time; retry: @@ -2015,9 +2020,9 @@ next: if (keyunit) { GST_DEBUG_OBJECT (demux, "seek to key unit, adjusting segment start to %" GST_TIME_FORMAT, GST_TIME_ARGS (entry->time)); - seeksegment.start = entry->time; - seeksegment.last_stop = entry->time; - seeksegment.time = entry->time - demux->stream_start_time; + seeksegment.start = MAX (entry->time, demux->stream_start_time); + seeksegment.last_stop = seeksegment.start; + seeksegment.time = seeksegment.start - demux->stream_start_time; } exit: @@ -3287,6 +3292,8 @@ gst_matroska_demux_parse_blockgroup_or_simpleblock (GstMatroskaDemux * demux, /* need to refresh segment info ASAP */ if (GST_CLOCK_TIME_IS_VALID (lace_time) && demux->need_newsegment) { + guint64 clace_time; + GST_DEBUG_OBJECT (demux, "generating segment starting at %" GST_TIME_FORMAT, GST_TIME_ARGS (lace_time)); @@ -3296,9 +3303,10 @@ gst_matroska_demux_parse_blockgroup_or_simpleblock (GstMatroskaDemux * demux, "Setting stream start time to %" GST_TIME_FORMAT, GST_TIME_ARGS (lace_time)); } + clace_time = MAX (lace_time, demux->stream_start_time); gst_segment_set_newsegment (&demux->common.segment, FALSE, - demux->common.segment.rate, GST_FORMAT_TIME, lace_time, - GST_CLOCK_TIME_NONE, lace_time - demux->stream_start_time); + demux->common.segment.rate, GST_FORMAT_TIME, clace_time, + GST_CLOCK_TIME_NONE, clace_time - demux->stream_start_time); /* now convey our segment notion downstream */ gst_matroska_demux_send_event (demux, gst_event_new_new_segment (FALSE, demux->common.segment.rate, demux->common.segment.format, From ebfc77da194911ebbfa94af3fab06203a74004bf Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Tue, 13 Dec 2011 18:18:45 +0100 Subject: [PATCH 26/65] matroskademux: filter bogus index entries with missing block number ... to avoid contradictory information resulting in seeks sending more downstream than needed for the corresponding segment. --- gst/matroska/matroska-read-common.c | 15 +++++++++++++++ 1 file changed, 15 insertions(+) diff --git a/gst/matroska/matroska-read-common.c b/gst/matroska/matroska-read-common.c index 5fd178fe38..188dea91a2 100644 --- a/gst/matroska/matroska-read-common.c +++ b/gst/matroska/matroska-read-common.c @@ -982,6 +982,21 @@ gst_matroska_read_common_parse_index_cuetrack (GstMatroskaReadCommon * common, DEBUG_ELEMENT_STOP (common, ebml, "CueTrackPositions", ret); + /* (e.g.) lavf typically creates entries without a block number, + * which is bogus and leads to contradictory information */ + if (common->index->len) { + GstMatroskaIndex *last_idx; + + last_idx = &g_array_index (common->index, GstMatroskaIndex, + common->index->len - 1); + if (last_idx->block == idx.block && last_idx->pos == idx.pos && + last_idx->track == idx.track && idx.time > last_idx->time) { + GST_DEBUG_OBJECT (common, "Cue entry refers to same location, " + "but has different time than previous entry; discarding"); + idx.track = 0; + } + } + if ((ret == GST_FLOW_OK || ret == GST_FLOW_UNEXPECTED) && idx.pos != (guint64) - 1 && idx.track > 0) { g_array_append_val (common->index, idx); From cf344d50b1a9e304695c664ab9fe0b77090de1d8 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 14 Dec 2011 18:34:25 +0000 Subject: [PATCH 27/65] cairotextoverlay: port to GstCollectPads2 --- ext/cairo/gsttextoverlay.c | 38 +++++++++++++++++++------------------- ext/cairo/gsttextoverlay.h | 8 ++++---- 2 files changed, 23 insertions(+), 23 deletions(-) diff --git a/ext/cairo/gsttextoverlay.c b/ext/cairo/gsttextoverlay.c index 8cb39cc11e..f8da308e4d 100644 --- a/ext/cairo/gsttextoverlay.c +++ b/ext/cairo/gsttextoverlay.c @@ -106,7 +106,7 @@ static gboolean gst_text_overlay_setcaps (GstPad * pad, GstCaps * caps); static GstPadLinkReturn gst_text_overlay_text_pad_linked (GstPad * pad, GstPad * peer); static void gst_text_overlay_text_pad_unlinked (GstPad * pad); -static GstFlowReturn gst_text_overlay_collected (GstCollectPads * pads, +static GstFlowReturn gst_text_overlay_collected (GstCollectPads2 * pads, gpointer data); static void gst_text_overlay_finalize (GObject * object); static void gst_text_overlay_font_init (GstCairoTextOverlay * overlay); @@ -215,7 +215,7 @@ gst_text_overlay_finalize (GObject * object) { GstCairoTextOverlay *overlay = GST_CAIRO_TEXT_OVERLAY (object); - gst_collect_pads_stop (overlay->collect); + gst_collect_pads2_stop (overlay->collect); gst_object_unref (overlay->collect); g_free (overlay->text_fill_image); @@ -279,16 +279,16 @@ gst_text_overlay_init (GstCairoTextOverlay * overlay, overlay->fps_n = 0; overlay->fps_d = 1; - overlay->collect = gst_collect_pads_new (); + overlay->collect = gst_collect_pads2_new (); - gst_collect_pads_set_function (overlay->collect, + gst_collect_pads2_set_function (overlay->collect, GST_DEBUG_FUNCPTR (gst_text_overlay_collected), overlay); - overlay->video_collect_data = gst_collect_pads_add_pad (overlay->collect, - overlay->video_sinkpad, sizeof (GstCollectData)); + overlay->video_collect_data = gst_collect_pads2_add_pad (overlay->collect, + overlay->video_sinkpad, sizeof (GstCollectData2)); /* FIXME: hacked way to override/extend the event function of - * GstCollectPads; because it sets its own event function giving the + * GstCollectPads2; because it sets its own event function giving the * element no access to events. Nicked from avimux. */ overlay->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (overlay->video_sinkpad); @@ -640,8 +640,8 @@ gst_text_overlay_text_pad_linked (GstPad * pad, GstPad * peer) GST_DEBUG_OBJECT (overlay, "Text pad linked"); if (overlay->text_collect_data == NULL) { - overlay->text_collect_data = gst_collect_pads_add_pad (overlay->collect, - overlay->text_sinkpad, sizeof (GstCollectData)); + overlay->text_collect_data = gst_collect_pads2_add_pad (overlay->collect, + overlay->text_sinkpad, sizeof (GstCollectData2)); } overlay->need_render = TRUE; @@ -660,7 +660,7 @@ gst_text_overlay_text_pad_unlinked (GstPad * pad) GST_DEBUG_OBJECT (overlay, "Text pad unlinked"); if (overlay->text_collect_data) { - gst_collect_pads_remove_pad (overlay->collect, overlay->text_sinkpad); + gst_collect_pads2_remove_pad (overlay->collect, overlay->text_sinkpad); overlay->text_collect_data = NULL; } @@ -807,7 +807,7 @@ gst_text_overlay_pop_video (GstCairoTextOverlay * overlay) { GstBuffer *buf; - buf = gst_collect_pads_pop (overlay->collect, overlay->video_collect_data); + buf = gst_collect_pads2_pop (overlay->collect, overlay->video_collect_data); g_return_if_fail (buf != NULL); gst_buffer_unref (buf); } @@ -818,7 +818,7 @@ gst_text_overlay_pop_text (GstCairoTextOverlay * overlay) GstBuffer *buf; if (overlay->text_collect_data) { - buf = gst_collect_pads_pop (overlay->collect, overlay->text_collect_data); + buf = gst_collect_pads2_pop (overlay->collect, overlay->text_collect_data); g_return_if_fail (buf != NULL); gst_buffer_unref (buf); } @@ -828,7 +828,7 @@ gst_text_overlay_pop_text (GstCairoTextOverlay * overlay) /* This function is called when there is data on all pads */ static GstFlowReturn -gst_text_overlay_collected (GstCollectPads * pads, gpointer data) +gst_text_overlay_collected (GstCollectPads2 * pads, gpointer data) { GstCairoTextOverlay *overlay; GstFlowReturn ret = GST_FLOW_OK; @@ -842,14 +842,14 @@ gst_text_overlay_collected (GstCollectPads * pads, gpointer data) GST_DEBUG ("Collecting"); - video_frame = gst_collect_pads_peek (overlay->collect, + video_frame = gst_collect_pads2_peek (overlay->collect, overlay->video_collect_data); /* send EOS if video stream EOSed regardless of text stream */ if (video_frame == NULL) { GST_DEBUG ("Video stream at EOS"); if (overlay->text_collect_data) { - text_buf = gst_collect_pads_pop (overlay->collect, + text_buf = gst_collect_pads2_pop (overlay->collect, overlay->text_collect_data); } gst_pad_push_event (overlay->srcpad, gst_event_new_eos ()); @@ -892,7 +892,7 @@ gst_text_overlay_collected (GstCollectPads * pads, gpointer data) goto done; } - text_buf = gst_collect_pads_peek (overlay->collect, + text_buf = gst_collect_pads2_peek (overlay->collect, overlay->text_collect_data); /* just push the video frame if the text stream has EOSed */ @@ -1004,7 +1004,7 @@ gst_text_overlay_video_event (GstPad * pad, GstEvent * event) gst_pad_push_event (overlay->srcpad, event); } - /* now GstCollectPads can take care of the rest, e.g. EOS */ + /* now GstCollectPads2 can take care of the rest, e.g. EOS */ ret = overlay->collect_event (pad, event); gst_object_unref (overlay); return ret; @@ -1018,12 +1018,12 @@ gst_text_overlay_change_state (GstElement * element, GstStateChange transition) switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_collect_pads_start (overlay->collect); + gst_collect_pads2_start (overlay->collect); break; case GST_STATE_CHANGE_PAUSED_TO_READY: /* need to unblock the collectpads before calling the * parent change_state so that streaming can finish */ - gst_collect_pads_stop (overlay->collect); + gst_collect_pads2_stop (overlay->collect); break; default: break; diff --git a/ext/cairo/gsttextoverlay.h b/ext/cairo/gsttextoverlay.h index dbb21546ff..1f98d3d58b 100644 --- a/ext/cairo/gsttextoverlay.h +++ b/ext/cairo/gsttextoverlay.h @@ -3,7 +3,7 @@ #define __GST_CAIRO_TEXT_OVERLAY_H__ #include -#include +#include G_BEGIN_DECLS @@ -45,9 +45,9 @@ struct _GstCairoTextOverlay { GstPad *text_sinkpad; GstPad *srcpad; - GstCollectPads *collect; - GstCollectData *video_collect_data; - GstCollectData *text_collect_data; + GstCollectPads2 *collect; + GstCollectData2 *video_collect_data; + GstCollectData2 *text_collect_data; GstPadEventFunction collect_event; gint width; From a89e2b7e48f920b7d8a239cb52e0e1498153b919 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 14 Dec 2011 18:38:09 +0000 Subject: [PATCH 28/65] avimux: port to GstCollectPads2 --- gst/avi/gstavimux.c | 28 ++++++++++++++-------------- gst/avi/gstavimux.h | 8 ++++---- 2 files changed, 18 insertions(+), 18 deletions(-) diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index fe50cef9e2..0089ef5f46 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -195,7 +195,7 @@ static void gst_avi_mux_class_init (GstAviMuxClass * klass); static void gst_avi_mux_init (GstAviMux * avimux); static void gst_avi_mux_pad_reset (GstAviPad * avipad, gboolean free); -static GstFlowReturn gst_avi_mux_collect_pads (GstCollectPads * pads, +static GstFlowReturn gst_avi_mux_collect_pads (GstCollectPads2 * pads, GstAviMux * avimux); static gboolean gst_avi_mux_handle_event (GstPad * pad, GstEvent * event); static GstPad *gst_avi_mux_request_new_pad (GstElement * element, @@ -418,9 +418,9 @@ gst_avi_mux_init (GstAviMux * avimux) /* property */ avimux->enable_large_avi = DEFAULT_BIGFILE; - avimux->collect = gst_collect_pads_new (); - gst_collect_pads_set_function (avimux->collect, - (GstCollectPadsFunction) (GST_DEBUG_FUNCPTR (gst_avi_mux_collect_pads)), + avimux->collect = gst_collect_pads2_new (); + gst_collect_pads2_set_function (avimux->collect, + (GstCollectPads2Function) (GST_DEBUG_FUNCPTR (gst_avi_mux_collect_pads)), avimux); /* set to clean state */ @@ -1024,11 +1024,11 @@ gst_avi_mux_request_new_pad (GstElement * element, g_free (name); - avipad->collect = gst_collect_pads_add_pad (avimux->collect, + avipad->collect = gst_collect_pads2_add_pad (avimux->collect, newpad, sizeof (GstAviCollectData)); ((GstAviCollectData *) (avipad->collect))->avipad = avipad; /* FIXME: hacked way to override/extend the event function of - * GstCollectPads; because it sets its own event function giving the + * GstCollectPads2; because it sets its own event function giving the * element no access to events */ avimux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad); gst_pad_set_event_function (newpad, @@ -1085,7 +1085,7 @@ gst_avi_mux_release_pad (GstElement * element, GstPad * pad) * as it also represent number of streams present */ avipad->collect = NULL; GST_DEBUG_OBJECT (avimux, "removed pad '%s'", GST_PAD_NAME (pad)); - gst_collect_pads_remove_pad (avimux->collect, pad); + gst_collect_pads2_remove_pad (avimux->collect, pad); gst_element_remove_pad (element, pad); /* if not started yet, we can remove any sign this pad ever existed */ /* in this case _start will take care of the real pad count */ @@ -1882,7 +1882,7 @@ gst_avi_mux_handle_event (GstPad * pad, GstEvent * event) break; } - /* now GstCollectPads can take care of the rest, e.g. EOS */ + /* now GstCollectPads2 can take care of the rest, e.g. EOS */ ret = avimux->collect_event (pad, event); gst_object_unref (avimux); @@ -1911,7 +1911,7 @@ gst_avi_mux_do_buffer (GstAviMux * avimux, GstAviPad * avipad) gulong total_size, pad_bytes = 0; guint flags; - data = gst_collect_pads_pop (avimux->collect, avipad->collect); + data = gst_collect_pads2_pop (avimux->collect, avipad->collect); /* arrange downstream running time */ data = gst_buffer_make_metadata_writable (data); GST_BUFFER_TIMESTAMP (data) = @@ -2037,7 +2037,7 @@ gst_avi_mux_do_one_buffer (GstAviMux * avimux) if (!avipad->hdr.fcc_handler) goto not_negotiated; - buffer = gst_collect_pads_peek (avimux->collect, avipad->collect); + buffer = gst_collect_pads2_peek (avimux->collect, avipad->collect); if (!buffer) continue; time = GST_BUFFER_TIMESTAMP (buffer); @@ -2050,7 +2050,7 @@ gst_avi_mux_do_one_buffer (GstAviMux * avimux) if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) { GST_DEBUG_OBJECT (avimux, "clipping buffer on pad %s outside segment", GST_PAD_NAME (avipad->collect->pad)); - buffer = gst_collect_pads_pop (avimux->collect, avipad->collect); + buffer = gst_collect_pads2_pop (avimux->collect, avipad->collect); gst_buffer_unref (buffer); return GST_FLOW_OK; } @@ -2089,7 +2089,7 @@ not_negotiated: } static GstFlowReturn -gst_avi_mux_collect_pads (GstCollectPads * pads, GstAviMux * avimux) +gst_avi_mux_collect_pads (GstCollectPads2 * pads, GstAviMux * avimux) { GstFlowReturn res; @@ -2148,12 +2148,12 @@ gst_avi_mux_change_state (GstElement * element, GstStateChange transition) switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_collect_pads_start (avimux->collect); + gst_collect_pads2_start (avimux->collect); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_collect_pads_stop (avimux->collect); + gst_collect_pads2_stop (avimux->collect); break; default: break; diff --git a/gst/avi/gstavimux.h b/gst/avi/gstavimux.h index 0d903928e0..1457fce938 100644 --- a/gst/avi/gstavimux.h +++ b/gst/avi/gstavimux.h @@ -23,7 +23,7 @@ #include -#include +#include #include #include "avi-ids.h" @@ -74,7 +74,7 @@ typedef GstFlowReturn (*GstAviPadHook) (GstAviMux * avi, GstAviPad * avipad, struct _GstAviPad { /* do not extend, link to it */ /* is NULL if original sink request pad has been removed */ - GstCollectData *collect; + GstCollectData2 *collect; /* type */ gboolean is_video; @@ -129,7 +129,7 @@ typedef struct _GstAviAudioPad { typedef struct _GstAviCollectData { /* extend the CollectData */ - GstCollectData collect; + GstCollectData2 collect; GstAviPad *avipad; } GstAviCollectData; @@ -143,7 +143,7 @@ struct _GstAviMux { GSList *sinkpads; /* video restricted to 1 pad */ guint video_pads, audio_pads; - GstCollectPads *collect; + GstCollectPads2 *collect; GstPadEventFunction collect_event; /* the AVI header */ From 44d001346a0434f774a251552f16aaea59c916e3 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 14 Dec 2011 18:52:37 +0000 Subject: [PATCH 29/65] flxmux: port to GstCollectPads2 --- gst/flv/gstflvmux.c | 30 +++++++++++++++--------------- gst/flv/gstflvmux.h | 6 +++--- 2 files changed, 18 insertions(+), 18 deletions(-) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index d4c5cb5fbc..fc4cf7606e 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -99,7 +99,7 @@ GST_BOILERPLATE_FULL (GstFlvMux, gst_flv_mux, GstElement, GST_TYPE_ELEMENT, static void gst_flv_mux_finalize (GObject * object); static GstFlowReturn -gst_flv_mux_collected (GstCollectPads * pads, gpointer user_data); +gst_flv_mux_collected (GstCollectPads2 * pads, gpointer user_data); static gboolean gst_flv_mux_handle_src_event (GstPad * pad, GstEvent * event); static GstPad *gst_flv_mux_request_new_pad (GstElement * element, @@ -197,9 +197,9 @@ gst_flv_mux_init (GstFlvMux * mux, GstFlvMuxClass * g_class) mux->new_tags = FALSE; - mux->collect = gst_collect_pads_new (); - gst_collect_pads_set_function (mux->collect, - (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_flv_mux_collected), mux); + mux->collect = gst_collect_pads2_new (); + gst_collect_pads2_set_function (mux->collect, + (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_flv_mux_collected), mux); gst_flv_mux_reset (GST_ELEMENT (mux)); } @@ -280,7 +280,7 @@ gst_flv_mux_handle_sink_event (GstPad * pad, GstEvent * event) break; } - /* now GstCollectPads can take care of the rest, e.g. EOS */ + /* now GstCollectPads2 can take care of the rest, e.g. EOS */ if (ret) ret = mux->collect_event (pad, event); gst_object_unref (mux); @@ -536,14 +536,14 @@ gst_flv_mux_request_new_pad (GstElement * element, pad = gst_pad_new_from_template (templ, name); cpad = (GstFlvPad *) - gst_collect_pads_add_pad (mux->collect, pad, sizeof (GstFlvPad)); + gst_collect_pads2_add_pad (mux->collect, pad, sizeof (GstFlvPad)); cpad->audio_codec_data = NULL; cpad->video_codec_data = NULL; gst_flv_mux_reset_pad (mux, cpad, video); /* FIXME: hacked way to override/extend the event function of - * GstCollectPads; because it sets its own event function giving the + * GstCollectPads2; because it sets its own event function giving the * element no access to events. */ mux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (pad); @@ -564,7 +564,7 @@ gst_flv_mux_release_pad (GstElement * element, GstPad * pad) GstFlvPad *cpad = (GstFlvPad *) gst_pad_get_element_private (pad); gst_flv_mux_reset_pad (mux, cpad, cpad->video); - gst_collect_pads_remove_pad (mux->collect, pad); + gst_collect_pads2_remove_pad (mux->collect, pad); gst_element_remove_pad (element, pad); } @@ -756,7 +756,7 @@ gst_flv_mux_create_metadata (GstFlvMux * mux) guint64 dur; for (l = mux->collect->data; l; l = l->next) { - GstCollectData *cdata = l->data; + GstCollectData2 *cdata = l->data; fmt = GST_FORMAT_TIME; @@ -1201,7 +1201,7 @@ gst_flv_mux_write_buffer (GstFlvMux * mux, GstFlvPad * cpad) { GstBuffer *tag; GstBuffer *buffer = - gst_collect_pads_pop (mux->collect, (GstCollectData *) cpad); + gst_collect_pads2_pop (mux->collect, (GstCollectData2 *) cpad); GstFlowReturn ret; /* arrange downstream running time */ @@ -1390,7 +1390,7 @@ gst_flv_mux_rewrite_header (GstFlvMux * mux) } static GstFlowReturn -gst_flv_mux_collected (GstCollectPads * pads, gpointer user_data) +gst_flv_mux_collected (GstCollectPads2 * pads, gpointer user_data) { GstFlvMux *mux = GST_FLV_MUX (user_data); GstFlvPad *best; @@ -1428,7 +1428,7 @@ gst_flv_mux_collected (GstCollectPads * pads, gpointer user_data) best_time = GST_CLOCK_TIME_NONE; for (sl = mux->collect->data; sl; sl = sl->next) { GstFlvPad *cpad = sl->data; - GstBuffer *buffer = gst_collect_pads_peek (pads, (GstCollectData *) cpad); + GstBuffer *buffer = gst_collect_pads2_peek (pads, (GstCollectData2 *) cpad); GstClockTime time; if (!buffer) @@ -1453,7 +1453,7 @@ gst_flv_mux_collected (GstCollectPads * pads, gpointer user_data) if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) { GST_DEBUG_OBJECT (mux, "clipping buffer on pad %s outside segment", GST_PAD_NAME (cpad->collect.pad)); - buffer = gst_collect_pads_pop (pads, (GstCollectData *) cpad); + buffer = gst_collect_pads2_pop (pads, (GstCollectData2 *) cpad); gst_buffer_unref (buffer); return GST_FLOW_OK; } @@ -1533,12 +1533,12 @@ gst_flv_mux_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_collect_pads_start (mux->collect); + gst_collect_pads2_start (mux->collect); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_collect_pads_stop (mux->collect); + gst_collect_pads2_stop (mux->collect); break; default: break; diff --git a/gst/flv/gstflvmux.h b/gst/flv/gstflvmux.h index fff51b9b32..ad38abe9ee 100644 --- a/gst/flv/gstflvmux.h +++ b/gst/flv/gstflvmux.h @@ -22,7 +22,7 @@ #define __GST_FLV_MUX_H__ #include -#include +#include G_BEGIN_DECLS @@ -39,7 +39,7 @@ G_BEGIN_DECLS typedef struct { - GstCollectData collect; + GstCollectData2 collect; gboolean video; @@ -65,7 +65,7 @@ typedef struct _GstFlvMux { GstElement element; GstPad *srcpad; - GstCollectPads *collect; + GstCollectPads2 *collect; /* */ GstPadEventFunction collect_event; From 3135cab5a36a900784bd7b4e1bce77bbf8f3c8d0 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 14 Dec 2011 18:55:36 +0000 Subject: [PATCH 30/65] interleave: port to GstCollectPads2 --- gst/interleave/interleave.c | 36 ++++++++++++++++++------------------ gst/interleave/interleave.h | 4 ++-- 2 files changed, 20 insertions(+), 20 deletions(-) diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c index bc672d10c0..8815e1c049 100644 --- a/gst/interleave/interleave.c +++ b/gst/interleave/interleave.c @@ -227,7 +227,7 @@ static gboolean gst_interleave_sink_setcaps (GstPad * pad, GstCaps * caps); static GstCaps *gst_interleave_sink_getcaps (GstPad * pad); -static GstFlowReturn gst_interleave_collected (GstCollectPads * pads, +static GstFlowReturn gst_interleave_collected (GstCollectPads2 * pads, GstInterleave * self); static void @@ -405,9 +405,9 @@ gst_interleave_init (GstInterleave * self, GstInterleaveClass * klass) gst_element_add_pad (GST_ELEMENT (self), self->src); - self->collect = gst_collect_pads_new (); - gst_collect_pads_set_function (self->collect, - (GstCollectPadsFunction) gst_interleave_collected, self); + self->collect = gst_collect_pads2_new (); + gst_collect_pads2_set_function (self->collect, + (GstCollectPads2Function) gst_interleave_collected, self); self->input_channel_positions = g_value_array_new (0); self->channel_positions_from_input = TRUE; @@ -498,10 +498,10 @@ gst_interleave_request_new_pad (GstElement * element, GstPadTemplate * templ, gst_pad_set_getcaps_function (new_pad, GST_DEBUG_FUNCPTR (gst_interleave_sink_getcaps)); - gst_collect_pads_add_pad (self->collect, new_pad, sizeof (GstCollectData)); + gst_collect_pads2_add_pad (self->collect, new_pad, sizeof (GstCollectData2)); /* FIXME: hacked way to override/extend the event function of - * GstCollectPads; because it sets its own event function giving the + * GstCollectPads2; because it sets its own event function giving the * element no access to events */ self->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (new_pad); gst_pad_set_event_function (new_pad, @@ -547,7 +547,7 @@ not_sink_pad: could_not_add: { GST_DEBUG_OBJECT (self, "could not add pad %s", GST_PAD_NAME (new_pad)); - gst_collect_pads_remove_pad (self->collect, new_pad); + gst_collect_pads2_remove_pad (self->collect, new_pad); gst_object_unref (new_pad); return NULL; } @@ -601,7 +601,7 @@ gst_interleave_release_pad (GstElement * element, GstPad * pad) GST_OBJECT_UNLOCK (self->collect); - gst_collect_pads_remove_pad (self->collect, pad); + gst_collect_pads2_remove_pad (self->collect, pad); gst_element_remove_pad (element, pad); } @@ -623,7 +623,7 @@ gst_interleave_change_state (GstElement * element, GstStateChange transition) self->segment_position = 0; self->segment_rate = 1.0; gst_segment_init (&self->segment, GST_FORMAT_UNDEFINED); - gst_collect_pads_start (self->collect); + gst_collect_pads2_start (self->collect); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; @@ -632,11 +632,11 @@ gst_interleave_change_state (GstElement * element, GstStateChange transition) } /* Stop before calling the parent's state change function as - * GstCollectPads might take locks and we would deadlock in that + * GstCollectPads2 might take locks and we would deadlock in that * case */ if (transition == GST_STATE_CHANGE_PAUSED_TO_READY) - gst_collect_pads_stop (self->collect); + gst_collect_pads2_stop (self->collect); ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); @@ -876,7 +876,7 @@ gst_interleave_sink_event (GstPad * pad, GstEvent * event) break; } - /* now GstCollectPads can take care of the rest, e.g. EOS */ + /* now GstCollectPads2 can take care of the rest, e.g. EOS */ ret = self->collect_event (pad, event); gst_object_unref (self); @@ -1157,7 +1157,7 @@ gst_interleave_src_event (GstPad * pad, GstEvent * event) /* check if we are flushing */ if (flags & GST_SEEK_FLAG_FLUSH) { /* make sure we accept nothing anymore and return WRONG_STATE */ - gst_collect_pads_set_flushing (self->collect, TRUE); + gst_collect_pads2_set_flushing (self->collect, TRUE); /* flushing seek, start flush downstream, the flush will be done * when all pads received a FLUSH_STOP. */ @@ -1192,7 +1192,7 @@ gst_interleave_src_event (GstPad * pad, GstEvent * event) } static GstFlowReturn -gst_interleave_collected (GstCollectPads * pads, GstInterleave * self) +gst_interleave_collected (GstCollectPads2 * pads, GstInterleave * self) { guint size; GstBuffer *outbuf; @@ -1208,7 +1208,7 @@ gst_interleave_collected (GstCollectPads * pads, GstInterleave * self) g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED); g_return_val_if_fail (self->rate > 0, GST_FLOW_NOT_NEGOTIATED); - size = gst_collect_pads_available (pads); + size = gst_collect_pads2_available (pads); g_return_val_if_fail (size % width == 0, GST_FLOW_ERROR); @@ -1235,13 +1235,13 @@ gst_interleave_collected (GstCollectPads * pads, GstInterleave * self) memset (GST_BUFFER_DATA (outbuf), 0, size * self->channels); for (collected = pads->data; collected != NULL; collected = collected->next) { - GstCollectData *cdata; + GstCollectData2 *cdata; GstBuffer *inbuf; guint8 *outdata; - cdata = (GstCollectData *) collected->data; + cdata = (GstCollectData2 *) collected->data; - inbuf = gst_collect_pads_take_buffer (pads, cdata, size); + inbuf = gst_collect_pads2_take_buffer (pads, cdata, size); if (inbuf == NULL) { GST_DEBUG_OBJECT (cdata->pad, "No buffer available"); goto next; diff --git a/gst/interleave/interleave.h b/gst/interleave/interleave.h index fb3b27415d..abe3439e4b 100644 --- a/gst/interleave/interleave.h +++ b/gst/interleave/interleave.h @@ -27,7 +27,7 @@ #define __INTERLEAVE_H__ #include -#include +#include G_BEGIN_DECLS @@ -49,7 +49,7 @@ struct _GstInterleave GstElement element; /*< private >*/ - GstCollectPads *collect; + GstCollectPads2 *collect; gint channels; gint padcounter; From 79a6e23c1fbce3a9a04399ead08176bdc82e5763 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 14 Dec 2011 19:02:23 +0000 Subject: [PATCH 31/65] qtmux: port to GstCollectPads2 --- gst/isomp4/gstqtmux.c | 41 +++++++++++++++++++++-------------------- gst/isomp4/gstqtmux.h | 6 +++--- 2 files changed, 24 insertions(+), 23 deletions(-) diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index ed32e4954f..b83e853c51 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -226,7 +226,7 @@ static void gst_qt_mux_release_pad (GstElement * element, GstPad * pad); /* event */ static gboolean gst_qt_mux_sink_event (GstPad * pad, GstEvent * event); -static GstFlowReturn gst_qt_mux_collected (GstCollectPads * pads, +static GstFlowReturn gst_qt_mux_collected (GstCollectPads2 * pads, gpointer user_data); static GstFlowReturn gst_qt_mux_add_buffer (GstQTMux * qtmux, GstQTPad * pad, GstBuffer * buf); @@ -482,9 +482,10 @@ gst_qt_mux_init (GstQTMux * qtmux, GstQTMuxClass * qtmux_klass) gst_element_add_pad (GST_ELEMENT (qtmux), qtmux->srcpad); qtmux->sinkpads = NULL; - qtmux->collect = gst_collect_pads_new (); - gst_collect_pads_set_function (qtmux->collect, - (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_qt_mux_collected), qtmux); + qtmux->collect = gst_collect_pads2_new (); + gst_collect_pads2_set_function (qtmux->collect, + (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_qt_mux_collected), + qtmux); /* properties set to default upon construction */ @@ -1642,7 +1643,7 @@ gst_qt_mux_start_file (GstQTMux * qtmux) gst_buffer_unref (prefix); for (walk = qtmux->sinkpads; walk && !fail; walk = g_slist_next (walk)) { - GstCollectData *cdata = (GstCollectData *) walk->data; + GstCollectData2 *cdata = (GstCollectData2 *) walk->data; GstQTPad *qpad = (GstQTPad *) cdata; /* write info for each stream */ fail = atoms_recov_write_trak_info (qtmux->moov_recov_file, qpad->trak); @@ -1742,7 +1743,7 @@ gst_qt_mux_stop_file (GstQTMux * qtmux) /* pushing last buffers for each pad */ for (walk = qtmux->collect->data; walk; walk = g_slist_next (walk)) { - GstCollectData *cdata = (GstCollectData *) walk->data; + GstCollectData2 *cdata = (GstCollectData2 *) walk->data; GstQTPad *qtpad = (GstQTPad *) cdata; /* avoid add_buffer complaining if not negotiated @@ -1835,7 +1836,7 @@ gst_qt_mux_stop_file (GstQTMux * qtmux) /* check for late streams */ first_ts = GST_CLOCK_TIME_NONE; for (walk = qtmux->collect->data; walk; walk = g_slist_next (walk)) { - GstCollectData *cdata = (GstCollectData *) walk->data; + GstCollectData2 *cdata = (GstCollectData2 *) walk->data; GstQTPad *qtpad = (GstQTPad *) cdata; if (!GST_CLOCK_TIME_IS_VALID (first_ts) || @@ -1848,7 +1849,7 @@ gst_qt_mux_stop_file (GstQTMux * qtmux) GST_TIME_ARGS (first_ts)); /* add EDTSs for late streams */ for (walk = qtmux->collect->data; walk; walk = g_slist_next (walk)) { - GstCollectData *cdata = (GstCollectData *) walk->data; + GstCollectData2 *cdata = (GstCollectData2 *) walk->data; GstQTPad *qtpad = (GstQTPad *) cdata; guint32 lateness; guint32 duration; @@ -2501,7 +2502,7 @@ not_negotiated: } static GstFlowReturn -gst_qt_mux_collected (GstCollectPads * pads, gpointer user_data) +gst_qt_mux_collected (GstCollectPads2 * pads, gpointer user_data) { GstFlowReturn ret = GST_FLOW_OK; GstQTMux *qtmux = GST_QT_MUX_CAST (user_data); @@ -2524,14 +2525,14 @@ gst_qt_mux_collected (GstCollectPads * pads, gpointer user_data) walk = qtmux->collect->data; while (walk) { GstQTPad *pad; - GstCollectData *data; + GstCollectData2 *data; - data = (GstCollectData *) walk->data; + data = (GstCollectData2 *) walk->data; pad = (GstQTPad *) data; walk = g_slist_next (walk); - buf = gst_collect_pads_peek (pads, data); + buf = gst_collect_pads2_peek (pads, data); if (buf == NULL) { GST_LOG_OBJECT (qtmux, "Pad %s has no buffers", GST_PAD_NAME (pad->collect.pad)); @@ -2547,7 +2548,7 @@ gst_qt_mux_collected (GstCollectPads * pads, gpointer user_data) if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) { GST_DEBUG_OBJECT (qtmux, "clipping buffer on pad %s outside segment", GST_PAD_NAME (data->pad)); - buf = gst_collect_pads_pop (pads, data); + buf = gst_collect_pads2_pop (pads, data); gst_buffer_unref (buf); return GST_FLOW_OK; } @@ -2563,7 +2564,7 @@ gst_qt_mux_collected (GstCollectPads * pads, gpointer user_data) if (best_pad != NULL) { GST_LOG_OBJECT (qtmux, "selected pad %s with time %" GST_TIME_FORMAT, GST_PAD_NAME (best_pad->collect.pad), GST_TIME_ARGS (best_time)); - buf = gst_collect_pads_pop (pads, &best_pad->collect); + buf = gst_collect_pads2_pop (pads, &best_pad->collect); buf = gst_buffer_make_metadata_writable (buf); GST_BUFFER_TIMESTAMP (buf) = best_time; ret = gst_qt_mux_add_buffer (qtmux, best_pad, buf); @@ -3322,7 +3323,7 @@ gst_qt_mux_release_pad (GstElement * element, GstPad * pad) } } - gst_collect_pads_remove_pad (mux->collect, pad); + gst_collect_pads2_remove_pad (mux->collect, pad); } static GstPad * @@ -3357,8 +3358,8 @@ gst_qt_mux_request_new_pad (GstElement * element, newpad = gst_pad_new_from_template (templ, name); g_free (name); collect_pad = (GstQTPad *) - gst_collect_pads_add_pad_full (qtmux->collect, newpad, sizeof (GstQTPad), - (GstCollectDataDestroyNotify) (gst_qt_mux_pad_reset)); + gst_collect_pads2_add_pad_full (qtmux->collect, newpad, sizeof (GstQTPad), + (GstCollectData2DestroyNotify) (gst_qt_mux_pad_reset), TRUE); /* set up pad */ gst_qt_mux_pad_reset (collect_pad); collect_pad->trak = atom_trak_new (qtmux->context); @@ -3375,7 +3376,7 @@ gst_qt_mux_request_new_pad (GstElement * element, GST_DEBUG_FUNCPTR (gst_qt_mux_video_sink_set_caps)); /* FIXME: hacked way to override/extend the event function of - * GstCollectPads; because it sets its own event function giving the + * GstCollectPads2; because it sets its own event function giving the * element no access to events. */ qtmux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad); @@ -3518,13 +3519,13 @@ gst_qt_mux_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_collect_pads_start (qtmux->collect); + gst_collect_pads2_start (qtmux->collect); qtmux->state = GST_QT_MUX_STATE_STARTED; break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_collect_pads_stop (qtmux->collect); + gst_collect_pads2_stop (qtmux->collect); break; default: break; diff --git a/gst/isomp4/gstqtmux.h b/gst/isomp4/gstqtmux.h index 1851973e56..0295d4d1d9 100644 --- a/gst/isomp4/gstqtmux.h +++ b/gst/isomp4/gstqtmux.h @@ -44,7 +44,7 @@ #define __GST_QT_MUX_H__ #include -#include +#include #include "fourcc.h" #include "atoms.h" @@ -82,7 +82,7 @@ typedef GstBuffer * (*GstQTPadPrepareBufferFunc) (GstQTPad * pad, struct _GstQTPad { - GstCollectData collect; /* we extend the CollectData */ + GstCollectData2 collect; /* we extend the CollectData2 */ /* fourcc id of stream */ guint32 fourcc; @@ -144,7 +144,7 @@ struct _GstQTMux GstElement element; GstPad *srcpad; - GstCollectPads *collect; + GstCollectPads2 *collect; GSList *sinkpads; /* state */ From c422a4e2e68a8cb7a89501adaca3f4d816fab819 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 14 Dec 2011 19:07:23 +0000 Subject: [PATCH 32/65] matroskamux: port to GstCollectPads2 --- gst/matroska/matroska-mux.c | 36 ++++++++++++++++++------------------ gst/matroska/matroska-mux.h | 6 +++--- 2 files changed, 21 insertions(+), 21 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index 30c0cdbd4c..841e3aec69 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -223,7 +223,7 @@ static void gst_matroska_mux_finalize (GObject * object); /* Pads collected callback */ static GstFlowReturn -gst_matroska_mux_collected (GstCollectPads * pads, gpointer user_data); +gst_matroska_mux_collected (GstCollectPads2 * pads, gpointer user_data); /* pad functions */ static gboolean gst_matroska_mux_handle_src_event (GstPad * pad, @@ -450,9 +450,9 @@ gst_matroska_mux_init (GstMatroskaMux * mux, GstMatroskaMuxClass * g_class) gst_pad_set_event_function (mux->srcpad, gst_matroska_mux_handle_src_event); gst_element_add_pad (GST_ELEMENT (mux), mux->srcpad); - mux->collect = gst_collect_pads_new (); - gst_collect_pads_set_function (mux->collect, - (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_matroska_mux_collected), + mux->collect = gst_collect_pads2_new (); + gst_collect_pads2_set_function (mux->collect, + (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_matroska_mux_collected), mux); mux->ebml_write = gst_ebml_write_new (mux->srcpad); @@ -781,7 +781,7 @@ gst_matroska_mux_handle_sink_event (GstPad * pad, GstEvent * event) break; } - /* now GstCollectPads can take care of the rest, e.g. EOS */ + /* now GstCollectPads2 can take care of the rest, e.g. EOS */ if (event) ret = mux->collect_event (pad, event); @@ -2008,17 +2008,17 @@ gst_matroska_mux_request_new_pad (GstElement * element, gst_matroskamux_pad_init (newpad); collect_pad = (GstMatroskaPad *) - gst_collect_pads_add_pad_full (mux->collect, GST_PAD (newpad), + gst_collect_pads2_add_pad_full (mux->collect, GST_PAD (newpad), sizeof (GstMatroskamuxPad), - (GstCollectDataDestroyNotify) gst_matroska_pad_free); + (GstCollectData2DestroyNotify) gst_matroska_pad_free, TRUE); collect_pad->track = context; gst_matroska_pad_reset (collect_pad, FALSE); /* FIXME: hacked way to override/extend the event function of - * GstCollectPads; because it sets its own event function giving the + * GstCollectPads2; because it sets its own event function giving the * element no access to events. - * TODO GstCollectPads should really give its 'users' a clean chance to + * TODO GstCollectPads2 should really give its 'users' a clean chance to * properly handle events that are not meant for collectpads itself. * Perhaps a callback or so, though rejected (?) in #340060. * This would allow (clean) transcoding of info from demuxer/streams @@ -2063,7 +2063,7 @@ gst_matroska_mux_release_pad (GstElement * element, GstPad * pad) mux = GST_MATROSKA_MUX (GST_PAD_PARENT (pad)); for (walk = mux->collect->data; walk; walk = g_slist_next (walk)) { - GstCollectData *cdata = (GstCollectData *) walk->data; + GstCollectData2 *cdata = (GstCollectData2 *) walk->data; GstMatroskaPad *collect_pad = (GstMatroskaPad *) cdata; if (cdata->pad == pad) { @@ -2084,7 +2084,7 @@ gst_matroska_mux_release_pad (GstElement * element, GstPad * pad) } } - gst_collect_pads_remove_pad (mux->collect, pad); + gst_collect_pads2_remove_pad (mux->collect, pad); if (gst_element_remove_pad (element, pad)) mux->num_streams--; } @@ -2564,7 +2564,7 @@ gst_matroska_mux_finish (GstMatroskaMux * mux) /** * gst_matroska_mux_best_pad: * @mux: #GstMatroskaMux - * @popped: True if at least one buffer was popped from #GstCollectPads + * @popped: True if at least one buffer was popped from #GstCollectPads2 * * Find a pad with the oldest data * (data from this pad should be written first). @@ -2585,8 +2585,8 @@ gst_matroska_mux_best_pad (GstMatroskaMux * mux, gboolean * popped) collect_pad = (GstMatroskaPad *) collected->data; /* fetch a new buffer if needed */ if (collect_pad->buffer == NULL) { - collect_pad->buffer = gst_collect_pads_pop (mux->collect, - (GstCollectData *) collect_pad); + collect_pad->buffer = gst_collect_pads2_pop (mux->collect, + (GstCollectData2 *) collect_pad); if (collect_pad->buffer != NULL) { GstClockTime time; @@ -2975,7 +2975,7 @@ gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad) /** * gst_matroska_mux_collected: - * @pads: #GstCollectPads + * @pads: #GstCollectPads2 * @uuser_data: #GstMatroskaMux * * Collectpads callback. @@ -2983,7 +2983,7 @@ gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad) * Returns: #GstFlowReturn */ static GstFlowReturn -gst_matroska_mux_collected (GstCollectPads * pads, gpointer user_data) +gst_matroska_mux_collected (GstCollectPads2 * pads, gpointer user_data) { GstMatroskaMux *mux = GST_MATROSKA_MUX (user_data); GstEbmlWrite *ebml = mux->ebml_write; @@ -3077,12 +3077,12 @@ gst_matroska_mux_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: - gst_collect_pads_start (mux->collect); + gst_collect_pads2_start (mux->collect); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_collect_pads_stop (mux->collect); + gst_collect_pads2_stop (mux->collect); break; default: break; diff --git a/gst/matroska/matroska-mux.h b/gst/matroska/matroska-mux.h index 73bdc09ac3..584e5e42d0 100644 --- a/gst/matroska/matroska-mux.h +++ b/gst/matroska/matroska-mux.h @@ -24,7 +24,7 @@ #define __GST_MATROSKA_MUX_H__ #include -#include +#include #include "ebml-write.h" #include "matroska-ids.h" @@ -56,7 +56,7 @@ typedef struct _GstMatroskaMetaSeekIndex { /* all information needed for one matroska stream */ typedef struct { - GstCollectData collect; /* we extend the CollectData */ + GstCollectData2 collect; /* we extend the CollectData */ GstMatroskaTrackContext *track; GstBuffer *buffer; /* the queued buffer for this pad */ @@ -76,7 +76,7 @@ typedef struct _GstMatroskaMux { /* pads */ GstPad *srcpad; - GstCollectPads *collect; + GstCollectPads2 *collect; GstPadEventFunction collect_event; GstEbmlWrite *ebml_write; From 738ee836f3f44d9c9bdd9612813967ff22f22076 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 14 Dec 2011 19:10:53 +0000 Subject: [PATCH 33/65] multipartmux: port to GstCollectPads2 --- gst/multipart/multipartmux.c | 20 ++++++++++---------- gst/multipart/multipartmux.h | 6 +++--- 2 files changed, 13 insertions(+), 13 deletions(-) diff --git a/gst/multipart/multipartmux.c b/gst/multipart/multipartmux.c index 9d3e56a1c6..0417b7688e 100644 --- a/gst/multipart/multipartmux.c +++ b/gst/multipart/multipartmux.c @@ -88,7 +88,7 @@ static GstPad *gst_multipart_mux_request_new_pad (GstElement * element, static GstStateChangeReturn gst_multipart_mux_change_state (GstElement * element, GstStateChange transition); -static GstFlowReturn gst_multipart_mux_collected (GstCollectPads * pads, +static GstFlowReturn gst_multipart_mux_collected (GstCollectPads2 * pads, GstMultipartMux * mux); static void gst_multipart_mux_set_property (GObject * object, guint prop_id, @@ -180,9 +180,9 @@ gst_multipart_mux_init (GstMultipartMux * multipart_mux) multipart_mux->boundary = g_strdup (DEFAULT_BOUNDARY); - multipart_mux->collect = gst_collect_pads_new (); - gst_collect_pads_set_function (multipart_mux->collect, - (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_multipart_mux_collected), + multipart_mux->collect = gst_collect_pads2_new (); + gst_collect_pads2_set_function (multipart_mux->collect, + (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_multipart_mux_collected), multipart_mux); } @@ -226,7 +226,7 @@ gst_multipart_mux_request_new_pad (GstElement * element, GstMultipartPadData *multipartpad; multipartpad = (GstMultipartPadData *) - gst_collect_pads_add_pad (multipart_mux->collect, newpad, + gst_collect_pads2_add_pad (multipart_mux->collect, newpad, sizeof (GstMultipartPadData)); /* save a pointer to our data in the pad */ @@ -375,7 +375,7 @@ gst_multipart_mux_queue_pads (GstMultipartMux * mux) /* try to make sure we have a buffer from each usable pad first */ walk = mux->collect->data; while (walk) { - GstCollectData *data = (GstCollectData *) walk->data; + GstCollectData2 *data = (GstCollectData2 *) walk->data; GstMultipartPadData *pad = (GstMultipartPadData *) data; walk = g_slist_next (walk); @@ -384,7 +384,7 @@ gst_multipart_mux_queue_pads (GstMultipartMux * mux) if (pad->buffer == NULL) { GstBuffer *buf = NULL; - buf = gst_collect_pads_pop (mux->collect, data); + buf = gst_collect_pads2_pop (mux->collect, data); /* Store timestamp with segment_start and preroll */ if (buf && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { @@ -418,7 +418,7 @@ gst_multipart_mux_queue_pads (GstMultipartMux * mux) * 3) push both buffers on best pad, go to 1 */ static GstFlowReturn -gst_multipart_mux_collected (GstCollectPads * pads, GstMultipartMux * mux) +gst_multipart_mux_collected (GstCollectPads2 * pads, GstMultipartMux * mux) { GstMultipartPadData *best; GstFlowReturn ret = GST_FLOW_OK; @@ -648,11 +648,11 @@ gst_multipart_mux_change_state (GstElement * element, GstStateChange transition) multipart_mux->negotiated = FALSE; multipart_mux->need_segment = TRUE; GST_DEBUG_OBJECT (multipart_mux, "starting collect pads"); - gst_collect_pads_start (multipart_mux->collect); + gst_collect_pads2_start (multipart_mux->collect); break; case GST_STATE_CHANGE_PAUSED_TO_READY: GST_DEBUG_OBJECT (multipart_mux, "stopping collect pads"); - gst_collect_pads_stop (multipart_mux->collect); + gst_collect_pads2_stop (multipart_mux->collect); break; default: break; diff --git a/gst/multipart/multipartmux.h b/gst/multipart/multipartmux.h index a52db36ee3..1ea2d8ad6f 100644 --- a/gst/multipart/multipartmux.h +++ b/gst/multipart/multipartmux.h @@ -23,7 +23,7 @@ #define __GST_MULTIPART_MUX__ #include -#include +#include #include @@ -42,7 +42,7 @@ typedef struct _GstMultipartMuxClass GstMultipartMuxClass; /* all information needed for one multipart stream */ typedef struct { - GstCollectData collect; /* we extend the CollectData */ + GstCollectData2 collect; /* we extend the CollectData2 */ GstBuffer *buffer; /* the queued buffer for this pad */ GstClockTime timestamp; /* its timestamp, converted to running_time so that we can @@ -63,7 +63,7 @@ struct _GstMultipartMux GstPad *srcpad; /* sinkpads */ - GstCollectPads *collect; + GstCollectPads2 *collect; gint numpads; From 994dbb4c39d7b9abe0a8f2ea1b04c7af32bb1ad8 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 14 Dec 2011 19:13:21 +0000 Subject: [PATCH 34/65] smpte: port to GstCollectPads2 --- gst/smpte/gstsmpte.c | 32 ++++++++++++++++---------------- gst/smpte/gstsmpte.h | 4 ++-- 2 files changed, 18 insertions(+), 18 deletions(-) diff --git a/gst/smpte/gstsmpte.c b/gst/smpte/gstsmpte.c index 4b21cd002a..0579844405 100644 --- a/gst/smpte/gstsmpte.c +++ b/gst/smpte/gstsmpte.c @@ -152,7 +152,7 @@ static void gst_smpte_base_init (GstSMPTEClass * klass); static void gst_smpte_init (GstSMPTE * smpte); static void gst_smpte_finalize (GstSMPTE * smpte); -static GstFlowReturn gst_smpte_collected (GstCollectPads * pads, +static GstFlowReturn gst_smpte_collected (GstCollectPads2 * pads, GstSMPTE * smpte); static void gst_smpte_set_property (GObject * object, guint prop_id, @@ -360,15 +360,15 @@ gst_smpte_init (GstSMPTE * smpte) gst_pad_new_from_static_template (&gst_smpte_src_template, "src"); gst_element_add_pad (GST_ELEMENT (smpte), smpte->srcpad); - smpte->collect = gst_collect_pads_new (); - gst_collect_pads_set_function (smpte->collect, - (GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_smpte_collected), smpte); - gst_collect_pads_start (smpte->collect); + smpte->collect = gst_collect_pads2_new (); + gst_collect_pads2_set_function (smpte->collect, + (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_smpte_collected), smpte); + gst_collect_pads2_start (smpte->collect); - gst_collect_pads_add_pad (smpte->collect, smpte->sinkpad1, - sizeof (GstCollectData)); - gst_collect_pads_add_pad (smpte->collect, smpte->sinkpad2, - sizeof (GstCollectData)); + gst_collect_pads2_add_pad (smpte->collect, smpte->sinkpad1, + sizeof (GstCollectData2)); + gst_collect_pads2_add_pad (smpte->collect, smpte->sinkpad2, + sizeof (GstCollectData2)); smpte->fps = DEFAULT_PROP_FPS; smpte->type = DEFAULT_PROP_TYPE; @@ -441,7 +441,7 @@ gst_smpte_blend_i420 (guint8 * in1, guint8 * in2, guint8 * out, GstMask * mask, } static GstFlowReturn -gst_smpte_collected (GstCollectPads * pads, GstSMPTE * smpte) +gst_smpte_collected (GstCollectPads2 * pads, GstSMPTE * smpte) { GstBuffer *outbuf; GstClockTime ts; @@ -458,14 +458,14 @@ gst_smpte_collected (GstCollectPads * pads, GstSMPTE * smpte) smpte->fps_denom, smpte->fps_num); for (collected = pads->data; collected; collected = g_slist_next (collected)) { - GstCollectData *data; + GstCollectData2 *data; - data = (GstCollectData *) collected->data; + data = (GstCollectData2 *) collected->data; if (data->pad == smpte->sinkpad1) - in1 = gst_collect_pads_pop (pads, data); + in1 = gst_collect_pads2_pop (pads, data); else if (data->pad == smpte->sinkpad2) - in2 = gst_collect_pads_pop (pads, data); + in2 = gst_collect_pads2_pop (pads, data); } if (in1 == NULL) { @@ -623,11 +623,11 @@ gst_smpte_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_READY_TO_PAUSED: gst_smpte_reset (smpte); GST_LOG_OBJECT (smpte, "starting collectpads"); - gst_collect_pads_start (smpte->collect); + gst_collect_pads2_start (smpte->collect); break; case GST_STATE_CHANGE_PAUSED_TO_READY: GST_LOG_OBJECT (smpte, "stopping collectpads"); - gst_collect_pads_stop (smpte->collect); + gst_collect_pads2_stop (smpte->collect); break; default: break; diff --git a/gst/smpte/gstsmpte.h b/gst/smpte/gstsmpte.h index dc6bc6d8aa..ac138f4e76 100644 --- a/gst/smpte/gstsmpte.h +++ b/gst/smpte/gstsmpte.h @@ -22,7 +22,7 @@ #define __GST_SMPTE_H__ #include -#include +#include G_BEGIN_DECLS @@ -49,7 +49,7 @@ struct _GstSMPTE { GstPad *srcpad, *sinkpad1, *sinkpad2; - GstCollectPads *collect; + GstCollectPads2 *collect; /* properties */ gint type; From 7480b4407e3a987ef7c2fec0ae87f7de832d984e Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 15 Dec 2011 16:15:22 +0100 Subject: [PATCH 35/65] qtmux: use GstCollectPads2 event callback ... in stead of local HACK. --- gst/isomp4/gstqtmux.c | 27 +++++++++++---------------- gst/isomp4/gstqtmux.h | 3 --- 2 files changed, 11 insertions(+), 19 deletions(-) diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index b83e853c51..5c4de2e50a 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -224,7 +224,8 @@ static GstPad *gst_qt_mux_request_new_pad (GstElement * element, static void gst_qt_mux_release_pad (GstElement * element, GstPad * pad); /* event */ -static gboolean gst_qt_mux_sink_event (GstPad * pad, GstEvent * event); +static gboolean gst_qt_mux_sink_event (GstCollectPads2 * pads, + GstCollectData2 * data, GstEvent * event, gpointer user_data); static GstFlowReturn gst_qt_mux_collected (GstCollectPads2 * pads, gpointer user_data); @@ -486,6 +487,9 @@ gst_qt_mux_init (GstQTMux * qtmux, GstQTMuxClass * qtmux_klass) gst_collect_pads2_set_function (qtmux->collect, (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_qt_mux_collected), qtmux); + gst_collect_pads2_set_event_function (qtmux->collect, + (GstCollectPads2EventFunction) GST_DEBUG_FUNCPTR (gst_qt_mux_sink_event), + qtmux); /* properties set to default upon construction */ @@ -3258,13 +3262,14 @@ refuse_renegotiation: } static gboolean -gst_qt_mux_sink_event (GstPad * pad, GstEvent * event) +gst_qt_mux_sink_event (GstCollectPads2 * pads, GstCollectData2 * data, + GstEvent * event, gpointer user_data) { - gboolean ret; GstQTMux *qtmux; guint32 avg_bitrate = 0, max_bitrate = 0; + GstPad *pad = data->pad; - qtmux = GST_QT_MUX_CAST (gst_pad_get_parent (pad)); + qtmux = GST_QT_MUX_CAST (user_data); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_TAG:{ GstTagList *list; @@ -3298,10 +3303,8 @@ gst_qt_mux_sink_event (GstPad * pad, GstEvent * event) break; } - ret = qtmux->collect_event (pad, event); - gst_object_unref (qtmux); - - return ret; + /* now GstCollectPads2 can take care of the rest, e.g. EOS */ + return FALSE; } static void @@ -3375,14 +3378,6 @@ gst_qt_mux_request_new_pad (GstElement * element, gst_pad_set_setcaps_function (newpad, GST_DEBUG_FUNCPTR (gst_qt_mux_video_sink_set_caps)); - /* FIXME: hacked way to override/extend the event function of - * GstCollectPads2; because it sets its own event function giving the - * element no access to events. - */ - qtmux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad); - gst_pad_set_event_function (newpad, - GST_DEBUG_FUNCPTR (gst_qt_mux_sink_event)); - gst_pad_set_active (newpad, TRUE); gst_element_add_pad (element, newpad); diff --git a/gst/isomp4/gstqtmux.h b/gst/isomp4/gstqtmux.h index 0295d4d1d9..b23982b5bd 100644 --- a/gst/isomp4/gstqtmux.h +++ b/gst/isomp4/gstqtmux.h @@ -191,9 +191,6 @@ struct _GstQTMux guint32 fragment_duration; gboolean streamable; - /* for collect pads event handling function */ - GstPadEventFunction collect_event; - /* for request pad naming */ guint video_pads, audio_pads; }; From 97391f140224a52672a7af2ef72f6cdb605017d8 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 15 Dec 2011 16:16:52 +0100 Subject: [PATCH 36/65] avimux: use GstCollectPads2 event callback ... in stead of local HACK. --- gst/avi/gstavimux.c | 25 +++++++++---------------- gst/avi/gstavimux.h | 1 - 2 files changed, 9 insertions(+), 17 deletions(-) diff --git a/gst/avi/gstavimux.c b/gst/avi/gstavimux.c index 0089ef5f46..5e3521df79 100644 --- a/gst/avi/gstavimux.c +++ b/gst/avi/gstavimux.c @@ -197,7 +197,8 @@ static void gst_avi_mux_pad_reset (GstAviPad * avipad, gboolean free); static GstFlowReturn gst_avi_mux_collect_pads (GstCollectPads2 * pads, GstAviMux * avimux); -static gboolean gst_avi_mux_handle_event (GstPad * pad, GstEvent * event); +static gboolean gst_avi_mux_handle_event (GstCollectPads2 * pad, + GstCollectData2 * data, GstEvent * event, gpointer user_data); static GstPad *gst_avi_mux_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name); static void gst_avi_mux_release_pad (GstElement * element, GstPad * pad); @@ -422,6 +423,9 @@ gst_avi_mux_init (GstAviMux * avimux) gst_collect_pads2_set_function (avimux->collect, (GstCollectPads2Function) (GST_DEBUG_FUNCPTR (gst_avi_mux_collect_pads)), avimux); + gst_collect_pads2_set_event_function (avimux->collect, + (GstCollectPads2EventFunction) (GST_DEBUG_FUNCPTR + (gst_avi_mux_handle_event)), avimux); /* set to clean state */ gst_avi_mux_reset (avimux); @@ -1027,13 +1031,6 @@ gst_avi_mux_request_new_pad (GstElement * element, avipad->collect = gst_collect_pads2_add_pad (avimux->collect, newpad, sizeof (GstAviCollectData)); ((GstAviCollectData *) (avipad->collect))->avipad = avipad; - /* FIXME: hacked way to override/extend the event function of - * GstCollectPads2; because it sets its own event function giving the - * element no access to events */ - avimux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad); - gst_pad_set_event_function (newpad, - GST_DEBUG_FUNCPTR (gst_avi_mux_handle_event)); - if (!gst_element_add_pad (element, newpad)) goto pad_add_failed; @@ -1861,12 +1858,12 @@ gst_avi_mux_restart_file (GstAviMux * avimux) /* handle events (search) */ static gboolean -gst_avi_mux_handle_event (GstPad * pad, GstEvent * event) +gst_avi_mux_handle_event (GstCollectPads2 * pads, GstCollectData2 * data, + GstEvent * event, gpointer user_data) { GstAviMux *avimux; - gboolean ret; - avimux = GST_AVI_MUX (gst_pad_get_parent (pad)); + avimux = GST_AVI_MUX (user_data); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_TAG:{ @@ -1883,11 +1880,7 @@ gst_avi_mux_handle_event (GstPad * pad, GstEvent * event) } /* now GstCollectPads2 can take care of the rest, e.g. EOS */ - ret = avimux->collect_event (pad, event); - - gst_object_unref (avimux); - - return ret; + return FALSE; } /* send extra 'padding' data */ diff --git a/gst/avi/gstavimux.h b/gst/avi/gstavimux.h index 1457fce938..cb0cdb7b31 100644 --- a/gst/avi/gstavimux.h +++ b/gst/avi/gstavimux.h @@ -144,7 +144,6 @@ struct _GstAviMux { /* video restricted to 1 pad */ guint video_pads, audio_pads; GstCollectPads2 *collect; - GstPadEventFunction collect_event; /* the AVI header */ /* still some single stream video data in mux struct */ From f6b38f2c3934f9228b4416e87142c3a3cb4157eb Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 15 Dec 2011 16:30:17 +0100 Subject: [PATCH 37/65] matroskamux: use GstCollectPads2 event callback ... in stead of local HACK. --- gst/matroska/matroska-mux.c | 37 ++++++++++++++----------------------- gst/matroska/matroska-mux.h | 1 - 2 files changed, 14 insertions(+), 24 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index 841e3aec69..dbbeab50f3 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -224,6 +224,8 @@ static void gst_matroska_mux_finalize (GObject * object); /* Pads collected callback */ static GstFlowReturn gst_matroska_mux_collected (GstCollectPads2 * pads, gpointer user_data); +static gboolean gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads, + GstCollectData2 * data, GstEvent * event, gpointer user_data); /* pad functions */ static gboolean gst_matroska_mux_handle_src_event (GstPad * pad, @@ -454,6 +456,9 @@ gst_matroska_mux_init (GstMatroskaMux * mux, GstMatroskaMuxClass * g_class) gst_collect_pads2_set_function (mux->collect, (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_matroska_mux_collected), mux); + gst_collect_pads2_set_event_function (mux->collect, + (GstCollectPads2EventFunction) GST_DEBUG_FUNCPTR + (gst_matroska_mux_handle_sink_event), mux); mux->ebml_write = gst_ebml_write_new (mux->srcpad); mux->doctype = GST_MATROSKA_DOCTYPE_MATROSKA; @@ -709,15 +714,18 @@ gst_matroska_mux_handle_src_event (GstPad * pad, GstEvent * event) * Returns: #TRUE on success. */ static gboolean -gst_matroska_mux_handle_sink_event (GstPad * pad, GstEvent * event) +gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads, + GstCollectData2 * data, GstEvent * event, gpointer user_data) { GstMatroskaTrackContext *context; GstMatroskaPad *collect_pad; GstMatroskaMux *mux; + GstPad *pad; GstTagList *list; - gboolean ret = TRUE; - mux = GST_MATROSKA_MUX (gst_pad_get_parent (pad)); + mux = GST_MATROSKA_MUX (user_data); + collect_pad = (GstMatroskaPad *) data; + pad = data->pad; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_TAG:{ @@ -726,8 +734,6 @@ gst_matroska_mux_handle_sink_event (GstPad * pad, GstEvent * event) GST_DEBUG_OBJECT (mux, "received tag event"); gst_event_parse_tag (event, &list); - collect_pad = (GstMatroskaPad *) gst_pad_get_element_private (pad); - g_assert (collect_pad); context = collect_pad->track; g_assert (context); @@ -760,7 +766,6 @@ gst_matroska_mux_handle_sink_event (GstPad * pad, GstEvent * event) gst_event_parse_new_segment (event, NULL, NULL, &format, NULL, NULL, NULL); if (format != GST_FORMAT_TIME) { - ret = FALSE; gst_event_unref (event); event = NULL; } @@ -783,11 +788,9 @@ gst_matroska_mux_handle_sink_event (GstPad * pad, GstEvent * event) /* now GstCollectPads2 can take care of the rest, e.g. EOS */ if (event) - ret = mux->collect_event (pad, event); - - gst_object_unref (mux); - - return ret; + return FALSE; + else + return TRUE; } @@ -2015,18 +2018,6 @@ gst_matroska_mux_request_new_pad (GstElement * element, collect_pad->track = context; gst_matroska_pad_reset (collect_pad, FALSE); - /* FIXME: hacked way to override/extend the event function of - * GstCollectPads2; because it sets its own event function giving the - * element no access to events. - * TODO GstCollectPads2 should really give its 'users' a clean chance to - * properly handle events that are not meant for collectpads itself. - * Perhaps a callback or so, though rejected (?) in #340060. - * This would allow (clean) transcoding of info from demuxer/streams - * to another muxer */ - mux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad); - gst_pad_set_event_function (GST_PAD (newpad), - GST_DEBUG_FUNCPTR (gst_matroska_mux_handle_sink_event)); - gst_pad_set_setcaps_function (GST_PAD (newpad), setcapsfunc); gst_pad_set_active (GST_PAD (newpad), TRUE); if (!gst_element_add_pad (element, GST_PAD (newpad))) diff --git a/gst/matroska/matroska-mux.h b/gst/matroska/matroska-mux.h index 584e5e42d0..11483235bb 100644 --- a/gst/matroska/matroska-mux.h +++ b/gst/matroska/matroska-mux.h @@ -77,7 +77,6 @@ typedef struct _GstMatroskaMux { /* pads */ GstPad *srcpad; GstCollectPads2 *collect; - GstPadEventFunction collect_event; GstEbmlWrite *ebml_write; guint num_streams, From 50be1d5dae18209047663759e3df7874197454ac Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 15 Dec 2011 16:40:21 +0100 Subject: [PATCH 38/65] flvmux: use GstCollectPads2 event callback ... in stead of local HACK. --- gst/flv/gstflvmux.c | 26 ++++++++++---------------- gst/flv/gstflvmux.h | 2 -- 2 files changed, 10 insertions(+), 18 deletions(-) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index fc4cf7606e..8a2201ee46 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -100,6 +100,9 @@ GST_BOILERPLATE_FULL (GstFlvMux, gst_flv_mux, GstElement, GST_TYPE_ELEMENT, static void gst_flv_mux_finalize (GObject * object); static GstFlowReturn gst_flv_mux_collected (GstCollectPads2 * pads, gpointer user_data); +static gboolean +gst_flv_mux_handle_sink_event (GstCollectPads2 * pads, GstCollectData2 * data, + GstEvent * event, gpointer user_data); static gboolean gst_flv_mux_handle_src_event (GstPad * pad, GstEvent * event); static GstPad *gst_flv_mux_request_new_pad (GstElement * element, @@ -200,6 +203,9 @@ gst_flv_mux_init (GstFlvMux * mux, GstFlvMuxClass * g_class) mux->collect = gst_collect_pads2_new (); gst_collect_pads2_set_function (mux->collect, (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_flv_mux_collected), mux); + gst_collect_pads2_set_event_function (mux->collect, + (GstCollectPads2EventFunction) + GST_DEBUG_FUNCPTR (gst_flv_mux_handle_sink_event), mux); gst_flv_mux_reset (GST_ELEMENT (mux)); } @@ -260,10 +266,10 @@ gst_flv_mux_handle_src_event (GstPad * pad, GstEvent * event) } static gboolean -gst_flv_mux_handle_sink_event (GstPad * pad, GstEvent * event) +gst_flv_mux_handle_sink_event (GstCollectPads2 * pads, GstCollectData2 * data, + GstEvent * event, gpointer user_data) { - GstFlvMux *mux = GST_FLV_MUX (gst_pad_get_parent (pad)); - gboolean ret = TRUE; + GstFlvMux *mux = GST_FLV_MUX (user_data); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_TAG:{ @@ -281,11 +287,7 @@ gst_flv_mux_handle_sink_event (GstPad * pad, GstEvent * event) } /* now GstCollectPads2 can take care of the rest, e.g. EOS */ - if (ret) - ret = mux->collect_event (pad, event); - gst_object_unref (mux); - - return ret; + return FALSE; } static gboolean @@ -542,14 +544,6 @@ gst_flv_mux_request_new_pad (GstElement * element, cpad->video_codec_data = NULL; gst_flv_mux_reset_pad (mux, cpad, video); - /* FIXME: hacked way to override/extend the event function of - * GstCollectPads2; because it sets its own event function giving the - * element no access to events. - */ - mux->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (pad); - gst_pad_set_event_function (pad, - GST_DEBUG_FUNCPTR (gst_flv_mux_handle_sink_event)); - gst_pad_set_setcaps_function (pad, setcapsfunc); gst_pad_set_active (pad, TRUE); gst_element_add_pad (element, pad); diff --git a/gst/flv/gstflvmux.h b/gst/flv/gstflvmux.h index ad38abe9ee..b0e10ea320 100644 --- a/gst/flv/gstflvmux.h +++ b/gst/flv/gstflvmux.h @@ -68,8 +68,6 @@ typedef struct _GstFlvMux { GstCollectPads2 *collect; /* */ - GstPadEventFunction collect_event; - GstFlvMuxState state; gboolean have_audio; gboolean have_video; From 16a4c596eb39bfc845e2675385d95ff808920edc Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Wed, 7 Dec 2011 13:24:55 +0000 Subject: [PATCH 39/65] aacparse: parse LOAS variant The LOAS variant seems to have three different subvariants itself, only one of them is implemented as my two samples happen to be using that one. The sample rate is not always reported correctly, as the "main" sample rate is apparently sometimes half what it should be (both of my samples report 24000 Hz there), and there are two other parts of the subvariant with different sampling rates. One of them is parsed, but not the other, as it's located after some other large amount of variable data that needs parsing first, and there seems to be a LOT of it, which is useless for our needs here. This ends up being rather inconsequential, as ffdec_aac_latm, which is the only decoder that can decode such streams, does not need the sample rate on the caps anyway. https://bugzilla.gnome.org/show_bug.cgi?id=665394 --- gst/audioparsers/gstaacparse.c | 429 +++++++++++++++++++++++++++++++-- gst/audioparsers/gstaacparse.h | 2 + 2 files changed, 408 insertions(+), 23 deletions(-) diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c index d8c0995c17..8a0d6f606b 100644 --- a/gst/audioparsers/gstaacparse.c +++ b/gst/audioparsers/gstaacparse.c @@ -44,6 +44,7 @@ #include +#include #include "gstaacparse.h" @@ -52,7 +53,7 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/mpeg, " "framed = (boolean) true, " "mpegversion = (int) { 2, 4 }, " - "stream-format = (string) { raw, adts, adif };")); + "stream-format = (string) { raw, adts, adif, loas };")); static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, @@ -65,10 +66,21 @@ GST_DEBUG_CATEGORY_STATIC (aacparse_debug); #define ADIF_MAX_SIZE 40 /* Should be enough */ #define ADTS_MAX_SIZE 10 /* Should be enough */ +#define LOAS_MAX_SIZE 3 /* Should be enough */ #define AAC_FRAME_DURATION(parse) (GST_SECOND/parse->frames_per_sec) +static const gint loas_sample_rate_table[32] = { + 96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, + 16000, 12000, 11025, 8000, 7350, 0, 0, 0 +}; + +static const gint loas_channels_table[32] = { + 0, 1, 2, 3, 4, 5, 6, 8, + 0, 0, 0, 0, 0, 0, 0, 0 +}; + static gboolean gst_aac_parse_start (GstBaseParse * parse); static gboolean gst_aac_parse_stop (GstBaseParse * parse); @@ -192,6 +204,9 @@ gst_aac_parse_set_src_caps (GstAacParse * aacparse, GstCaps * sink_caps) case DSPAAC_HEADER_ADIF: stream_format = "adif"; break; + case DSPAAC_HEADER_LOAS: + stream_format = "loas"; + break; default: stream_format = NULL; } @@ -327,6 +342,8 @@ gst_aac_parse_check_adts_frame (GstAacParse * aacparse, const guint8 * data, const guint avail, gboolean drain, guint * framesize, guint * needed_data) { + *needed_data = 0; + if (G_UNLIKELY (avail < 2)) return FALSE; @@ -365,6 +382,294 @@ gst_aac_parse_check_adts_frame (GstAacParse * aacparse, return FALSE; } +static gboolean +gst_aac_parse_latm_get_value (GstAacParse * aacparse, GstBitReader * br, + guint32 * value) +{ + guint8 bytes, i, byte; + + *value = 0; + if (!gst_bit_reader_get_bits_uint8 (br, &bytes, 2)) + return FALSE; + for (i = 0; i < bytes; ++i) { + *value <<= 8; + if (!gst_bit_reader_get_bits_uint8 (br, &byte, 8)) + return FALSE; + *value += byte; + } + return TRUE; +} + +static gboolean +gst_aac_parse_get_audio_object_type (GstAacParse * aacparse, GstBitReader * br, + guint8 * audio_object_type) +{ + if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 5)) + return FALSE; + if (*audio_object_type == 31) { + if (!gst_bit_reader_get_bits_uint8 (br, audio_object_type, 6)) + return FALSE; + *audio_object_type += 32; + } + GST_LOG_OBJECT (aacparse, "audio object type %u", *audio_object_type); + return TRUE; +} + +static gboolean +gst_aac_parse_get_audio_sample_rate (GstAacParse * aacparse, GstBitReader * br, + gint * sample_rate) +{ + guint8 sampling_frequency_index; + if (!gst_bit_reader_get_bits_uint8 (br, &sampling_frequency_index, 4)) + return FALSE; + GST_LOG_OBJECT (aacparse, "sampling_frequency_index: %u", + sampling_frequency_index); + if (sampling_frequency_index == 0xf) { + guint32 sampling_rate; + if (!gst_bit_reader_get_bits_uint32 (br, &sampling_rate, 24)) + return FALSE; + *sample_rate = sampling_rate; + } else { + *sample_rate = loas_sample_rate_table[sampling_frequency_index]; + if (!*sample_rate) + return FALSE; + } + return TRUE; +} + +/* See table 1.13 in ISO/IEC 14496-3 */ +static gboolean +gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse, + GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits) +{ + guint8 audio_object_type, channel_configuration; + gboolean sbr = FALSE; + + if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type)) + return FALSE; + + if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate)) + return FALSE; + + if (!gst_bit_reader_get_bits_uint8 (br, &channel_configuration, 4)) + return FALSE; + GST_LOG_OBJECT (aacparse, "channel_configuration: %d", channel_configuration); + *channels = loas_channels_table[channel_configuration]; + if (!*channels) + return FALSE; + + if (audio_object_type == 5) { + GST_LOG_OBJECT (aacparse, + "Audio object type 5, so rereading sampling rate..."); + sbr = TRUE; + if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate)) + return FALSE; + } + + GST_INFO_OBJECT (aacparse, "Found LOAS config: %d Hz, %d channels", + *sample_rate, *channels); + + /* There's LOTS of stuff next, but we ignore it for now as we have + what we want (sample rate and number of channels */ + GST_DEBUG_OBJECT (aacparse, + "Need more code to parse humongous LOAS data, currently ignored"); + if (bits) + *bits = 0; + return TRUE; +} + + +static gboolean +gst_aac_parse_read_loas_config (GstAacParse * aacparse, const guint8 * data, + guint avail, gint * sample_rate, gint * channels, gint * version) +{ + GstBitReader br; + guint8 u8, v, vA; + + /* No version in the bitstream, but the spec has LOAS in the MPEG-4 section */ + if (version) + *version = 4; + + gst_bit_reader_init (&br, data, avail); + + /* skip sync word (11 bits) and size (13 bits) */ + gst_bit_reader_skip (&br, 11 + 13); + + /* First bit is "use last config" */ + if (!gst_bit_reader_get_bits_uint8 (&br, &u8, 1)) + return FALSE; + if (u8) { + GST_DEBUG_OBJECT (aacparse, "Frame uses previous config"); + if (!aacparse->sample_rate || !aacparse->channels) { + GST_WARNING_OBJECT (aacparse, "No previous config to use"); + } + *sample_rate = aacparse->sample_rate; + *channels = aacparse->channels; + return TRUE; + } + + GST_DEBUG_OBJECT (aacparse, "Frame contains new config"); + + if (!gst_bit_reader_get_bits_uint8 (&br, &v, 1)) + return FALSE; + if (v) { + if (!gst_bit_reader_get_bits_uint8 (&br, &vA, 1)) + return FALSE; + } else + vA = 0; + + GST_LOG_OBJECT (aacparse, "v %d, vA %d", v, vA); + if (vA == 0) { + guint8 same_time, subframes, num_program, prog; + if (v == 1) { + guint32 value; + if (!gst_aac_parse_latm_get_value (aacparse, &br, &value)) + return FALSE; + } + if (!gst_bit_reader_get_bits_uint8 (&br, &same_time, 1)) + return FALSE; + if (!gst_bit_reader_get_bits_uint8 (&br, &subframes, 6)) + return FALSE; + if (!gst_bit_reader_get_bits_uint8 (&br, &num_program, 4)) + return FALSE; + GST_LOG_OBJECT (aacparse, "same_time %d, subframes %d, num_program %d", + same_time, subframes, num_program); + + for (prog = 0; prog <= num_program; ++prog) { + guint8 num_layer, layer; + if (!gst_bit_reader_get_bits_uint8 (&br, &num_layer, 3)) + return FALSE; + GST_LOG_OBJECT (aacparse, "Program %d: %d layers", prog, num_layer); + + for (layer = 0; layer <= num_layer; ++layer) { + guint8 use_same_config; + if (prog == 0 && layer == 0) { + use_same_config = 0; + } else { + if (!gst_bit_reader_get_bits_uint8 (&br, &use_same_config, 1)) + return FALSE; + } + if (!use_same_config) { + if (v == 0) { + if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br, + sample_rate, channels, NULL)) + return FALSE; + } else { + guint32 bits, asc_len; + if (!gst_aac_parse_latm_get_value (aacparse, &br, &asc_len)) + return FALSE; + if (!gst_aac_parse_read_loas_audio_specific_config (aacparse, &br, + sample_rate, channels, &bits)) + return FALSE; + asc_len -= bits; + gst_bit_reader_skip (&br, asc_len); + } + } + } + } + GST_WARNING_OBJECT (aacparse, "More data ignored"); + } else { + GST_WARNING_OBJECT (aacparse, "Spec says \"TBD\"..."); + } + return TRUE; +} + +/** + * gst_aac_parse_loas_get_frame_len: + * @data: block of data containing a LOAS header. + * + * This function calculates LOAS frame length from the given header. + * + * Returns: size of the LOAS frame. + */ +static inline guint +gst_aac_parse_loas_get_frame_len (const guint8 * data) +{ + return (((data[1] & 0x1f) << 8) | data[2]) + 3; +} + + +/** + * gst_aac_parse_check_loas_frame: + * @aacparse: #GstAacParse. + * @data: Data to be checked. + * @avail: Amount of data passed. + * @framesize: If valid LOAS frame was found, this will be set to tell the + * found frame size in bytes. + * @needed_data: If frame was not found, this may be set to tell how much + * more data is needed in the next round to detect the frame + * reliably. This may happen when a frame header candidate + * is found but it cannot be guaranteed to be the header without + * peeking the following data. + * + * Check if the given data contains contains LOAS frame. The algorithm + * will examine LOAS frame header and calculate the frame size. Also, another + * consecutive LOAS frame header need to be present after the found frame. + * Otherwise the data is not considered as a valid LOAS frame. However, this + * "extra check" is omitted when EOS has been received. In this case it is + * enough when data[0] contains a valid LOAS header. + * + * This function may set the #needed_data to indicate that a possible frame + * candidate has been found, but more data (#needed_data bytes) is needed to + * be absolutely sure. When this situation occurs, FALSE will be returned. + * + * When a valid frame is detected, this function will use + * gst_base_parse_set_min_frame_size() function from #GstBaseParse class + * to set the needed bytes for next frame.This way next data chunk is already + * of correct size. + * + * LOAS can have three different formats, if I read the spec correctly. Only + * one of them is supported here, as the two samples I have use this one. + * + * Returns: TRUE if the given data contains a valid LOAS header. + */ +static gboolean +gst_aac_parse_check_loas_frame (GstAacParse * aacparse, + const guint8 * data, const guint avail, gboolean drain, + guint * framesize, guint * needed_data) +{ + *needed_data = 0; + + /* 3 byte header */ + if (G_UNLIKELY (avail < 3)) + return FALSE; + + if ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) { + *framesize = gst_aac_parse_loas_get_frame_len (data); + GST_DEBUG_OBJECT (aacparse, "Found %u byte LOAS frame", *framesize); + + /* In EOS mode this is enough. No need to examine the data further. + We also relax the check when we have sync, on the assumption that + if we're not looking at random data, we have a much higher chance + to get the correct sync, and this avoids losing two frames when + a single bit corruption happens. */ + if (drain || !GST_BASE_PARSE_LOST_SYNC (aacparse)) { + return TRUE; + } + + if (*framesize + LOAS_MAX_SIZE > avail) { + /* We have found a possible frame header candidate, but can't be + sure since we don't have enough data to check the next frame */ + GST_DEBUG ("NEED MORE DATA: we need %d, available %d", + *framesize + LOAS_MAX_SIZE, avail); + *needed_data = *framesize + LOAS_MAX_SIZE; + gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), + *framesize + LOAS_MAX_SIZE); + return FALSE; + } + + if ((data[*framesize] == 0x56) && ((data[*framesize + 1] & 0xe0) == 0xe0)) { + guint nextlen = gst_aac_parse_loas_get_frame_len (data + (*framesize)); + + GST_LOG ("LOAS frame found, len: %d bytes", *framesize); + gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), + nextlen + LOAS_MAX_SIZE); + return TRUE; + } + } + return FALSE; +} + /* caller ensure sufficient data */ static inline void gst_aac_parse_parse_adts_header (GstAacParse * aacparse, const guint8 * data, @@ -410,7 +715,7 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse, guint * framesize, gint * skipsize) { gboolean found = FALSE; - guint need_data = 0; + guint need_data_adts = 0, need_data_loas; guint i = 0; GST_DEBUG_OBJECT (aacparse, "Parsing header data"); @@ -419,12 +724,16 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse, stream */ /* Can we even parse the header? */ - if (avail < ADTS_MAX_SIZE) + if (avail < MAX (ADTS_MAX_SIZE, LOAS_MAX_SIZE)) { + GST_DEBUG_OBJECT (aacparse, "Not enough data to check"); return FALSE; + } for (i = 0; i < avail - 4; i++) { if (((data[i] == 0xff) && ((data[i + 1] & 0xf6) == 0xf0)) || + ((data[0] == 0x56) && ((data[1] & 0xe0) == 0xe0)) || strncmp ((char *) data + i, "ADIF", 4) == 0) { + GST_DEBUG_OBJECT (aacparse, "Found ADIF signature at offset %u", i); found = TRUE; if (i) { @@ -444,7 +753,7 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse, } if (gst_aac_parse_check_adts_frame (aacparse, data, avail, drain, - framesize, &need_data)) { + framesize, &need_data_adts)) { gint rate, channels; GST_INFO ("ADTS ID: %d, framesize: %d", (data[1] & 0x08) >> 3, *framesize); @@ -462,7 +771,38 @@ gst_aac_parse_detect_stream (GstAacParse * aacparse, gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE); return TRUE; - } else if (need_data) { + } + + if (gst_aac_parse_check_loas_frame (aacparse, data, avail, drain, + framesize, &need_data_loas)) { + gint rate, channels; + + GST_INFO ("LOAS, framesize: %d", *framesize); + + aacparse->header_type = DSPAAC_HEADER_LOAS; + + if (!gst_aac_parse_read_loas_config (aacparse, data, avail, &rate, + &channels, &aacparse->mpegversion)) { + GST_WARNING_OBJECT (aacparse, "Error reading LOAS config"); + return FALSE; + } + + if (rate && channels) { + gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), rate, + aacparse->frame_samples, 2, 2); + + GST_DEBUG ("LOAS: samplerate %d, channels %d, objtype %d, version %d", + rate, channels, aacparse->object_type, aacparse->mpegversion); + aacparse->sample_rate = rate; + aacparse->channels = channels; + } + + gst_base_parse_set_syncable (GST_BASE_PARSE (aacparse), TRUE); + + return TRUE; + } + + if (need_data_adts || need_data_loas) { /* This tells the parent class not to skip any data */ *skipsize = 0; return FALSE; @@ -607,6 +947,19 @@ gst_aac_parse_check_valid_frame (GstBaseParse * parse, needed_data); } + } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) { + guint needed_data = 1024; + + ret = gst_aac_parse_check_loas_frame (aacparse, data, + GST_BUFFER_SIZE (buffer), GST_BASE_PARSE_DRAINING (parse), + framesize, &needed_data); + + if (!ret) { + GST_DEBUG ("buffer didn't contain valid frame"); + gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), + needed_data); + } + } else { GST_DEBUG ("buffer didn't contain valid frame"); gst_base_parse_set_min_frame_size (GST_BASE_PARSE (aacparse), @@ -626,7 +979,7 @@ gst_aac_parse_check_valid_frame (GstBaseParse * parse, * * Also determines frame overhead. * ADTS streams have a 7 byte header in each frame. MP4 and ADIF streams don't have - * a per-frame header. + * a per-frame header. LOAS has 3 bytes. * * We're making a couple of simplifying assumptions: * @@ -651,29 +1004,59 @@ gst_aac_parse_parse_frame (GstBaseParse * parse, GstBaseParseFrame * frame) aacparse = GST_AAC_PARSE (parse); buffer = frame->buffer; - if (G_UNLIKELY (aacparse->header_type != DSPAAC_HEADER_ADTS)) - return ret; + if (aacparse->header_type == DSPAAC_HEADER_ADTS) { + /* see above */ + frame->overhead = 7; - /* see above */ - frame->overhead = 7; + gst_aac_parse_parse_adts_header (aacparse, GST_BUFFER_DATA (buffer), + &rate, &channels, NULL, NULL); + GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels); - gst_aac_parse_parse_adts_header (aacparse, GST_BUFFER_DATA (buffer), - &rate, &channels, NULL, NULL); - GST_LOG_OBJECT (aacparse, "rate: %d, chans: %d", rate, channels); + if (G_UNLIKELY (rate != aacparse->sample_rate + || channels != aacparse->channels)) { + aacparse->sample_rate = rate; + aacparse->channels = channels; - if (G_UNLIKELY (rate != aacparse->sample_rate - || channels != aacparse->channels)) { - aacparse->sample_rate = rate; - aacparse->channels = channels; + if (!gst_aac_parse_set_src_caps (aacparse, + GST_PAD_CAPS (GST_BASE_PARSE (aacparse)->sinkpad))) { + /* If linking fails, we need to return appropriate error */ + ret = GST_FLOW_NOT_LINKED; + } - if (!gst_aac_parse_set_src_caps (aacparse, - GST_PAD_CAPS (GST_BASE_PARSE (aacparse)->sinkpad))) { - /* If linking fails, we need to return appropriate error */ - ret = GST_FLOW_NOT_LINKED; + gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), + aacparse->sample_rate, aacparse->frame_samples, 2, 2); + } + } else if (aacparse->header_type == DSPAAC_HEADER_LOAS) { + gboolean setcaps = FALSE; + + /* see above */ + frame->overhead = 3; + + if (!gst_aac_parse_read_loas_config (aacparse, GST_BUFFER_DATA (buffer), + GST_BUFFER_SIZE (buffer), &rate, &channels, NULL)) { + GST_WARNING_OBJECT (aacparse, "Error reading LOAS config"); + } else if (G_UNLIKELY (rate != aacparse->sample_rate + || channels != aacparse->channels)) { + aacparse->sample_rate = rate; + aacparse->channels = channels; + setcaps = TRUE; + GST_INFO_OBJECT (aacparse, "New LOAS config: %d Hz, %d channels", rate, + channels); } - gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), - aacparse->sample_rate, aacparse->frame_samples, 2, 2); + /* We want to set caps both at start, and when rate/channels change. + Since only some LOAS frames have that info, we may receive frames + before knowing about rate/channels. */ + if (setcaps || !GST_PAD_CAPS (GST_BASE_PARSE_SRC_PAD (aacparse))) { + if (!gst_aac_parse_set_src_caps (aacparse, + GST_PAD_CAPS (GST_BASE_PARSE (aacparse)->sinkpad))) { + /* If linking fails, we need to return appropriate error */ + ret = GST_FLOW_NOT_LINKED; + } + + gst_base_parse_set_frame_rate (GST_BASE_PARSE (aacparse), + aacparse->sample_rate, aacparse->frame_samples, 2, 2); + } } return ret; diff --git a/gst/audioparsers/gstaacparse.h b/gst/audioparsers/gstaacparse.h index 1907c2e440..11f75e683b 100644 --- a/gst/audioparsers/gstaacparse.h +++ b/gst/audioparsers/gstaacparse.h @@ -45,6 +45,7 @@ G_BEGIN_DECLS * @DSPAAC_HEADER_UNKNOWN: Unknown (not recognized) header. * @DSPAAC_HEADER_ADIF: ADIF header found. * @DSPAAC_HEADER_ADTS: ADTS header found. + * @DSPAAC_HEADER_LOAS: LOAS header found. * @DSPAAC_HEADER_NONE: Raw stream, no header. * * Type header enumeration set in #header_type. @@ -54,6 +55,7 @@ typedef enum { DSPAAC_HEADER_UNKNOWN, DSPAAC_HEADER_ADIF, DSPAAC_HEADER_ADTS, + DSPAAC_HEADER_LOAS, DSPAAC_HEADER_NONE } GstAacHeaderType; From 7723d64a730af46e895dc1f0857612b1a15462ec Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 15 Dec 2011 21:45:17 +0100 Subject: [PATCH 40/65] matroskamux: use GstCollectPads2 buffer callback and running time clipper --- gst/matroska/matroska-mux.c | 102 ++++++------------------------------ 1 file changed, 16 insertions(+), 86 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index dbbeab50f3..e9e854638c 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -222,8 +222,8 @@ GST_BOILERPLATE_FULL (GstMatroskaMux, gst_matroska_mux, GstElement, static void gst_matroska_mux_finalize (GObject * object); /* Pads collected callback */ -static GstFlowReturn -gst_matroska_mux_collected (GstCollectPads2 * pads, gpointer user_data); +static GstFlowReturn gst_matroska_mux_handle_buffer (GstCollectPads2 * pads, + GstCollectData2 * data, GstBuffer * buf, gpointer user_data); static gboolean gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads, GstCollectData2 * data, GstEvent * event, gpointer user_data); @@ -453,12 +453,12 @@ gst_matroska_mux_init (GstMatroskaMux * mux, GstMatroskaMuxClass * g_class) gst_element_add_pad (GST_ELEMENT (mux), mux->srcpad); mux->collect = gst_collect_pads2_new (); - gst_collect_pads2_set_function (mux->collect, - (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_matroska_mux_collected), - mux); + gst_collect_pads2_set_clip_function (mux->collect, + GST_DEBUG_FUNCPTR (gst_collect_pads2_clip_running_time), mux); + gst_collect_pads2_set_buffer_function (mux->collect, + GST_DEBUG_FUNCPTR (gst_matroska_mux_handle_buffer), mux); gst_collect_pads2_set_event_function (mux->collect, - (GstCollectPads2EventFunction) GST_DEBUG_FUNCPTR - (gst_matroska_mux_handle_sink_event), mux); + GST_DEBUG_FUNCPTR (gst_matroska_mux_handle_sink_event), mux); mux->ebml_write = gst_ebml_write_new (mux->srcpad); mux->doctype = GST_MATROSKA_DOCTYPE_MATROSKA; @@ -2551,77 +2551,6 @@ gst_matroska_mux_finish (GstMatroskaMux * mux) gst_ebml_write_master_finish (ebml, mux->segment_pos); } - -/** - * gst_matroska_mux_best_pad: - * @mux: #GstMatroskaMux - * @popped: True if at least one buffer was popped from #GstCollectPads2 - * - * Find a pad with the oldest data - * (data from this pad should be written first). - * - * Returns: Selected pad. - */ -static GstMatroskaPad * -gst_matroska_mux_best_pad (GstMatroskaMux * mux, gboolean * popped) -{ - GSList *collected; - GstMatroskaPad *best = NULL; - - *popped = FALSE; - for (collected = mux->collect->data; collected; - collected = g_slist_next (collected)) { - GstMatroskaPad *collect_pad; - - collect_pad = (GstMatroskaPad *) collected->data; - /* fetch a new buffer if needed */ - if (collect_pad->buffer == NULL) { - collect_pad->buffer = gst_collect_pads2_pop (mux->collect, - (GstCollectData2 *) collect_pad); - - if (collect_pad->buffer != NULL) { - GstClockTime time; - - *popped = TRUE; - /* convert to running time */ - time = GST_BUFFER_TIMESTAMP (collect_pad->buffer); - /* invalid should pass */ - if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (time))) { - time = gst_segment_to_running_time (&collect_pad->collect.segment, - GST_FORMAT_TIME, time); - if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) { - GST_DEBUG_OBJECT (mux, "clipping buffer on pad %s outside segment", - GST_PAD_NAME (collect_pad->collect.pad)); - gst_buffer_unref (collect_pad->buffer); - collect_pad->buffer = NULL; - return NULL; - } else { - GST_LOG_OBJECT (mux, "buffer ts %" GST_TIME_FORMAT " -> %" - GST_TIME_FORMAT " running time", - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (collect_pad->buffer)), - GST_TIME_ARGS (time)); - collect_pad->buffer = - gst_buffer_make_metadata_writable (collect_pad->buffer); - GST_BUFFER_TIMESTAMP (collect_pad->buffer) = time; - } - } - } - } - - /* if we have a buffer check if it is better then the current best one */ - if (collect_pad->buffer != NULL) { - if (best == NULL || !GST_BUFFER_TIMESTAMP_IS_VALID (collect_pad->buffer) - || (GST_BUFFER_TIMESTAMP_IS_VALID (best->buffer) - && GST_BUFFER_TIMESTAMP (collect_pad->buffer) < - GST_BUFFER_TIMESTAMP (best->buffer))) { - best = collect_pad; - } - } - } - - return best; -} - /** * gst_matroska_mux_buffer_header: * @track: Track context. @@ -2963,9 +2892,8 @@ gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad) } } - /** - * gst_matroska_mux_collected: + * gst_matroska_mux_handle_buffer: * @pads: #GstCollectPads2 * @uuser_data: #GstMatroskaMux * @@ -2974,7 +2902,8 @@ gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad) * Returns: #GstFlowReturn */ static GstFlowReturn -gst_matroska_mux_collected (GstCollectPads2 * pads, gpointer user_data) +gst_matroska_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * data, + GstBuffer * buf, gpointer user_data) { GstMatroskaMux *mux = GST_MATROSKA_MUX (user_data); GstEbmlWrite *ebml = mux->ebml_write; @@ -2999,14 +2928,11 @@ gst_matroska_mux_collected (GstCollectPads2 * pads, gpointer user_data) } do { - /* which stream to write from? */ - best = gst_matroska_mux_best_pad (mux, &popped); + /* provided with stream to write from */ + best = (GstMatroskaPad *) data; /* if there is no best pad, we have reached EOS */ if (best == NULL) { - /* buffer popped, but none returned means it was clipped */ - if (popped) - break; GST_DEBUG_OBJECT (mux, "No best pad finishing..."); if (!mux->streamable) { gst_matroska_mux_finish (mux); @@ -3017,6 +2943,10 @@ gst_matroska_mux_collected (GstCollectPads2 * pads, gpointer user_data) ret = GST_FLOW_UNEXPECTED; break; } + + best->buffer = buf; + popped = TRUE; + GST_DEBUG_OBJECT (best->collect.pad, "best pad - buffer ts %" GST_TIME_FORMAT " dur %" GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (best->buffer)), From 0845a3718af64c57ddfbd0a1cd129adb37e9f529 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Thu, 15 Dec 2011 21:50:42 +0100 Subject: [PATCH 41/65] matroskamux: additional buffer handling cleanup --- gst/matroska/matroska-mux.c | 90 +++++++++++++++++-------------------- gst/matroska/matroska-mux.h | 2 - 2 files changed, 40 insertions(+), 52 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index e9e854638c..a6a2d917e3 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -575,12 +575,6 @@ gst_matroska_pad_reset (GstMatroskaPad * collect_pad, gboolean full) collect_pad->track = NULL; } - /* free cached buffer */ - if (collect_pad->buffer != NULL) { - gst_buffer_unref (collect_pad->buffer); - collect_pad->buffer = NULL; - } - if (!full && type != 0) { GstMatroskaTrackContext *context; @@ -608,7 +602,6 @@ gst_matroska_pad_reset (GstMatroskaPad * collect_pad, gboolean full) /* TODO: check default values for the context */ context->flags = GST_MATROSKA_TRACK_ENABLED | GST_MATROSKA_TRACK_DEFAULT; collect_pad->track = context; - collect_pad->buffer = NULL; collect_pad->duration = 0; collect_pad->start_ts = GST_CLOCK_TIME_NONE; collect_pad->end_ts = GST_CLOCK_TIME_NONE; @@ -2689,10 +2682,11 @@ gst_matroska_mux_stop_streamheader (GstMatroskaMux * mux) * Returns: Result of the gst_pad_push issued to write the data. */ static GstFlowReturn -gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad) +gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad, + GstBuffer * buf) { GstEbmlWrite *ebml = mux->ebml_write; - GstBuffer *buf, *hdr; + GstBuffer *hdr; guint64 blockgroup; gboolean write_duration; gint16 relative_timestamp; @@ -2702,8 +2696,6 @@ gst_matroska_mux_write_data (GstMatroskaMux * mux, GstMatroskaPad * collect_pad) GstMatroskamuxPad *pad; /* write data */ - buf = collect_pad->buffer; - collect_pad->buffer = NULL; pad = GST_MATROSKAMUX_PAD_CAST (collect_pad->collect.pad); /* vorbis/theora headers are retrieved from caps and put in CodecPrivate */ @@ -2908,7 +2900,6 @@ gst_matroska_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * data, GstMatroskaMux *mux = GST_MATROSKA_MUX (user_data); GstEbmlWrite *ebml = mux->ebml_write; GstMatroskaPad *best; - gboolean popped; GstFlowReturn ret = GST_FLOW_OK; GST_DEBUG_OBJECT (mux, "Collected pads"); @@ -2927,54 +2918,53 @@ gst_matroska_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * data, mux->state = GST_MATROSKA_MUX_STATE_DATA; } - do { - /* provided with stream to write from */ - best = (GstMatroskaPad *) data; + /* provided with stream to write from */ + best = (GstMatroskaPad *) data; - /* if there is no best pad, we have reached EOS */ - if (best == NULL) { - GST_DEBUG_OBJECT (mux, "No best pad finishing..."); - if (!mux->streamable) { - gst_matroska_mux_finish (mux); - } else { - GST_DEBUG_OBJECT (mux, "... but streamable, nothing to finish"); - } - gst_pad_push_event (mux->srcpad, gst_event_new_eos ()); - ret = GST_FLOW_UNEXPECTED; - break; + /* if there is no best pad, we have reached EOS */ + if (best == NULL) { + GST_DEBUG_OBJECT (mux, "No best pad finishing..."); + if (!mux->streamable) { + gst_matroska_mux_finish (mux); + } else { + GST_DEBUG_OBJECT (mux, "... but streamable, nothing to finish"); } + gst_pad_push_event (mux->srcpad, gst_event_new_eos ()); + ret = GST_FLOW_UNEXPECTED; + goto exit; + } - best->buffer = buf; - popped = TRUE; + /* if we have a best stream, should also have a buffer */ + g_assert (buf); - GST_DEBUG_OBJECT (best->collect.pad, "best pad - buffer ts %" - GST_TIME_FORMAT " dur %" GST_TIME_FORMAT, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (best->buffer)), - GST_TIME_ARGS (GST_BUFFER_DURATION (best->buffer))); + GST_DEBUG_OBJECT (best->collect.pad, "best pad - buffer ts %" + GST_TIME_FORMAT " dur %" GST_TIME_FORMAT, + GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)), + GST_TIME_ARGS (GST_BUFFER_DURATION (buf))); - /* make note of first and last encountered timestamps, so we can calculate - * the actual duration later when we send an updated header on eos */ - if (GST_BUFFER_TIMESTAMP_IS_VALID (best->buffer)) { - GstClockTime start_ts = GST_BUFFER_TIMESTAMP (best->buffer); - GstClockTime end_ts = start_ts; + /* make note of first and last encountered timestamps, so we can calculate + * the actual duration later when we send an updated header on eos */ + if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) { + GstClockTime start_ts = GST_BUFFER_TIMESTAMP (buf); + GstClockTime end_ts = start_ts; - if (GST_BUFFER_DURATION_IS_VALID (best->buffer)) - end_ts += GST_BUFFER_DURATION (best->buffer); - else if (best->track->default_duration) - end_ts += best->track->default_duration; + if (GST_BUFFER_DURATION_IS_VALID (buf)) + end_ts += GST_BUFFER_DURATION (buf); + else if (best->track->default_duration) + end_ts += best->track->default_duration; - if (!GST_CLOCK_TIME_IS_VALID (best->end_ts) || end_ts > best->end_ts) - best->end_ts = end_ts; + if (!GST_CLOCK_TIME_IS_VALID (best->end_ts) || end_ts > best->end_ts) + best->end_ts = end_ts; - if (G_UNLIKELY (best->start_ts == GST_CLOCK_TIME_NONE || - start_ts < best->start_ts)) - best->start_ts = start_ts; - } + if (G_UNLIKELY (best->start_ts == GST_CLOCK_TIME_NONE || + start_ts < best->start_ts)) + best->start_ts = start_ts; + } - /* write one buffer */ - ret = gst_matroska_mux_write_data (mux, best); - } while (ret == GST_FLOW_OK && !popped); + /* write one buffer */ + ret = gst_matroska_mux_write_data (mux, best, buf); +exit: return ret; } diff --git a/gst/matroska/matroska-mux.h b/gst/matroska/matroska-mux.h index 11483235bb..6eb584d111 100644 --- a/gst/matroska/matroska-mux.h +++ b/gst/matroska/matroska-mux.h @@ -59,8 +59,6 @@ typedef struct GstCollectData2 collect; /* we extend the CollectData */ GstMatroskaTrackContext *track; - GstBuffer *buffer; /* the queued buffer for this pad */ - guint64 duration; GstClockTime start_ts; GstClockTime end_ts; /* last timestamp + (if available) duration */ From 06267242629e2912dd2ca56efc03d95486ce6a23 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 16 Dec 2011 16:56:37 +0100 Subject: [PATCH 42/65] matroskamux: additional subtitle support --- gst/matroska/matroska-mux.c | 236 ++++++++++++++++++++++++++++++++---- 1 file changed, 210 insertions(+), 26 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index a6a2d917e3..65b31eabf7 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -2,6 +2,7 @@ * (c) 2003 Ronald Bultje * (c) 2005 Michal Benes * (c) 2008 Sebastian Dröge + * (c) 2011 Mark Nauwelaerts * * matroska-mux.c: matroska file/stream muxer * @@ -205,10 +206,14 @@ static GstStaticPadTemplate audiosink_templ = ); static GstStaticPadTemplate subtitlesink_templ = -GST_STATIC_PAD_TEMPLATE ("subtitle_%d", + GST_STATIC_PAD_TEMPLATE ("subtitle_%d", GST_PAD_SINK, GST_PAD_REQUEST, - GST_STATIC_CAPS ("subtitle/x-kate")); + GST_STATIC_CAPS ("subtitle/x-kate; " + "text/plain; application/x-ssa; application/x-ass; " + "application/x-usf; video/x-dvd-subpicture; " + "application/x-subtitle-unknown") + ); static GArray *used_uids; G_LOCK_DEFINE_STATIC (used_uids); @@ -697,6 +702,44 @@ gst_matroska_mux_handle_src_event (GstPad * pad, GstEvent * event) return gst_pad_event_default (pad, event); } +static void +gst_matroska_mux_build_vobsub_private (GstMatroskaTrackContext * context, + const guint * clut) +{ + gchar *clutv[17]; + gchar *sclut; + gint i; + guint32 col; + gdouble y, u, v; + guint8 r, g, b; + + /* produce comma-separated list in hex format */ + for (i = 0; i < 16; ++i) { + col = clut[i]; + /* replicate vobsub's slightly off RGB conversion calculation */ + y = (((col >> 16) & 0xff) - 16) * 255 / 219; + u = ((col >> 8) & 0xff) - 128; + v = (col & 0xff) - 128; + r = CLAMP (1.0 * y + 1.4022 * u, 0, 255); + g = CLAMP (1.0 * y - 0.3456 * u - 0.7145 * v, 0, 255); + b = CLAMP (1.0 * y + 1.7710 * v, 0, 255); + clutv[i] = g_strdup_printf ("%02x%02x%02x", r, g, b); + } + clutv[i] = NULL; + sclut = g_strjoinv (",", clutv); + + /* build codec private; only palette for now */ + g_free (context->codec_priv); + context->codec_priv = (guint8 *) g_strdup_printf ("palette: %s", sclut); + /* include terminating 0 */ + context->codec_priv_size = strlen ((gchar *) context->codec_priv) + 1; + g_free (sclut); + for (i = 0; i < 16; ++i) { + g_free (clutv[i]); + } +} + + /** * gst_matroska_mux_handle_sink_event: * @pad: Pad which received the event. @@ -719,6 +762,8 @@ gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads, mux = GST_MATROSKA_MUX (user_data); collect_pad = (GstMatroskaPad *) data; pad = data->pad; + context = collect_pad->track; + g_assert (context); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_TAG:{ @@ -727,9 +772,6 @@ gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads, GST_DEBUG_OBJECT (mux, "received tag event"); gst_event_parse_tag (event, &list); - context = collect_pad->track; - g_assert (context); - /* Matroska wants ISO 639-2B code, taglist most likely contains 639-1 */ if (gst_tag_list_get_string (list, GST_TAG_LANGUAGE_CODE, &lang)) { const gchar *lang_code; @@ -772,6 +814,31 @@ gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads, gst_event_replace (&mux->force_key_unit_event, NULL); mux->force_key_unit_event = event; event = NULL; + } else if (gst_structure_has_name (structure, "application/x-gst-dvd") && + !strcmp ("dvd-spu-clut-change", + gst_structure_get_string (structure, "event"))) { + gchar name[16]; + gint i, value; + guint clut[16]; + + GST_DEBUG_OBJECT (pad, "New DVD colour table received"); + if (context->type != GST_MATROSKA_TRACK_TYPE_SUBTITLE) { + GST_DEBUG_OBJECT (pad, "... discarding"); + break; + } + /* first transform event data into table form */ + for (i = 0; i < 16; i++) { + g_snprintf (name, sizeof (name), "clut%02d", i); + if (!gst_structure_get_int (structure, name, &value)) { + GST_ERROR_OBJECT (mux, "dvd-spu-clut-change event did not " + "contain %s field", name); + break; + } + clut[i] = value; + } + + /* transform into private data for stream; text form */ + gst_matroska_mux_build_vobsub_private (context, clut); } break; } @@ -1854,6 +1921,10 @@ refuse_caps: } } +/* we probably don't have the data at start, + * so have to reserve (a maximum) space to write this at the end. + * bit spacy, but some formats can hold quite some */ +#define SUBTITLE_MAX_CODEC_PRIVATE 2048 /* must be > 128 */ /** * gst_matroska_mux_subtitle_pad_setcaps: @@ -1867,11 +1938,6 @@ refuse_caps: static gboolean gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps) { - /* FIXME: - * Consider this as boilerplate code for now. There is - * no single subtitle creation element in GStreamer, - * neither do I know how subtitling works at all. */ - /* There is now (at least) one such alement (kateenc), and I'm going to handle it here and claim it works when it can be piped back through GStreamer and VLC */ @@ -1882,6 +1948,10 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps) GstMatroskaPad *collect_pad; const gchar *mimetype; GstStructure *structure; + const GValue *value = NULL; + const GstBuffer *buf = NULL; + gchar *id = NULL; + gboolean ret = TRUE; mux = GST_MATROSKA_MUX (GST_PAD_PARENT (pad)); @@ -1896,13 +1966,14 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps) structure = gst_caps_get_structure (caps, 0); mimetype = gst_structure_get_name (structure); + /* keep track of default set in request_pad */ + id = context->codec_id; + /* general setup */ scontext->check_utf8 = 1; scontext->invalid_utf8 = 0; context->default_duration = 0; - /* TODO: - other format than Kate */ - if (!strcmp (mimetype, "subtitle/x-kate")) { const GValue *streamheader; @@ -1918,12 +1989,59 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps) if (!kate_streamheader_to_codecdata (streamheader, context)) { GST_ELEMENT_ERROR (mux, STREAM, MUX, (NULL), ("kate stream headers missing or malformed")); - return FALSE; + ret = FALSE; + goto exit; } - return TRUE; + } else if (!strcmp (mimetype, "text/plain")) { + context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_UTF8); + } else if (!strcmp (mimetype, "application/x-ssa")) { + context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_SSA); + } else if (!strcmp (mimetype, "application/x-ass")) { + context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_ASS); + } else if (!strcmp (mimetype, "application/x-usf")) { + context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_USF); + } else if (!strcmp (mimetype, "video/x-dvd-subpicture")) { + context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_VOBSUB); + } else { + id = NULL; + ret = FALSE; + goto exit; } - return FALSE; + /* maybe some private data, e.g. vobsub */ + value = gst_structure_get_value (structure, "codec_data"); + if (value) + buf = gst_value_get_buffer (value); + if (buf != NULL) { + guint8 *priv_data = NULL; + guint priv_data_size = 0; + + priv_data_size = GST_BUFFER_SIZE (buf); + if (priv_data_size > SUBTITLE_MAX_CODEC_PRIVATE) { + GST_WARNING_OBJECT (mux, "pad %" GST_PTR_FORMAT " subtitle private data" + " exceeded maximum (%d); discarding", pad, + SUBTITLE_MAX_CODEC_PRIVATE); + return TRUE; + } + + if (context->codec_priv != NULL) + g_free (context->codec_priv); + + priv_data = g_malloc0 (priv_data_size); + memcpy (priv_data, GST_BUFFER_DATA (buf), priv_data_size); + context->codec_priv = priv_data; + context->codec_priv_size = priv_data_size; + } + + GST_DEBUG_OBJECT (pad, "codec_id %s, codec data size %u", + GST_STR_NULL (context->codec_id), context->codec_priv_size); + +exit: + /* free default if modified */ + if (id) + g_free (id); + + return ret; } @@ -1950,6 +2068,8 @@ gst_matroska_mux_request_new_pad (GstElement * element, GstPadSetCapsFunction setcapsfunc = NULL; GstMatroskaTrackContext *context = NULL; gint pad_id; + gboolean locked = TRUE; + gchar *id = NULL; if (templ == gst_element_class_get_pad_template (klass, "audio_%d")) { /* don't mix named and unnamed pads, if the pad already exists we fail when @@ -1993,6 +2113,9 @@ gst_matroska_mux_request_new_pad (GstElement * element, g_new0 (GstMatroskaTrackSubtitleContext, 1); context->type = GST_MATROSKA_TRACK_TYPE_SUBTITLE; context->name = g_strdup ("Subtitle"); + /* setcaps may only provide proper one a lot later */ + id = g_strdup ("S_SUB_UNKNOWN"); + locked = FALSE; } else { GST_WARNING_OBJECT (mux, "This is not our template!"); return NULL; @@ -2006,10 +2129,11 @@ gst_matroska_mux_request_new_pad (GstElement * element, collect_pad = (GstMatroskaPad *) gst_collect_pads2_add_pad_full (mux->collect, GST_PAD (newpad), sizeof (GstMatroskamuxPad), - (GstCollectData2DestroyNotify) gst_matroska_pad_free, TRUE); + (GstCollectData2DestroyNotify) gst_matroska_pad_free, locked); collect_pad->track = context; gst_matroska_pad_reset (collect_pad, FALSE); + collect_pad->track->codec_id = id; gst_pad_set_setcaps_function (GST_PAD (newpad), setcapsfunc); gst_pad_set_active (GST_PAD (newpad), TRUE); @@ -2105,6 +2229,14 @@ gst_matroska_mux_track_header (GstMatroskaMux * mux, context->language); } + /* FIXME: until we have a nice way of getting the codecname + * out of the caps, I'm not going to enable this. Too much + * (useless, double, boring) work... */ + /* TODO: Use value from tags if any */ + /*gst_ebml_write_utf8 (ebml, GST_MATROSKA_ID_CODECNAME, + context->codec_name); */ + gst_ebml_write_utf8 (ebml, GST_MATROSKA_ID_TRACKNAME, context->name); + /* type-specific stuff */ switch (context->type) { case GST_MATROSKA_TRACK_TYPE_VIDEO:{ @@ -2155,6 +2287,24 @@ gst_matroska_mux_track_header (GstMatroskaMux * mux, break; } + /* this is what we write for now and must be filled + * and remainder void'ed later on */ +#define SUBTITLE_DUMMY_SIZE (1 + 1 + 14 + 1 + 2 + SUBTITLE_MAX_CODEC_PRIVATE) + + case GST_MATROSKA_TRACK_TYPE_SUBTITLE:{ + gpointer buf; + + context->pos = ebml->pos; + /* CodecID is mandatory ... */ + gst_ebml_write_ascii (ebml, GST_MATROSKA_ID_CODECID, "S_SUB_UNKNOWN"); + /* reserve space */ + buf = g_malloc0 (SUBTITLE_MAX_CODEC_PRIVATE); + gst_ebml_write_binary (ebml, GST_EBML_ID_VOID, buf, + SUBTITLE_MAX_CODEC_PRIVATE); + g_free (buf); + /* real data has to be written at finish */ + return; + } default: /* doesn't need type-specific data */ break; @@ -2164,13 +2314,6 @@ gst_matroska_mux_track_header (GstMatroskaMux * mux, if (context->codec_priv) gst_ebml_write_binary (ebml, GST_MATROSKA_ID_CODECPRIVATE, context->codec_priv, context->codec_priv_size); - /* FIXME: until we have a nice way of getting the codecname - * out of the caps, I'm not going to enable this. Too much - * (useless, double, boring) work... */ - /* TODO: Use value from tags if any */ - /*gst_ebml_write_utf8 (ebml, GST_MATROSKA_ID_CODECNAME, - context->codec_name); */ - gst_ebml_write_utf8 (ebml, GST_MATROSKA_ID_TRACKNAME, context->name); } @@ -2490,16 +2633,23 @@ gst_matroska_mux_finish (GstMatroskaMux * mux) gst_ebml_write_seek (ebml, my_pos); } - /* update duration */ - /* first get the overall duration */ - /* a released track may have left a duration in here */ + /* loop tracks: + * - first get the overall duration + * (a released track may have left a duration in here) + * - write some track header data for subtitles + */ duration = mux->duration; + pos = ebml->pos; for (collected = mux->collect->data; collected; collected = g_slist_next (collected)) { GstMatroskaPad *collect_pad; GstClockTime min_duration; /* observed minimum duration */ + GstMatroskaTrackContext *context; + gint voidleft = 0, fill = 0; + gpointer codec_id; collect_pad = (GstMatroskaPad *) collected->data; + context = collect_pad->track; GST_DEBUG_OBJECT (mux, "Pad %" GST_PTR_FORMAT " start ts %" GST_TIME_FORMAT @@ -2521,7 +2671,41 @@ gst_matroska_mux_finish (GstMatroskaMux * mux) if (GST_CLOCK_TIME_IS_VALID (collect_pad->duration) && duration < collect_pad->duration) duration = collect_pad->duration; + + if (context->type != GST_MATROSKA_TRACK_TYPE_SUBTITLE || !context->pos) + continue; + + again: + /* write subtitle type and possible private data */ + gst_ebml_write_seek (ebml, context->pos); + /* complex way to write ascii to account for extra filling */ + codec_id = g_malloc0 (strlen (context->codec_id) + 1 + fill); + strcpy (codec_id, context->codec_id); + gst_ebml_write_binary (ebml, GST_MATROSKA_ID_CODECID, + codec_id, strlen (context->codec_id) + 1 + fill); + g_free (codec_id); + if (context->codec_priv) + gst_ebml_write_binary (ebml, GST_MATROSKA_ID_CODECPRIVATE, + context->codec_priv, context->codec_priv_size); + voidleft = SUBTITLE_DUMMY_SIZE - (ebml->pos - context->pos); + /* void'ify; sigh, variable sized length field */ + if (voidleft == 1) { + fill = 1; + goto again; + } else if (voidleft && voidleft <= 128) + gst_ebml_write_buffer_header (ebml, GST_EBML_ID_VOID, voidleft - 2); + else if (voidleft >= 130) + gst_ebml_write_buffer_header (ebml, GST_EBML_ID_VOID, voidleft - 3); + else if (voidleft == 129) { + gst_ebml_write_buffer_header (ebml, GST_EBML_ID_VOID, 64); + gst_ebml_write_buffer_header (ebml, GST_EBML_ID_VOID, 63); + } } + + /* seek back (optional, but do anyway) */ + gst_ebml_write_seek (ebml, pos); + + /* update duration */ if (duration != 0) { GST_DEBUG_OBJECT (mux, "final total duration: %" GST_TIME_FORMAT, GST_TIME_ARGS (duration)); From f06d7416333a6912f5f6661a2abe0025980bf03f Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 16 Dec 2011 17:20:51 +0100 Subject: [PATCH 43/65] matroskamux: bring a few debug statements up to specs ... and minor spelling fix. --- gst/matroska/matroska-mux.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index 65b31eabf7..f454247af6 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -2383,7 +2383,7 @@ gst_matroska_mux_start (GstMatroskaMux * mux) if (tags != NULL && !gst_tag_list_is_empty (tags)) { guint64 master_tags, master_tag; - GST_DEBUG ("Writing tags"); + GST_DEBUG_OBJECT (mux, "Writing tags"); /* TODO: maybe limit via the TARGETS id by looking at the source pad */ mux->tags_pos = ebml->pos; @@ -2458,7 +2458,7 @@ gst_matroska_mux_start (GstMatroskaMux * mux) child = gst_ebml_write_master_start (ebml, GST_MATROSKA_ID_TRACKENTRY); gst_matroska_mux_track_header (mux, collect_pad->track); gst_ebml_write_master_finish (ebml, child); - /* some remaing pad/track setup */ + /* some remaining pad/track setup */ collect_pad->default_duration_scaled = gst_util_uint64_scale (collect_pad->track->default_duration, 1, mux->time_scale); @@ -2585,7 +2585,7 @@ gst_matroska_mux_finish (GstMatroskaMux * mux) if (tags != NULL && !gst_tag_list_is_empty (tags)) { guint64 master_tags, master_tag; - GST_DEBUG ("Writing tags"); + GST_DEBUG_OBJECT (mux, "Writing tags"); /* TODO: maybe limit via the TARGETS id by looking at the source pad */ mux->tags_pos = ebml->pos; From 0c7a491a6f5b712ec013fc26c81a298495618cae Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 16 Dec 2011 18:03:01 +0100 Subject: [PATCH 44/65] qtmux: use GstCollectPads2 buffer callback and running time clipper ... since default collection heuristics suffice. --- gst/isomp4/gstqtmux.c | 67 +++++++++---------------------------------- 1 file changed, 14 insertions(+), 53 deletions(-) diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index 5c4de2e50a..5d182396e5 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -227,8 +227,8 @@ static void gst_qt_mux_release_pad (GstElement * element, GstPad * pad); static gboolean gst_qt_mux_sink_event (GstCollectPads2 * pads, GstCollectData2 * data, GstEvent * event, gpointer user_data); -static GstFlowReturn gst_qt_mux_collected (GstCollectPads2 * pads, - gpointer user_data); +static GstFlowReturn gst_qt_mux_handle_buffer (GstCollectPads2 * pads, + GstCollectData2 * cdata, GstBuffer * buf, gpointer user_data); static GstFlowReturn gst_qt_mux_add_buffer (GstQTMux * qtmux, GstQTPad * pad, GstBuffer * buf); @@ -484,12 +484,12 @@ gst_qt_mux_init (GstQTMux * qtmux, GstQTMuxClass * qtmux_klass) qtmux->sinkpads = NULL; qtmux->collect = gst_collect_pads2_new (); - gst_collect_pads2_set_function (qtmux->collect, - (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_qt_mux_collected), - qtmux); + gst_collect_pads2_set_buffer_function (qtmux->collect, + GST_DEBUG_FUNCPTR (gst_qt_mux_handle_buffer), qtmux); gst_collect_pads2_set_event_function (qtmux->collect, - (GstCollectPads2EventFunction) GST_DEBUG_FUNCPTR (gst_qt_mux_sink_event), - qtmux); + GST_DEBUG_FUNCPTR (gst_qt_mux_sink_event), qtmux); + gst_collect_pads2_set_clip_function (qtmux->collect, + GST_DEBUG_FUNCPTR (gst_collect_pads2_clip_running_time), qtmux); /* properties set to default upon construction */ @@ -2506,14 +2506,13 @@ not_negotiated: } static GstFlowReturn -gst_qt_mux_collected (GstCollectPads2 * pads, gpointer user_data) +gst_qt_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * cdata, + GstBuffer * buf, gpointer user_data) { GstFlowReturn ret = GST_FLOW_OK; GstQTMux *qtmux = GST_QT_MUX_CAST (user_data); - GSList *walk; GstQTPad *best_pad = NULL; - GstClockTime time, best_time = GST_CLOCK_TIME_NONE; - GstBuffer *buf; + GstClockTime best_time = GST_CLOCK_TIME_NONE; if (G_UNLIKELY (qtmux->state == GST_QT_MUX_STATE_STARTED)) { if ((ret = gst_qt_mux_start_file (qtmux)) != GST_FLOW_OK) @@ -2525,52 +2524,14 @@ gst_qt_mux_collected (GstCollectPads2 * pads, gpointer user_data) if (G_UNLIKELY (qtmux->state == GST_QT_MUX_STATE_EOS)) return GST_FLOW_UNEXPECTED; - /* select the best buffer */ - walk = qtmux->collect->data; - while (walk) { - GstQTPad *pad; - GstCollectData2 *data; - - data = (GstCollectData2 *) walk->data; - pad = (GstQTPad *) data; - - walk = g_slist_next (walk); - - buf = gst_collect_pads2_peek (pads, data); - if (buf == NULL) { - GST_LOG_OBJECT (qtmux, "Pad %s has no buffers", - GST_PAD_NAME (pad->collect.pad)); - continue; - } - time = GST_BUFFER_TIMESTAMP (buf); - gst_buffer_unref (buf); - - /* invalid should pass */ - if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (time))) { - time = - gst_segment_to_running_time (&data->segment, GST_FORMAT_TIME, time); - if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) { - GST_DEBUG_OBJECT (qtmux, "clipping buffer on pad %s outside segment", - GST_PAD_NAME (data->pad)); - buf = gst_collect_pads2_pop (pads, data); - gst_buffer_unref (buf); - return GST_FLOW_OK; - } - } - - if (best_pad == NULL || !GST_CLOCK_TIME_IS_VALID (time) || - (GST_CLOCK_TIME_IS_VALID (best_time) && time < best_time)) { - best_pad = pad; - best_time = time; - } - } + best_pad = (GstQTPad *) cdata; + /* clipping already converted to running time */ if (best_pad != NULL) { + g_assert (buf); + best_time = GST_BUFFER_TIMESTAMP (buf); GST_LOG_OBJECT (qtmux, "selected pad %s with time %" GST_TIME_FORMAT, GST_PAD_NAME (best_pad->collect.pad), GST_TIME_ARGS (best_time)); - buf = gst_collect_pads2_pop (pads, &best_pad->collect); - buf = gst_buffer_make_metadata_writable (buf); - GST_BUFFER_TIMESTAMP (buf) = best_time; ret = gst_qt_mux_add_buffer (qtmux, best_pad, buf); } else { ret = gst_qt_mux_stop_file (qtmux); From 8c32195d51134469b274af67ba4aee4387932e27 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 16 Dec 2011 19:15:03 +0100 Subject: [PATCH 45/65] flvmux: use GstCollectPads2 buffer callback and running time clipper ... since the default collection heuristics suffice. --- gst/flv/gstflvmux.c | 83 ++++++++++++--------------------------------- 1 file changed, 22 insertions(+), 61 deletions(-) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index 8a2201ee46..0c0aa6dcbe 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -99,7 +99,8 @@ GST_BOILERPLATE_FULL (GstFlvMux, gst_flv_mux, GstElement, GST_TYPE_ELEMENT, static void gst_flv_mux_finalize (GObject * object); static GstFlowReturn -gst_flv_mux_collected (GstCollectPads2 * pads, gpointer user_data); +gst_flv_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * cdata, + GstBuffer * buf, gpointer user_data); static gboolean gst_flv_mux_handle_sink_event (GstCollectPads2 * pads, GstCollectData2 * data, GstEvent * event, gpointer user_data); @@ -201,11 +202,12 @@ gst_flv_mux_init (GstFlvMux * mux, GstFlvMuxClass * g_class) mux->new_tags = FALSE; mux->collect = gst_collect_pads2_new (); - gst_collect_pads2_set_function (mux->collect, - (GstCollectPads2Function) GST_DEBUG_FUNCPTR (gst_flv_mux_collected), mux); + gst_collect_pads2_set_buffer_function (mux->collect, + GST_DEBUG_FUNCPTR (gst_flv_mux_handle_buffer), mux); gst_collect_pads2_set_event_function (mux->collect, - (GstCollectPads2EventFunction) GST_DEBUG_FUNCPTR (gst_flv_mux_handle_sink_event), mux); + gst_collect_pads2_set_clip_function (mux->collect, + GST_DEBUG_FUNCPTR (gst_collect_pads2_clip_running_time), mux); gst_flv_mux_reset (GST_ELEMENT (mux)); } @@ -1191,18 +1193,12 @@ gst_flv_mux_update_index (GstFlvMux * mux, GstBuffer * buffer, GstFlvPad * cpad) } static GstFlowReturn -gst_flv_mux_write_buffer (GstFlvMux * mux, GstFlvPad * cpad) +gst_flv_mux_write_buffer (GstFlvMux * mux, GstFlvPad * cpad, GstBuffer * buffer) { GstBuffer *tag; - GstBuffer *buffer = - gst_collect_pads2_pop (mux->collect, (GstCollectData2 *) cpad); GstFlowReturn ret; - /* arrange downstream running time */ - buffer = gst_buffer_make_metadata_writable (buffer); - GST_BUFFER_TIMESTAMP (buffer) = - gst_segment_to_running_time (&cpad->collect.segment, - GST_FORMAT_TIME, GST_BUFFER_TIMESTAMP (buffer)); + /* clipping function arranged for running_time */ if (!mux->streamable) gst_flv_mux_update_index (mux, buffer, cpad); @@ -1384,14 +1380,13 @@ gst_flv_mux_rewrite_header (GstFlvMux * mux) } static GstFlowReturn -gst_flv_mux_collected (GstCollectPads2 * pads, gpointer user_data) +gst_flv_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * cdata, + GstBuffer * buffer, gpointer user_data) { GstFlvMux *mux = GST_FLV_MUX (user_data); GstFlvPad *best; GstClockTime best_time; GstFlowReturn ret; - GSList *sl; - gboolean eos = TRUE; if (mux->state == GST_FLV_MUX_STATE_HEADER) { if (mux->collect->data == NULL) { @@ -1417,46 +1412,12 @@ gst_flv_mux_collected (GstCollectPads2 * pads, gpointer user_data) mux->new_tags = FALSE; } - - best = NULL; - best_time = GST_CLOCK_TIME_NONE; - for (sl = mux->collect->data; sl; sl = sl->next) { - GstFlvPad *cpad = sl->data; - GstBuffer *buffer = gst_collect_pads2_peek (pads, (GstCollectData2 *) cpad); - GstClockTime time; - - if (!buffer) - continue; - - eos = FALSE; - - time = GST_BUFFER_TIMESTAMP (buffer); - gst_buffer_unref (buffer); - - /* Use buffers without valid timestamp first */ - if (!GST_CLOCK_TIME_IS_VALID (time)) { - GST_WARNING_OBJECT (pads, "Buffer without valid timestamp"); - - best_time = cpad->last_timestamp; - best = cpad; - break; - } - - time = gst_segment_to_running_time (&cpad->collect.segment, - GST_FORMAT_TIME, time); - if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) { - GST_DEBUG_OBJECT (mux, "clipping buffer on pad %s outside segment", - GST_PAD_NAME (cpad->collect.pad)); - buffer = gst_collect_pads2_pop (pads, (GstCollectData2 *) cpad); - gst_buffer_unref (buffer); - return GST_FLOW_OK; - } - - if (best == NULL || (GST_CLOCK_TIME_IS_VALID (best_time) - && time < best_time)) { - best = cpad; - best_time = time; - } + best = (GstFlvPad *) cdata; + if (best) { + g_assert (buffer); + best_time = GST_BUFFER_TIMESTAMP (buffer); + } else { + best_time = GST_CLOCK_TIME_NONE; } /* The FLV timestamp is an int32 field. For non-live streams error out if a @@ -1465,17 +1426,17 @@ gst_flv_mux_collected (GstCollectPads2 * pads, gpointer user_data) if (!mux->streamable && GST_CLOCK_TIME_IS_VALID (best_time) && best_time / GST_MSECOND > G_MAXINT32) { GST_WARNING_OBJECT (mux, "Timestamp larger than FLV supports - EOS"); - eos = TRUE; + gst_buffer_unref (buffer); + buffer = NULL; + best = NULL; } - if (!eos && best) { - return gst_flv_mux_write_buffer (mux, best); - } else if (eos) { + if (best) { + return gst_flv_mux_write_buffer (mux, best, buffer); + } else { gst_flv_mux_rewrite_header (mux); gst_pad_push_event (mux->srcpad, gst_event_new_eos ()); return GST_FLOW_UNEXPECTED; - } else { - return GST_FLOW_OK; } } From fa5c02b211c0a9b746fb42260a41c05cde6e8675 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Mon, 19 Dec 2011 13:56:30 +0100 Subject: [PATCH 46/65] flvmux: only write full metadata at start ... rather than having (potentially) unnecessary duplicates written all over, or even contradictory varying filesize info, or duration info that will not be rewritten upon header rewrite. --- gst/flv/gstflvmux.c | 24 ++++++++++++++++++++---- 1 file changed, 20 insertions(+), 4 deletions(-) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index 0c0aa6dcbe..4a8b18ff46 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -644,7 +644,7 @@ gst_flv_mux_create_number_script_value (const gchar * name, gdouble value) } static GstBuffer * -gst_flv_mux_create_metadata (GstFlvMux * mux) +gst_flv_mux_create_metadata (GstFlvMux * mux, gboolean full) { const GstTagList *tags; GstBuffer *script_tag, *tmp; @@ -687,6 +687,9 @@ gst_flv_mux_create_metadata (GstFlvMux * mux) GST_WRITE_UINT32_BE (data + 1, n_tags); script_tag = gst_buffer_join (script_tag, tmp); + if (!full) + goto tags; + /* Some players expect the 'duration' to be always set. Fill it out later, after querying the pads or after getting EOS */ if (!mux->streamable) { @@ -707,6 +710,7 @@ gst_flv_mux_create_metadata (GstFlvMux * mux) GST_DEBUG_OBJECT (mux, "not preallocating index, streamable mode"); } +tags: for (i = 0; tags && i < n_tags; i++) { const gchar *tag_name = gst_structure_nth_field_name ((const GstStructure *) tags, i); @@ -745,6 +749,9 @@ gst_flv_mux_create_metadata (GstFlvMux * mux) } } + if (!full) + goto end; + if (mux->duration == GST_CLOCK_TIME_NONE) { GSList *l; @@ -926,6 +933,13 @@ gst_flv_mux_create_metadata (GstFlvMux * mux) tags_written++; } +end: + + if (!tags_written) { + gst_buffer_unref (script_tag); + goto exit; + } + tmp = gst_buffer_new_and_alloc (2 + 0 + 1); data = GST_BUFFER_DATA (tmp); data[0] = 0; /* 0 byte size */ @@ -946,6 +960,7 @@ gst_flv_mux_create_metadata (GstFlvMux * mux) GST_WRITE_UINT32_BE (data + 11 + 13 + 1, tags_written); +exit: return script_tag; } @@ -1089,7 +1104,7 @@ gst_flv_mux_write_header (GstFlvMux * mux) GstFlowReturn ret; header = gst_flv_mux_create_header (mux); - metadata = gst_flv_mux_create_metadata (mux); + metadata = gst_flv_mux_create_metadata (mux, TRUE); video_codec_data = NULL; audio_codec_data = NULL; @@ -1407,8 +1422,9 @@ gst_flv_mux_handle_buffer (GstCollectPads2 * pads, GstCollectData2 * cdata, } if (mux->new_tags) { - GstBuffer *buf = gst_flv_mux_create_metadata (mux); - gst_flv_mux_push (mux, buf); + GstBuffer *buf = gst_flv_mux_create_metadata (mux, FALSE); + if (buf) + gst_flv_mux_push (mux, buf); mux->new_tags = FALSE; } From b175d1cbc491b33ad8b11a8a4098c0a5d1cacacd Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Fri, 16 Dec 2011 19:15:38 +0100 Subject: [PATCH 47/65] flvmux: properly determine final duration ... which can be authoratively obtained from our own written timestamps. --- gst/flv/gstflvmux.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index 4a8b18ff46..c7ee0a8316 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -1250,12 +1250,6 @@ gst_flv_mux_determine_duration (GstFlvMux * mux) } } - if (duration == GST_CLOCK_TIME_NONE) { - GST_DEBUG_OBJECT (mux, "not able to determine duration " - "from pad timestamps, assuming 0"); - return 0; - } - return duration; } @@ -1269,6 +1263,7 @@ gst_flv_mux_rewrite_header (GstFlvMux * mux) GList *l; guint32 index_len, allocate_size; guint32 i, index_skip; + GstClockTime dur; if (mux->streamable) return GST_FLOW_OK; @@ -1281,9 +1276,12 @@ gst_flv_mux_rewrite_header (GstFlvMux * mux) return GST_FLOW_OK; } - /* if we were not able to determine the duration before, set it now */ - if (mux->duration == GST_CLOCK_TIME_NONE) - mux->duration = gst_flv_mux_determine_duration (mux); + /* determine duration now based on our own timestamping, + * so that it is likely many times better and consistent + * than whatever obtained by some query */ + dur = gst_flv_mux_determine_duration (mux); + if (dur != GST_CLOCK_TIME_NONE) + mux->duration = dur; /* rewrite the duration tag */ d = gst_guint64_to_gdouble (mux->duration); From 93ac6fb47ed70039e9700e6c73770afa10b9cd31 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Fri, 9 Sep 2011 11:42:09 +0100 Subject: [PATCH 48/65] ac3parse: let bsid 9 and 10 through Files with 9 and 10 happen, and seem to comply with the <= 8 format, so let them through. The spec says nothing about 9 and 10. https://bugzilla.gnome.org/show_bug.cgi?id=658546 --- gst/audioparsers/gstac3parse.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/gst/audioparsers/gstac3parse.c b/gst/audioparsers/gstac3parse.c index 65c958cd94..6fa46584c4 100644 --- a/gst/audioparsers/gstac3parse.c +++ b/gst/audioparsers/gstac3parse.c @@ -321,7 +321,9 @@ gst_ac3_parse_frame_header_ac3 (GstAc3Parse * ac3parse, GstBuffer * buf, /* spec not quite clear here: decoder should decode if less than 8, * but seemingly only defines 6 and 8 cases */ - if (bsid > 8) { + /* Files with 9 and 10 happen, and seem to comply with the <= 8 + format, so let them through. The spec says nothing about 9 and 10 */ + if (bsid > 10) { GST_DEBUG_OBJECT (ac3parse, "unexpected bsid=%d", bsid); return FALSE; } else if (bsid != 8 && bsid != 6) { From 4f98b4ec853cface214d3b4a5e9008986c8d2ef2 Mon Sep 17 00:00:00 2001 From: "Oleksij Rempel (Alexey Fisher)" Date: Wed, 21 Dec 2011 09:09:27 +0100 Subject: [PATCH 49/65] ac3parse: remove unused variable remove unused variable to fix compile error: make -C audioparsers make[3]: Betrete Verzeichnis '/home/lex/tmp/gst-plugins-good/gst/audioparsers' CC libgstaudioparsers_la-gstaacparse.lo gstaacparse.c: In function 'gst_aac_parse_read_loas_audio_specific_config': gstaacparse.c:446:12: error: variable 'sbr' set but not used [-Werror=unused-but-set-variable] cc1: all warnings being treated as errors Signed-off-by: Oleksij Rempel (Alexey Fisher) --- gst/audioparsers/gstaacparse.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/gst/audioparsers/gstaacparse.c b/gst/audioparsers/gstaacparse.c index 8a0d6f606b..f1e4de2f16 100644 --- a/gst/audioparsers/gstaacparse.c +++ b/gst/audioparsers/gstaacparse.c @@ -443,7 +443,6 @@ gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse, GstBitReader * br, gint * sample_rate, gint * channels, guint32 * bits) { guint8 audio_object_type, channel_configuration; - gboolean sbr = FALSE; if (!gst_aac_parse_get_audio_object_type (aacparse, br, &audio_object_type)) return FALSE; @@ -461,7 +460,6 @@ gst_aac_parse_read_loas_audio_specific_config (GstAacParse * aacparse, if (audio_object_type == 5) { GST_LOG_OBJECT (aacparse, "Audio object type 5, so rereading sampling rate..."); - sbr = TRUE; if (!gst_aac_parse_get_audio_sample_rate (aacparse, br, sample_rate)) return FALSE; } From 628ae5c1a1001f02de605d89b753759fce4ed5c8 Mon Sep 17 00:00:00 2001 From: Mark Nauwelaerts Date: Wed, 21 Dec 2011 15:06:57 +0100 Subject: [PATCH 50/65] matroskademux: improve decision to fall back to scanning when seeking ... which is basically iff not streaming and no entry found in index --- gst/matroska/matroska-demux.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 81f83f0150..2ce2d76569 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -1962,7 +1962,7 @@ gst_matroska_demux_handle_seek_event (GstMatroskaDemux * demux, seeksegment.last_stop, &demux->seek_index, &demux->seek_entry)) == NULL) { /* pull mode without index can scan later on */ - if (demux->common.index || demux->streaming) { + if (demux->streaming) { GST_DEBUG_OBJECT (demux, "No matching seek entry in index"); GST_OBJECT_UNLOCK (demux); return FALSE; @@ -1999,7 +1999,7 @@ next: GST_PAD_STREAM_LOCK (demux->common.sinkpad); /* pull mode without index can do some scanning */ - if (!demux->streaming && !demux->common.index) { + if (!demux->streaming && !entry) { /* need to stop flushing upstream as we need it next */ if (flush) gst_pad_push_event (demux->common.sinkpad, gst_event_new_flush_stop ()); From a7d6690f92e6c739d3b3e28d99d1f7840f8c59d9 Mon Sep 17 00:00:00 2001 From: Branko Subasic Date: Wed, 21 Dec 2011 17:43:10 +0100 Subject: [PATCH 51/65] matroskademux: do not consider duration of non-finalized file ... to avoid it clamping requested seek position. Non-finalized file case, determined by whether _parse_blockgroup_or_simpleblock ever updates the segment duration. Fixes #652195. --- gst/matroska/matroska-demux.c | 18 ++++++++++++++++++ gst/matroska/matroska-demux.h | 3 +++ 2 files changed, 21 insertions(+) diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c index 2ce2d76569..e80febcb5f 100644 --- a/gst/matroska/matroska-demux.c +++ b/gst/matroska/matroska-demux.c @@ -480,6 +480,8 @@ gst_matroska_demux_reset (GstElement * element) gst_buffer_unref (demux->common.cached_buffer); demux->common.cached_buffer = NULL; } + + demux->invalid_duration = FALSE; } static GstBuffer * @@ -1928,6 +1930,16 @@ gst_matroska_demux_handle_seek_event (GstMatroskaDemux * demux, * segment when we close the current segment. */ memcpy (&seeksegment, &demux->common.segment, sizeof (GstSegment)); + /* pull mode without index means that the actual duration is not known, + * we might be playing a file that's still being recorded + * so, invalidate our current duration, which is only a moving target, + * and should not be used to clamp anything */ + if (!demux->streaming && !demux->common.index && + demux->invalid_duration) { + gst_segment_set_duration (&seeksegment, GST_FORMAT_TIME, + GST_CLOCK_TIME_NONE); + } + if (event) { GST_DEBUG_OBJECT (demux, "configuring seek"); gst_segment_set_seek (&seeksegment, rate, format, flags, @@ -1942,6 +1954,11 @@ gst_matroska_demux_handle_seek_event (GstMatroskaDemux * demux, } } + /* restore segment duration (if any effect), + * would be determined again when parsing, but anyway ... */ + gst_segment_set_duration (&seeksegment, GST_FORMAT_TIME, + demux->common.segment.duration); + flush = ! !(flags & GST_SEEK_FLAG_FLUSH); keyunit = ! !(flags & GST_SEEK_FLAG_KEY_UNIT); @@ -3501,6 +3518,7 @@ gst_matroska_demux_parse_blockgroup_or_simpleblock (GstMatroskaDemux * demux, gst_element_post_message (GST_ELEMENT_CAST (demux), gst_message_new_duration (GST_OBJECT_CAST (demux), GST_FORMAT_TIME, GST_CLOCK_TIME_NONE)); + demux->invalid_duration = TRUE; } else { GST_OBJECT_UNLOCK (demux); } diff --git a/gst/matroska/matroska-demux.h b/gst/matroska/matroska-demux.h index 2b08b2e39e..22093ea4a3 100644 --- a/gst/matroska/matroska-demux.h +++ b/gst/matroska/matroska-demux.h @@ -95,6 +95,9 @@ typedef struct _GstMatroskaDemux { /* gap handling */ guint64 max_gap_time; + + /* for non-finalized files, with invalid segment duration */ + gboolean invalid_duration; } GstMatroskaDemux; typedef struct _GstMatroskaDemuxClass { From 348afe1669a21998bb7bdff0209045b09ed63d44 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Sebastian=20Dr=C3=B6ge?= Date: Sat, 24 Dec 2011 10:57:42 +0100 Subject: [PATCH 52/65] wavparse: Use scale_ceil() functions from core instead of custom ones --- gst/wavparse/gstwavparse.c | 63 ++++++++++++-------------------------- 1 file changed, 19 insertions(+), 44 deletions(-) diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index b4e5df4ecd..719359b026 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -281,37 +281,6 @@ gst_wavparse_create_sourcepad (GstWavParse * wavparse) GST_DEBUG_OBJECT (wavparse, "srcpad created"); } -/* Compute (value * nom) % denom, avoiding overflow. This can be used - * to perform ceiling or rounding division together with - * gst_util_uint64_scale[_int]. */ -#define uint64_scale_modulo(val, nom, denom) \ - ((val % denom) * (nom % denom) % denom) - -/* Like gst_util_uint64_scale, but performs ceiling division. */ -static guint64 -uint64_ceiling_scale_int (guint64 val, gint num, gint denom) -{ - guint64 result = gst_util_uint64_scale_int (val, num, denom); - - if (uint64_scale_modulo (val, num, denom) == 0) - return result; - else - return result + 1; -} - -/* Like gst_util_uint64_scale, but performs ceiling division. */ -static guint64 -uint64_ceiling_scale (guint64 val, guint64 num, guint64 denom) -{ - guint64 result = gst_util_uint64_scale (val, num, denom); - - if (uint64_scale_modulo (val, num, denom) == 0) - return result; - else - return result + 1; -} - - /* FIXME: why is that not in use? */ #if 0 static void @@ -792,12 +761,12 @@ gst_wavparse_time_to_bytepos (GstWavParse * wav, gint64 ts, gint64 * bytepos) } if (wav->bps > 0) { - *bytepos = uint64_ceiling_scale (ts, (guint64) wav->bps, GST_SECOND); + *bytepos = gst_util_uint64_scale_ceil (ts, (guint64) wav->bps, GST_SECOND); return TRUE; } else if (wav->fact) { guint64 bps = gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact); - *bytepos = uint64_ceiling_scale (ts, bps, GST_SECOND); + *bytepos = gst_util_uint64_scale_ceil (ts, bps, GST_SECOND); return TRUE; } @@ -1159,12 +1128,14 @@ gst_wavparse_calculate_duration (GstWavParse * wav) if (wav->bps > 0) { GST_INFO_OBJECT (wav, "Got datasize %" G_GUINT64_FORMAT, wav->datasize); wav->duration = - uint64_ceiling_scale (wav->datasize, GST_SECOND, (guint64) wav->bps); + gst_util_uint64_scale_ceil (wav->datasize, GST_SECOND, + (guint64) wav->bps); GST_INFO_OBJECT (wav, "Got duration (bps) %" GST_TIME_FORMAT, GST_TIME_ARGS (wav->duration)); return TRUE; } else if (wav->fact) { - wav->duration = uint64_ceiling_scale_int (GST_SECOND, wav->fact, wav->rate); + wav->duration = + gst_util_uint64_scale_int_ceil (GST_SECOND, wav->fact, wav->rate); GST_INFO_OBJECT (wav, "Got duration (fact) %" GST_TIME_FORMAT, GST_TIME_ARGS (wav->duration)); return TRUE; @@ -1976,9 +1947,10 @@ iterate_adapter: if (wav->bps > 0) { /* and timestamps if we have a bitrate, be careful for overflows */ - timestamp = uint64_ceiling_scale (pos, GST_SECOND, (guint64) wav->bps); + timestamp = + gst_util_uint64_scale_ceil (pos, GST_SECOND, (guint64) wav->bps); next_timestamp = - uint64_ceiling_scale (nextpos, GST_SECOND, (guint64) wav->bps); + gst_util_uint64_scale_ceil (nextpos, GST_SECOND, (guint64) wav->bps); duration = next_timestamp - timestamp; /* update current running segment position */ @@ -1989,8 +1961,8 @@ iterate_adapter: guint64 bps = gst_util_uint64_scale_int (wav->datasize, wav->rate, wav->fact); /* and timestamps if we have a bitrate, be careful for overflows */ - timestamp = uint64_ceiling_scale (pos, GST_SECOND, bps); - next_timestamp = uint64_ceiling_scale (nextpos, GST_SECOND, bps); + timestamp = gst_util_uint64_scale_ceil (pos, GST_SECOND, bps); + next_timestamp = gst_util_uint64_scale_ceil (nextpos, GST_SECOND, bps); duration = next_timestamp - timestamp; } else { /* no bitrate, all we know is that the first sample has timestamp 0, all @@ -2277,10 +2249,12 @@ gst_wavparse_sink_event (GstPad * pad, GstEvent * event) if (bps) { if (start >= 0) start = - uint64_ceiling_scale (start, GST_SECOND, (guint64) wav->bps); + gst_util_uint64_scale_ceil (start, GST_SECOND, + (guint64) wav->bps); if (stop >= 0) stop = - uint64_ceiling_scale (stop, GST_SECOND, (guint64) wav->bps); + gst_util_uint64_scale_ceil (stop, GST_SECOND, + (guint64) wav->bps); } } } else { @@ -2395,13 +2369,14 @@ gst_wavparse_pad_convert (GstPad * pad, "src=%" G_GINT64_FORMAT ", offset=%" G_GINT64_FORMAT, src_value, wavparse->offset); if (wavparse->bps > 0) - *dest_value = uint64_ceiling_scale (src_value, GST_SECOND, + *dest_value = gst_util_uint64_scale_ceil (src_value, GST_SECOND, (guint64) wavparse->bps); else if (wavparse->fact) { - guint64 bps = uint64_ceiling_scale_int (wavparse->datasize, + guint64 bps = gst_util_uint64_scale_int_ceil (wavparse->datasize, wavparse->rate, wavparse->fact); - *dest_value = uint64_ceiling_scale_int (src_value, GST_SECOND, bps); + *dest_value = + gst_util_uint64_scale_int_ceil (src_value, GST_SECOND, bps); } else { res = FALSE; } From 89521166b24c394785842819c416575170677c5f Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sun, 25 Dec 2011 14:23:29 +0000 Subject: [PATCH 53/65] flvmux: don't try to push already-freed buffers Fixes unit test. --- gst/flv/gstflvmux.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/gst/flv/gstflvmux.c b/gst/flv/gstflvmux.c index c7ee0a8316..aac216ace8 100644 --- a/gst/flv/gstflvmux.c +++ b/gst/flv/gstflvmux.c @@ -567,6 +567,7 @@ gst_flv_mux_release_pad (GstElement * element, GstPad * pad) static GstFlowReturn gst_flv_mux_push (GstFlvMux * mux, GstBuffer * buffer) { + buffer = gst_buffer_make_metadata_writable (buffer); gst_buffer_set_caps (buffer, GST_PAD_CAPS (mux->srcpad)); /* pushing the buffer that rewrites the header will make it no longer be the * total output size in bytes, but it doesn't matter at that point */ @@ -937,6 +938,7 @@ end: if (!tags_written) { gst_buffer_unref (script_tag); + script_tag = NULL; goto exit; } From 7202d37c9d2c566f49d035489e04ddecec16b33d Mon Sep 17 00:00:00 2001 From: Nicola Murino Date: Wed, 21 Dec 2011 20:50:21 +0100 Subject: [PATCH 54/65] jpegdec: fix peer_caps leak https://bugzilla.gnome.org/show_bug.cgi?id=666688 --- ext/jpeg/gstjpegdec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/ext/jpeg/gstjpegdec.c b/ext/jpeg/gstjpegdec.c index fe5652bb9b..18cf20d320 100644 --- a/ext/jpeg/gstjpegdec.c +++ b/ext/jpeg/gstjpegdec.c @@ -808,7 +808,7 @@ gst_jpeg_dec_getcaps (GstPad * pad) templ_caps = gst_pad_get_pad_template_caps (pad); caps = gst_caps_intersect_full (peer_caps, templ_caps, GST_CAPS_INTERSECT_FIRST); - + gst_caps_unref (peer_caps); gst_object_unref (peer); } else { caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); From e72b55b6ac84ae5748ea3e462802a96a71b4acca Mon Sep 17 00:00:00 2001 From: Yaakov Selkowitz Date: Mon, 26 Dec 2011 22:22:59 +0000 Subject: [PATCH 55/65] waveform: add autotools bits for waveform plugin https://bugzilla.gnome.org/show_bug.cgi?id=666485 --- configure.ac | 7 +++++++ sys/Makefile.am | 8 +++++++- sys/waveform/Makefile.am | 10 +++------- 3 files changed, 17 insertions(+), 8 deletions(-) diff --git a/configure.ac b/configure.ac index 70fc566770..681bc37ef2 100644 --- a/configure.ac +++ b/configure.ac @@ -444,6 +444,12 @@ int main () AC_SUBST(HAVE_DIRECTSOUND) ]) +dnl *** Win32 WaveOut *** +translit(dnm, m, l) AM_CONDITIONAL(USE_WAVEFORM, true) +AG_GST_CHECK_FEATURE(WAVEFORM, [Win32 WaveForm], waveformsink, [ + AC_CHECK_HEADER(mmsystem.h, HAVE_WAVEFORM="yes", HAVE_WAVEFORM="no", [#include ]) +]) + dnl *** OSS audio *** (Linux, *BSD) translit(dnm, m, l) AM_CONDITIONAL(USE_OSS, true) AG_GST_CHECK_FEATURE(OSS, [OSS audio], ossaudio, [ @@ -1046,6 +1052,7 @@ AM_CONDITIONAL(USE_SOUP, false) AM_CONDITIONAL(USE_SPEEX, false) AM_CONDITIONAL(USE_SUNAUDIO, false) AM_CONDITIONAL(USE_TAGLIB, false) +AM_CONDITIONAL(USE_WAVEFORM, false) AM_CONDITIONAL(USE_WAVPACK, false) AM_CONDITIONAL(USE_X, false) AM_CONDITIONAL(USE_XSHM, false) diff --git a/sys/Makefile.am b/sys/Makefile.am index ab08d046d1..7ccd9e77aa 100644 --- a/sys/Makefile.am +++ b/sys/Makefile.am @@ -23,6 +23,12 @@ else DIRECTSOUND_DIR= endif +if USE_WAVEFORM +WAVEFORM_DIR=waveform +else +WAVEFORM_DIR= +endif + if USE_SUNAUDIO SUNAUDIO_DIR=sunaudio else @@ -77,7 +83,7 @@ else XIMAGE_DIR= endif -SUBDIRS=$(DIRECTSOUND_DIR) $(OSS_DIR) $(OSS4_DIR) $(OSX_AUDIO_DIR) $(OSX_VIDEO_DIR) $(SUNAUDIO_DIR) $(V4L2_DIR) $(XIMAGE_DIR) +SUBDIRS=$(DIRECTSOUND_DIR) $(WAVEFORM_DIR) $(OSS_DIR) $(OSS4_DIR) $(OSX_AUDIO_DIR) $(OSX_VIDEO_DIR) $(SUNAUDIO_DIR) $(V4L2_DIR) $(XIMAGE_DIR) DIST_SUBDIRS=directsound oss oss4 osxaudio osxvideo sunaudio v4l2 waveform ximage diff --git a/sys/waveform/Makefile.am b/sys/waveform/Makefile.am index a9562a818c..9a9fac3d75 100644 --- a/sys/waveform/Makefile.am +++ b/sys/waveform/Makefile.am @@ -1,19 +1,15 @@ plugin_LTLIBRARIES = libgstwaveformsink.la -# FIXME: Replace DIRECTSOUND CFLAGS+LIBS with waveform related ones and fix -# the configure.ac + sys/Makefile.am to get this stuff building in MingW -# For now, it's just disted for use in the VS builds. - libgstwaveformsink_la_SOURCES = gstwaveformsink.c gstwaveformplugin.c libgstwaveformsink_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS) \ - $(GST_PLUGINS_BASE_CFLAGS) $(DIRECTSOUND_CFLAGS) + $(GST_PLUGINS_BASE_CFLAGS) libgstwaveformsink_la_LIBADD = \ $(GST_PLUGINS_BASE_LIBS) \ -lgstaudio-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR) \ $(GST_BASE_LIBS) \ $(GST_LIBS) \ - $(DIRECTSOUND_LIBS) -libgstwaveformsink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) $(DIRECTSOUND_LDFLAGS) + -lwinmm +libgstwaveformsink_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstwaveformsink_la_LIBTOOLFLAGS = --tag=disable-static noinst_HEADERS = gstwaveformsink.h From 0c4b60f01044e097d30095ceb3b7c65e57126a8f Mon Sep 17 00:00:00 2001 From: John Ogness Date: Wed, 21 Dec 2011 13:22:03 +0100 Subject: [PATCH 56/65] udpsrc: drop dataless UDP packets It is allowed to send/receive UDP packets with no data. When such a packet is available, select() will return with success but ioctl(FIONREAD) will return 0. But a read() must still occur in order to clear off the UDP packet from the queue. This patch will read the dataless packet from the socket. If select() was woken for other reasons (and FIONREAD returns 0), this may result in a UDP packet getting accidentally dropped. But since UDP is not reliable, this is acceptable. NOTE: This patch fixes a nasty bug where sending a dataless UDP packet to a udpsrc instance will cause an infinite loop. https://bugzilla.gnome.org/show_bug.cgi?id=666644 Signed-off-by: John Ogness --- gst/udp/gstudpsrc.c | 16 ++++++++++++---- 1 file changed, 12 insertions(+), 4 deletions(-) diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c index ec644af00a..dce2c9981a 100644 --- a/gst/udp/gstudpsrc.c +++ b/gst/udp/gstudpsrc.c @@ -483,12 +483,20 @@ retry: IOCTL_SOCKET (udpsrc->sock.fd, FIONREAD, &readsize)) < 0)) goto ioctl_failed; - /* if we get here and there is nothing to read from the socket, the select got - * woken up by activity on the socket but it was not a read. We know someone - * will also do something with the socket so that we don't go into an infinite - * loop in the select(). */ + /* If we get here and the readsize is zero, then either select was woken up + * by activity that is not a read, or a poll error occurred, or a UDP packet + * was received that has no data. Since we cannot identify which case it is, + * we handle all of them. This could possibly lead to a UDP packet getting + * lost, but since UDP is not reliable, we can accept this. */ if (G_UNLIKELY (!readsize)) { + /* try to read a packet (and it will be ignored), + * in case a packet with no data arrived */ + recvfrom (udpsrc->sock.fd, (char *)&slen, 0, 0, &sa.sa, &slen); + + /* clear any error, in case a poll error occurred */ clear_error (udpsrc); + + /* poll again */ goto retry; } From b744ad03ccf85cfe34f00866964fc97fad385975 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 27 Dec 2011 11:49:10 +0000 Subject: [PATCH 57/65] udpsrc: add unit test that sends 0-size packet https://bugzilla.gnome.org/show_bug.cgi?id=666644 --- tests/check/Makefile.am | 4 ++ tests/check/elements/udpsrc.c | 120 ++++++++++++++++++++++++++++++++++ 2 files changed, 124 insertions(+) create mode 100644 tests/check/elements/udpsrc.c diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 27dde40f06..d16e98528c 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -147,6 +147,7 @@ check_PROGRAMS = \ elements/shapewipe \ elements/spectrum \ elements/udpsink \ + elements/udpsrc \ elements/videocrop \ elements/videofilter \ elements/y4menc \ @@ -261,6 +262,9 @@ elements_sunaudio_LDADD = \ $(GST_PLUGINS_BASE_LIBS) -lgstinterfaces-@GST_MAJORMINOR@ \ $(LDADD) +elements_udpsrc_CFLAGS = $(AM_CFLAGS) $(GIO_CFLAGS) +elements_udpsrc_LDADD = $(LDADD) $(GIO_LIBS) + elements_videocrop_LDADD = $(GST_BASE_LIBS) $(LDADD) elements_videocrop_CFLAGS = $(GST_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS) diff --git a/tests/check/elements/udpsrc.c b/tests/check/elements/udpsrc.c new file mode 100644 index 0000000000..d84334a3e1 --- /dev/null +++ b/tests/check/elements/udpsrc.c @@ -0,0 +1,120 @@ +/* GStreamer UDP source unit tests + * Copyright (C) 2011 Tim-Philipp Müller + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ +#include +#include +#include +#include + +static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS_ANY); + +GST_START_TEST (test_udpsrc_empty_packet) +{ + GstElement *udpsrc; + GSocket *socket; + GstPad *sinkpad; + int port = 0; + + udpsrc = gst_check_setup_element ("udpsrc"); + fail_unless (udpsrc != NULL); + g_object_set (udpsrc, "port", 0, NULL); + + sinkpad = gst_check_setup_sink_pad_by_name (udpsrc, &sinktemplate, "src"); + fail_unless (sinkpad != NULL); + gst_pad_set_active (sinkpad, TRUE); + + gst_element_set_state (udpsrc, GST_STATE_PLAYING); + g_object_get (udpsrc, "port", &port, NULL); + GST_INFO ("udpsrc port = %d", port); + + socket = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM, + G_SOCKET_PROTOCOL_UDP, NULL); + + if (socket != NULL) { + GSocketAddress *sa; + GInetAddress *ia; + gchar *s; + + ia = g_inet_address_new_loopback (G_SOCKET_FAMILY_IPV4); + s = g_inet_address_to_string (ia); + GST_LOG ("inet address %s", s); + g_free (s); + sa = g_inet_socket_address_new (ia, port); + + if (g_socket_send_to (socket, sa, "HeLL0", 0, NULL, NULL) == 0) { + GST_INFO ("sent 0 bytes"); + if (g_socket_send_to (socket, sa, "HeLL0", 6, NULL, NULL) == 6) { + GstBuffer *buf; + guint len; + + GST_INFO ("sent 6 bytes"); + + g_usleep (G_USEC_PER_SEC / 2); + + len = g_list_length (buffers); + GST_INFO ("%u buffers", len); + fail_unless (len == 1 || len == 2); + + /* last buffer should be our HeLL0 string */ + buf = GST_BUFFER (g_list_nth_data (buffers, len - 1)); + fail_unless_equals_int (GST_BUFFER_SIZE (buf), 6); + fail_unless_equals_string ((gchar *) GST_BUFFER_DATA (buf), "HeLL0"); + + /* if there's another buffer, it should be 0 bytes */ + if (len == 2) { + buf = GST_BUFFER (g_list_nth_data (buffers, 0)); + fail_unless_equals_int (GST_BUFFER_SIZE (buf), 0); + } + } else { + GST_WARNING ("send_to(6 bytes) failed"); + } + } else { + GST_WARNING ("send_to(0 bytes) failed"); + } + + g_object_unref (sa); + g_object_unref (ia); + } else { + GST_WARNING ("Could not create IPv4 UDP socket for unit test"); + } + + gst_element_set_state (udpsrc, GST_STATE_NULL); + + gst_check_teardown_pad_by_name (udpsrc, "src"); + gst_check_teardown_element (udpsrc); + + g_object_unref (socket); +} + +GST_END_TEST; + +static Suite * +udpsrc_suite (void) +{ + Suite *s = suite_create ("udpsrc"); + TCase *tc_chain = tcase_create ("udpsrc"); + + suite_add_tcase (s, tc_chain); + tcase_add_test (tc_chain, test_udpsrc_empty_packet); + return s; +} + +GST_CHECK_MAIN (udpsrc) From 86cd5bd7f2b0fe5aa594a94536dee0bd7d715798 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 27 Dec 2011 11:50:03 +0000 Subject: [PATCH 58/65] udpsrc: fix valgrind warning https://bugzilla.gnome.org/show_bug.cgi?id=666644 --- gst/udp/gstudpsrc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c index dce2c9981a..3377df24f7 100644 --- a/gst/udp/gstudpsrc.c +++ b/gst/udp/gstudpsrc.c @@ -491,7 +491,8 @@ retry: if (G_UNLIKELY (!readsize)) { /* try to read a packet (and it will be ignored), * in case a packet with no data arrived */ - recvfrom (udpsrc->sock.fd, (char *)&slen, 0, 0, &sa.sa, &slen); + slen = sizeof (sa); + recvfrom (udpsrc->sock.fd, (char *) &slen, 0, 0, &sa.sa, &slen); /* clear any error, in case a poll error occurred */ clear_error (udpsrc); From 56c1003f7a23a25d46e21487f086b88a405611f4 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 27 Dec 2011 12:06:16 +0000 Subject: [PATCH 59/65] tests: make git ignore new unit test binary --- tests/check/elements/.gitignore | 1 + 1 file changed, 1 insertion(+) diff --git a/tests/check/elements/.gitignore b/tests/check/elements/.gitignore index c868fc81ef..1e2c8b2e4a 100644 --- a/tests/check/elements/.gitignore +++ b/tests/check/elements/.gitignore @@ -53,6 +53,7 @@ souphttpsrc spectrum sunaudio udpsink +udpsrc videocrop videofilter wavpackdec From 2be912cb4559d85756a22e3e04cd4e609779f089 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Tue, 27 Dec 2011 22:59:03 +0000 Subject: [PATCH 60/65] waveformsink: fix compiler warnings with MingW https://bugzilla.gnome.org/show_bug.cgi?id=666485 --- sys/waveform/gstwaveformsink.c | 12 ++++-------- 1 file changed, 4 insertions(+), 8 deletions(-) diff --git a/sys/waveform/gstwaveformsink.c b/sys/waveform/gstwaveformsink.c index 492742b1d9..83fe7492a6 100644 --- a/sys/waveform/gstwaveformsink.c +++ b/sys/waveform/gstwaveformsink.c @@ -49,10 +49,6 @@ GST_DEBUG_CATEGORY_STATIC (waveformsink_debug); -static void gst_waveform_sink_base_init (gpointer g_class); -static void gst_waveform_sink_class_init (GstWaveFormSinkClass * klass); -static void gst_waveform_sink_init (GstWaveFormSink * wfsink, - GstWaveFormSinkClass * g_class); static void gst_waveform_sink_finalise (GObject * object); static void gst_waveform_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); @@ -154,7 +150,7 @@ static void gst_waveform_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { - GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (object); + /* GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (object); */ switch (prop_id) { default: @@ -167,7 +163,7 @@ static void gst_waveform_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { - GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (object); + /* GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (object); */ switch (prop_id) { default: @@ -348,7 +344,7 @@ gst_waveform_sink_getcaps (GstBaseSink * bsink) static gboolean gst_waveform_sink_open (GstAudioSink * asink) { - GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (asink); + /* GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (asink); */ /* nothing to do here as the device needs to be opened with the format we will use */ @@ -448,7 +444,7 @@ gst_waveform_sink_unprepare (GstAudioSink * asink) static gboolean gst_waveform_sink_close (GstAudioSink * asink) { - GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (asink); + /* GstWaveFormSink *wfsink = GST_WAVEFORM_SINK (asink); */ return TRUE; } From b82d9ca240542814883a26273d079962bb10680b Mon Sep 17 00:00:00 2001 From: Peter Seiderer Date: Wed, 14 Dec 2011 10:14:20 +0100 Subject: [PATCH 61/65] multifilesink: post better error message when we run out of disk space Map write errno ENOSPC to GST_RESOURCE_ERROR_NO_SPACE_LEFT. --- gst/multifile/gstmultifilesink.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/gst/multifile/gstmultifilesink.c b/gst/multifile/gstmultifilesink.c index 984e223961..2be3b56ea5 100644 --- a/gst/multifile/gstmultifilesink.c +++ b/gst/multifile/gstmultifilesink.c @@ -656,8 +656,15 @@ write_error: return GST_FLOW_ERROR; } stdio_write_error: - GST_ELEMENT_ERROR (multifilesink, RESOURCE, WRITE, - ("Error while writing to file."), (NULL)); + switch (errno) { + case ENOSPC: + GST_ELEMENT_ERROR (multifilesink, RESOURCE, NO_SPACE_LEFT, + ("Error while writing to file."), ("%s", g_strerror (errno))); + break; + default: + GST_ELEMENT_ERROR (multifilesink, RESOURCE, WRITE, + ("Error while writing to file."), ("%s", g_strerror (errno))); + } return GST_FLOW_ERROR; } From 12ff707854922305992564476ba0964d7c097bfe Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Wed, 4 Jan 2012 17:59:55 +0000 Subject: [PATCH 62/65] tests: fix some leaks and remove files when done in qtmux test --- tests/check/elements/qtmux.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/tests/check/elements/qtmux.c b/tests/check/elements/qtmux.c index 484296badf..a9364c984a 100644 --- a/tests/check/elements/qtmux.c +++ b/tests/check/elements/qtmux.c @@ -28,6 +28,8 @@ #include #endif +#include + #include #include @@ -614,8 +616,7 @@ test_mp3_enc_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - gst_element_class_add_static_pad_template (element_class, - &sink_template); + gst_element_class_add_static_pad_template (element_class, &sink_template); gst_element_class_add_static_pad_template (element_class, &src_template); gst_element_class_set_details_simple (element_class, "MPEG1 Audio Encoder", @@ -838,7 +839,12 @@ test_average_bitrate_custom (const gchar * elementname, (guint) gst_util_uint64_scale_round ((guint64) total_bytes, (guint64) 8 * GST_SECOND, (guint64) total_duration); fail_unless (bitrate == expected); + gst_tag_list_free (taglist); } + + /* delete file */ + g_unlink (location); + g_free (location); } GST_START_TEST (test_average_bitrate) From 3489b691c5e67abd38ee6e4335d31a925a237371 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Thu, 5 Jan 2012 19:08:03 +0000 Subject: [PATCH 63/65] isomp4: remove dead assignment --- gst/isomp4/gstqtmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index 5d182396e5..f575ab12ca 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -1481,7 +1481,7 @@ gst_qt_mux_set_header_on_caps (GstQTMux * mux, GstBuffer * buf) GstStructure *structure; GValue array = { 0 }; GValue value = { 0 }; - GstCaps *caps = GST_PAD_CAPS (mux->srcpad); + GstCaps *caps; caps = gst_caps_copy (GST_PAD_CAPS (mux->srcpad)); structure = gst_caps_get_structure (caps, 0); From a6d9f6a3ced3c64996623ecbc49e8499cb6a97e0 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Thu, 5 Jan 2012 19:25:33 +0000 Subject: [PATCH 64/65] isomp4: fix caps leak --- gst/isomp4/gstqtmux.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index f575ab12ca..2ce98400b3 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -2585,7 +2585,6 @@ gst_qt_mux_audio_sink_set_caps (GstPad * pad, GstCaps * caps) AtomInfo *ext_atom = NULL; gint constant_size = 0; const gchar *stream_format; - GstCaps *current_caps = NULL; /* find stream data */ qtpad = (GstQTPad *) gst_pad_get_element_private (pad); @@ -2597,10 +2596,14 @@ gst_qt_mux_audio_sink_set_caps (GstPad * pad, GstCaps * caps) * the old caps are a subset of the new one (this means upstream * added more info to the caps, as both should be 'fixed' caps) */ if (qtpad->fourcc) { + GstCaps *current_caps = NULL; + gboolean is_subset; g_object_get (pad, "caps", ¤t_caps, NULL); g_assert (caps != NULL); - if (!gst_qtmux_caps_is_subset_full (qtmux, current_caps, caps)) { + is_subset = gst_qtmux_caps_is_subset_full (qtmux, current_caps, caps); + gst_caps_unref (current_caps); + if (!is_subset) { goto refuse_renegotiation; } GST_DEBUG_OBJECT (qtmux, @@ -2907,7 +2910,6 @@ gst_qt_mux_video_sink_set_caps (GstPad * pad, GstCaps * caps) GList *ext_atom_list = NULL; gboolean sync = FALSE; int par_num, par_den; - GstCaps *current_caps = NULL; /* find stream data */ qtpad = (GstQTPad *) gst_pad_get_element_private (pad); @@ -2919,10 +2921,14 @@ gst_qt_mux_video_sink_set_caps (GstPad * pad, GstCaps * caps) * the old caps are a subset of the new one (this means upstream * added more info to the caps, as both should be 'fixed' caps) */ if (qtpad->fourcc) { + GstCaps *current_caps = NULL; + gboolean is_subset; g_object_get (pad, "caps", ¤t_caps, NULL); g_assert (caps != NULL); - if (!gst_qtmux_caps_is_subset_full (qtmux, current_caps, caps)) { + is_subset = gst_qtmux_caps_is_subset_full (qtmux, current_caps, caps); + gst_caps_unref (current_caps); + if (!is_subset) { goto refuse_renegotiation; } GST_DEBUG_OBJECT (qtmux, From 2b2c0940f1b7ce8a858a26ad5e246fd645a83830 Mon Sep 17 00:00:00 2001 From: Vincent Penquerc'h Date: Mon, 9 Jan 2012 17:28:17 +0000 Subject: [PATCH 65/65] matroskamux: fix codec string leaks --- gst/matroska/matroska-mux.c | 103 +++++++++++++++++++++++------------- 1 file changed, 67 insertions(+), 36 deletions(-) diff --git a/gst/matroska/matroska-mux.c b/gst/matroska/matroska-mux.c index f454247af6..e6fb0593cc 100644 --- a/gst/matroska/matroska-mux.c +++ b/gst/matroska/matroska-mux.c @@ -853,6 +853,15 @@ gst_matroska_mux_handle_sink_event (GstCollectPads2 * pads, return TRUE; } +static void +gst_matroska_mux_set_codec_id (GstMatroskaTrackContext * context, + const char *id) +{ + g_assert (context && id); + if (context->codec_id) + g_free (context->codec_id); + context->codec_id = g_strdup (id); +} /** * gst_matroska_mux_video_pad_setcaps: @@ -961,7 +970,8 @@ skip_details: /* find type */ if (!strcmp (mimetype, "video/x-raw-yuv")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_UNCOMPRESSED); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_UNCOMPRESSED); gst_structure_get_fourcc (structure, "format", &videocontext->fourcc); } else if (!strcmp (mimetype, "video/x-xvid") /* MS/VfW compatibility cases */ ||!strcmp (mimetype, "video/x-huffyuv") @@ -1053,11 +1063,13 @@ skip_details: GST_BUFFER_DATA (codec_buf), GST_BUFFER_SIZE (codec_buf)); } - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_VFW_FOURCC); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_VFW_FOURCC); context->codec_priv = (gpointer) bih; context->codec_priv_size = size; } else if (!strcmp (mimetype, "video/x-h264")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_AVC); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_AVC); if (context->codec_priv != NULL) { g_free (context->codec_priv); @@ -1075,7 +1087,7 @@ skip_details: } else if (!strcmp (mimetype, "video/x-theora")) { const GValue *streamheader; - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_THEORA); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_THEORA); if (context->codec_priv != NULL) { g_free (context->codec_priv); @@ -1090,22 +1102,25 @@ skip_details: goto refuse_caps; } } else if (!strcmp (mimetype, "video/x-dirac")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_DIRAC); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_DIRAC); } else if (!strcmp (mimetype, "video/x-vp8")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_VP8); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_VIDEO_VP8); } else if (!strcmp (mimetype, "video/mpeg")) { gint mpegversion; gst_structure_get_int (structure, "mpegversion", &mpegversion); switch (mpegversion) { case 1: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MPEG1); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_MPEG1); break; case 2: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MPEG2); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_MPEG2); break; case 4: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_ASP); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_MPEG4_ASP); break; default: goto refuse_caps; @@ -1129,16 +1144,20 @@ skip_details: gst_structure_get_int (structure, "rmversion", &rmversion); switch (rmversion) { case 1: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO1); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO1); break; case 2: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO2); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO2); break; case 3: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO3); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO3); break; case 4: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO4); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_VIDEO_REALVIDEO4); break; default: goto refuse_caps; @@ -1634,13 +1653,16 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) switch (layer) { case 1: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L1); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L1); break; case 2: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L2); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L2); break; case 3: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L3); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_MPEG1_L3); break; default: goto refuse_caps; @@ -1711,9 +1733,11 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) audiocontext->bitdepth = depth; if (endianness == G_BIG_ENDIAN) - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_PCM_INT_BE); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_PCM_INT_BE); else - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_PCM_INT_LE); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_PCM_INT_LE); } else if (!strcmp (mimetype, "audio/x-raw-float")) { gint width; @@ -1724,12 +1748,13 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) } audiocontext->bitdepth = width; - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_PCM_FLOAT); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_PCM_FLOAT); } else if (!strcmp (mimetype, "audio/x-vorbis")) { const GValue *streamheader; - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_VORBIS); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_VORBIS); if (context->codec_priv != NULL) { g_free (context->codec_priv); @@ -1746,7 +1771,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) } else if (!strcmp (mimetype, "audio/x-flac")) { const GValue *streamheader; - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_FLAC); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_FLAC); if (context->codec_priv != NULL) { g_free (context->codec_priv); context->codec_priv = NULL; @@ -1762,7 +1787,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) } else if (!strcmp (mimetype, "audio/x-speex")) { const GValue *streamheader; - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_SPEEX); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_SPEEX); if (context->codec_priv != NULL) { g_free (context->codec_priv); context->codec_priv = NULL; @@ -1776,11 +1801,11 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) goto refuse_caps; } } else if (!strcmp (mimetype, "audio/x-ac3")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_AC3); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_AC3); } else if (!strcmp (mimetype, "audio/x-eac3")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_EAC3); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_EAC3); } else if (!strcmp (mimetype, "audio/x-dts")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_DTS); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_DTS); } else if (!strcmp (mimetype, "audio/x-tta")) { gint width; @@ -1789,7 +1814,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) gst_structure_get_int (structure, "width", &width); audiocontext->bitdepth = width; - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_TTA); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_TTA); } else if (!strcmp (mimetype, "audio/x-pn-realaudio")) { gint raversion; @@ -1798,13 +1823,16 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) gst_structure_get_int (structure, "raversion", &raversion); switch (raversion) { case 1: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_REAL_14_4); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_REAL_14_4); break; case 2: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_REAL_28_8); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_REAL_28_8); break; case 8: - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_REAL_COOK); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_AUDIO_REAL_COOK); break; default: goto refuse_caps; @@ -1905,7 +1933,7 @@ gst_matroska_mux_audio_pad_setcaps (GstPad * pad, GstCaps * caps) GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf)); } - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_AUDIO_ACM); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_AUDIO_ACM); context->codec_priv = (gpointer) codec_priv; context->codec_priv_size = codec_priv_size; } @@ -1977,7 +2005,8 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps) if (!strcmp (mimetype, "subtitle/x-kate")) { const GValue *streamheader; - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_KATE); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_SUBTITLE_KATE); if (context->codec_priv != NULL) { g_free (context->codec_priv); @@ -1993,15 +2022,17 @@ gst_matroska_mux_subtitle_pad_setcaps (GstPad * pad, GstCaps * caps) goto exit; } } else if (!strcmp (mimetype, "text/plain")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_UTF8); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_SUBTITLE_UTF8); } else if (!strcmp (mimetype, "application/x-ssa")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_SSA); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_SUBTITLE_SSA); } else if (!strcmp (mimetype, "application/x-ass")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_ASS); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_SUBTITLE_ASS); } else if (!strcmp (mimetype, "application/x-usf")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_USF); + gst_matroska_mux_set_codec_id (context, GST_MATROSKA_CODEC_ID_SUBTITLE_USF); } else if (!strcmp (mimetype, "video/x-dvd-subpicture")) { - context->codec_id = g_strdup (GST_MATROSKA_CODEC_ID_SUBTITLE_VOBSUB); + gst_matroska_mux_set_codec_id (context, + GST_MATROSKA_CODEC_ID_SUBTITLE_VOBSUB); } else { id = NULL; ret = FALSE;