diff --git a/configure.ac b/configure.ac index 14b23879c0..7bad161b42 100644 --- a/configure.ac +++ b/configure.ac @@ -210,7 +210,7 @@ dnl *** plug-ins to include *** dnl Non ported plugins (non-dependant, then dependant) dnl Make sure you have a space before and after all plugins -GST_PLUGINS_NONPORTED=" dvdlpcmdec dvdsub iec958 mpegaudioparse mpegstream \ +GST_PLUGINS_NONPORTED=" dvdlpcmdec dvdsub iec958 xingmux mpegstream \ realmedia \ a52dec amrnb amrwb cdio dvdread lame mpeg2dec twolame x264 " AC_SUBST(GST_PLUGINS_NONPORTED) @@ -220,7 +220,7 @@ AG_GST_CHECK_PLUGIN(asfdemux) AG_GST_CHECK_PLUGIN(dvdlpcmdec) AG_GST_CHECK_PLUGIN(dvdsub) AG_GST_CHECK_PLUGIN(iec958) -AG_GST_CHECK_PLUGIN(mpegaudioparse) +AG_GST_CHECK_PLUGIN(xingmux) AG_GST_CHECK_PLUGIN(mpegstream) AG_GST_CHECK_PLUGIN(realmedia) AG_GST_CHECK_PLUGIN(synaesthesia) @@ -467,10 +467,10 @@ gst/asfdemux/Makefile gst/dvdlpcmdec/Makefile gst/dvdsub/Makefile gst/iec958/Makefile -gst/mpegaudioparse/Makefile gst/mpegstream/Makefile gst/realmedia/Makefile gst/synaesthesia/Makefile +gst/xingmux/Makefile ext/Makefile ext/a52dec/Makefile ext/amrnb/Makefile diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index ea7fb3a762..838459cf91 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -90,8 +90,7 @@ EXTRA_HFILES = \ $(top_srcdir)/ext/twolame/gsttwolame.h \ $(top_srcdir)/ext/x264/gstx264enc.h \ $(top_srcdir)/gst/asfdemux/gstrtspwms.h \ - $(top_srcdir)/gst/mpegaudioparse/gstmpegaudioparse.h \ - $(top_srcdir)/gst/mpegaudioparse/gstxingmux.h \ + $(top_srcdir)/gst/xingmux/gstxingmux.h \ $(top_srcdir)/gst/realmedia/rademux.h \ $(top_srcdir)/gst/realmedia/rdtmanager.h \ $(top_srcdir)/gst/realmedia/rmdemux.h \ diff --git a/docs/plugins/gst-plugins-ugly-plugins-docs.sgml b/docs/plugins/gst-plugins-ugly-plugins-docs.sgml index c5ce3211a1..2bdf11a0ea 100644 --- a/docs/plugins/gst-plugins-ugly-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-ugly-plugins-docs.sgml @@ -25,7 +25,6 @@ - @@ -52,7 +51,6 @@ - diff --git a/docs/plugins/gst-plugins-ugly-plugins-sections.txt b/docs/plugins/gst-plugins-ugly-plugins-sections.txt index 55f5650252..e15dde4a59 100644 --- a/docs/plugins/gst-plugins-ugly-plugins-sections.txt +++ b/docs/plugins/gst-plugins-ugly-plugins-sections.txt @@ -121,22 +121,6 @@ GST_TYPE_MAD gst_mad_get_type -
-element-mp3parse -mp3parse -GstMPEGAudioParse - -GstMPEGAudioParseClass -MPEGAudioPendingAccurateSeek -MPEGAudioSeekEntry -GST_MP3PARSE -GST_MP3PARSE_CLASS -GST_IS_MP3PARSE -GST_IS_MP3PARSE_CLASS -GST_TYPE_MP3PARSE -gst_mp3parse_get_type -
-
element-rademux rademux diff --git a/docs/plugins/inspect/plugin-mpegaudioparse.xml b/docs/plugins/inspect/plugin-mpegaudioparse.xml deleted file mode 100644 index c5a78c23b1..0000000000 --- a/docs/plugins/inspect/plugin-mpegaudioparse.xml +++ /dev/null @@ -1,55 +0,0 @@ - - mpegaudioparse - MPEG-1 layer 1/2/3 audio stream elements - ../../gst/mpegaudioparse/.libs/libgstmpegaudioparse.so - libgstmpegaudioparse.so - 0.10.18.1 - LGPL - gst-plugins-ugly - GStreamer Ugly Plug-ins git - Unknown package origin - - - mp3parse - MPEG1 Audio Parser - Codec/Parser/Audio - Parses and frames mpeg1 audio streams (levels 1-3), provides seek - Jan Schmidt <thaytan@mad.scientist.com>,Erik Walthinsen <omega@cse.ogi.edu> - - - sink - sink - always -
audio/mpeg, mpegversion=(int)1, parsed=(boolean)false
- - - src - source - always -
audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], rate=(int)[ 8000, 48000 ], channels=(int)[ 1, 2 ], parsed=(boolean)true
- - - - - xingmux - MP3 Xing muxer - Formatter/Metadata - Adds a Xing header to the beginning of a VBR MP3 file - Christophe Fergeau <teuf@gnome.org> - - - sink - sink - always -
audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ]
- - - src - source - always -
audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ]
- - - - - \ No newline at end of file diff --git a/gst-plugins-ugly.spec.in b/gst-plugins-ugly.spec.in index 6c3c302810..0b990cc84a 100644 --- a/gst-plugins-ugly.spec.in +++ b/gst-plugins-ugly.spec.in @@ -96,7 +96,7 @@ rm -rf $RPM_BUILD_ROOT %{_libdir}/gstreamer-%{majorminor}/libgstasf.so %{_libdir}/gstreamer-%{majorminor}/libgstdvdlpcmdec.so %{_libdir}/gstreamer-%{majorminor}/libgstiec958.so -%{_libdir}/gstreamer-%{majorminor}/libgstmpegaudioparse.so +%{_libdir}/gstreamer-%{majorminor}/libgstxingmux.so %{_libdir}/gstreamer-%{majorminor}/libgstmpegstream.so %{_libdir}/gstreamer-%{majorminor}/libgstrmdemux.so %{_libdir}/gstreamer-%{majorminor}/libgstdvdsub.so diff --git a/gst/mpegaudioparse/Makefile.am b/gst/mpegaudioparse/Makefile.am deleted file mode 100644 index 61a00bdedd..0000000000 --- a/gst/mpegaudioparse/Makefile.am +++ /dev/null @@ -1,23 +0,0 @@ -plugin_LTLIBRARIES = libgstmpegaudioparse.la - -libgstmpegaudioparse_la_SOURCES = plugin.c gstmpegaudioparse.c gstxingmux.c -libgstmpegaudioparse_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) -libgstmpegaudioparse_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) -libgstmpegaudioparse_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) -libgstmpegaudioparse_la_LIBTOOLFLAGS = --tag=disable-static - -noinst_HEADERS = gstmpegaudioparse.h gstxingmux.h - -Android.mk: Makefile.am $(BUILT_SOURCES) - androgenizer \ - -:PROJECT libgstmpegaudioparse -:SHARED libgstmpegaudioparse \ - -:TAGS eng debug \ - -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \ - -:SOURCES $(libgstmpegaudioparse_la_SOURCES) \ - -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstmpegaudioparse_la_CFLAGS) \ - -:LDFLAGS $(libgstmpegaudioparse_la_LDFLAGS) \ - $(libgstmpegaudioparse_la_LIBADD) \ - -ldl \ - -:PASSTHROUGH LOCAL_ARM_MODE:=arm \ - LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \ - > $@ diff --git a/gst/mpegaudioparse/gstmpegaudioparse.c b/gst/mpegaudioparse/gstmpegaudioparse.c deleted file mode 100644 index 9ebc20e35e..0000000000 --- a/gst/mpegaudioparse/gstmpegaudioparse.c +++ /dev/null @@ -1,2203 +0,0 @@ -/* GStreamer - * Copyright (C) <1999> Erik Walthinsen - * Copyright (C) <2006-2007> Jan Schmidt - * - * 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. - */ - -/** - * SECTION:element-mp3parse - * - * Parses and frames mpeg1 audio streams. Provides seeking. - * - * - * Example launch line - * |[ - * gst-launch filesrc location=test.mp3 ! mp3parse ! mad ! autoaudiosink - * ]| - * - */ - - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include - -#include "gstmpegaudioparse.h" - -GST_DEBUG_CATEGORY_STATIC (mp3parse_debug); -#define GST_CAT_DEFAULT mp3parse_debug - -#define MP3_CHANNEL_MODE_UNKNOWN -1 -#define MP3_CHANNEL_MODE_STEREO 0 -#define MP3_CHANNEL_MODE_JOINT_STEREO 1 -#define MP3_CHANNEL_MODE_DUAL_CHANNEL 2 -#define MP3_CHANNEL_MODE_MONO 3 - -#define CRC_UNKNOWN -1 -#define CRC_PROTECTED 0 -#define CRC_NOT_PROTECTED 1 - -#define XING_FRAMES_FLAG 0x0001 -#define XING_BYTES_FLAG 0x0002 -#define XING_TOC_FLAG 0x0004 -#define XING_VBR_SCALE_FLAG 0x0008 - -#ifndef GST_READ_UINT24_BE -#define GST_READ_UINT24_BE(p) (p[2] | (p[1] << 8) | (p[0] << 16)) -#endif - -/* Minimum number of consecutive, valid-looking frames to consider - for resyncing */ -#define MIN_RESYNC_FRAMES 3 - -static inline MPEGAudioSeekEntry * -mpeg_audio_seek_entry_new (void) -{ - return g_slice_new (MPEGAudioSeekEntry); -} - -static inline void -mpeg_audio_seek_entry_free (MPEGAudioSeekEntry * entry) -{ - g_slice_free (MPEGAudioSeekEntry, entry); -} - -static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/mpeg, " - "mpegversion = (int) 1, " - "layer = (int) [ 1, 3 ], " - "rate = (int) [ 8000, 48000 ], channels = (int) [ 1, 2 ]," - "parsed=(boolean) true") - ); - -static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1, parsed=(boolean)false") - ); - -/* GstMPEGAudioParse signals and args */ -enum -{ - /* FILL ME */ - LAST_SIGNAL -}; - -enum -{ - ARG_0, - ARG_SKIP, - ARG_BIT_RATE - /* FILL ME */ -}; - - -static gboolean gst_mp3parse_sink_event (GstPad * pad, GstEvent * event); -static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer); -static gboolean mp3parse_src_query (GstPad * pad, GstQuery * query); -static const GstQueryType *mp3parse_get_query_types (GstPad * pad); -static gboolean mp3parse_src_event (GstPad * pad, GstEvent * event); - -static int head_check (GstMPEGAudioParse * mp3parse, unsigned long head); - -static void gst_mp3parse_dispose (GObject * object); -static void gst_mp3parse_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_mp3parse_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static GstStateChangeReturn gst_mp3parse_change_state (GstElement * element, - GstStateChange transition); -static GstFlowReturn -gst_mp3parse_handle_data (GstMPEGAudioParse * mp3parse, gboolean at_eos); - -static gboolean mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse, - gint64 bytepos, GstClockTime * ts, gboolean from_total_time); -static gboolean -mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total); -static gboolean -mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total); - -GST_BOILERPLATE (GstMPEGAudioParse, gst_mp3parse, GstElement, GST_TYPE_ELEMENT); - -#define GST_TYPE_MP3_CHANNEL_MODE (gst_mp3_channel_mode_get_type()) - -static const GEnumValue mp3_channel_mode[] = { - {MP3_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"}, - {MP3_CHANNEL_MODE_MONO, "Mono", "mono"}, - {MP3_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"}, - {MP3_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"}, - {MP3_CHANNEL_MODE_STEREO, "Stereo", "stereo"}, - {0, NULL, NULL}, -}; - -static GType -gst_mp3_channel_mode_get_type (void) -{ - static GType mp3_channel_mode_type = 0; - - if (!mp3_channel_mode_type) { - mp3_channel_mode_type = - g_enum_register_static ("GstMp3ChannelMode", mp3_channel_mode); - } - return mp3_channel_mode_type; -} - -static const gchar * -gst_mp3_channel_mode_get_nick (gint mode) -{ - guint i; - for (i = 0; i < G_N_ELEMENTS (mp3_channel_mode); i++) { - if (mp3_channel_mode[i].value == mode) - return mp3_channel_mode[i].value_nick; - } - return NULL; -} - -static const guint mp3types_bitrates[2][3][16] = { - { - {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,}, - {0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,}, - {0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,} - }, - { - {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,}, - {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}, - {0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,} - }, -}; - -static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000}, -{22050, 24000, 16000}, -{11025, 12000, 8000} -}; - -static inline guint -mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header, - guint * put_version, guint * put_layer, guint * put_channels, - guint * put_bitrate, guint * put_samplerate, guint * put_mode, - guint * put_crc) -{ - guint length; - gulong mode, samplerate, bitrate, layer, channels, padding, crc; - gulong version; - gint lsf, mpg25; - - if (header & (1 << 20)) { - lsf = (header & (1 << 19)) ? 0 : 1; - mpg25 = 0; - } else { - lsf = 1; - mpg25 = 1; - } - - version = 1 + lsf + mpg25; - - layer = 4 - ((header >> 17) & 0x3); - - crc = (header >> 16) & 0x1; - - bitrate = (header >> 12) & 0xF; - bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000; - /* The caller has ensured we have a valid header, so bitrate can't be - zero here. */ - g_assert (bitrate != 0); - - samplerate = (header >> 10) & 0x3; - samplerate = mp3types_freqs[lsf + mpg25][samplerate]; - - padding = (header >> 9) & 0x1; - - mode = (header >> 6) & 0x3; - channels = (mode == 3) ? 1 : 2; - - switch (layer) { - case 1: - length = 4 * ((bitrate * 12) / samplerate + padding); - break; - case 2: - length = (bitrate * 144) / samplerate + padding; - break; - default: - case 3: - length = (bitrate * 144) / (samplerate << lsf) + padding; - break; - } - - GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes", - length); - GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, version = %lu, " - "layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version, - layer, channels, gst_mp3_channel_mode_get_nick (mode)); - - if (put_version) - *put_version = version; - if (put_layer) - *put_layer = layer; - if (put_channels) - *put_channels = channels; - if (put_bitrate) - *put_bitrate = bitrate; - if (put_samplerate) - *put_samplerate = samplerate; - if (put_mode) - *put_mode = mode; - if (put_crc) - *put_crc = crc; - - return length; -} - -static GstCaps * -mp3_caps_create (guint version, guint layer, guint channels, guint samplerate) -{ - GstCaps *new; - - g_assert (version); - g_assert (layer); - g_assert (samplerate); - g_assert (channels); - - new = gst_caps_new_simple ("audio/mpeg", - "mpegversion", G_TYPE_INT, 1, - "mpegaudioversion", G_TYPE_INT, version, - "layer", G_TYPE_INT, layer, - "rate", G_TYPE_INT, samplerate, - "channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL); - - return new; -} - -static void -gst_mp3parse_base_init (gpointer klass) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (klass); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&mp3_sink_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&mp3_src_template)); - - GST_DEBUG_CATEGORY_INIT (mp3parse_debug, "mp3parse", 0, "MPEG Audio Parser"); - - gst_element_class_set_details_simple (element_class, "MPEG1 Audio Parser", - "Codec/Parser/Audio", - "Parses and frames mpeg1 audio streams (levels 1-3), provides seek", - "Jan Schmidt ," - "Erik Walthinsen "); -} - -static void -gst_mp3parse_class_init (GstMPEGAudioParseClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *gstelement_class; - - gobject_class = (GObjectClass *) klass; - gstelement_class = (GstElementClass *) klass; - - parent_class = g_type_class_peek_parent (klass); - - gobject_class->set_property = gst_mp3parse_set_property; - gobject_class->get_property = gst_mp3parse_get_property; - gobject_class->dispose = gst_mp3parse_dispose; - - g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP, - g_param_spec_int ("skip", "skip", "skip", - G_MININT, G_MAXINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE, - g_param_spec_int ("bitrate", "Bitrate", "Bit Rate", - G_MININT, G_MAXINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); - - gstelement_class->change_state = gst_mp3parse_change_state; - -/* register tags */ -#define GST_TAG_CRC "has-crc" -#define GST_TAG_MODE "channel-mode" - - gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN, - "has crc", "Using CRC", NULL); - gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING, - "channel mode", "MPEG audio channel mode", NULL); - - g_type_class_ref (GST_TYPE_MP3_CHANNEL_MODE); -} - -static void -gst_mp3parse_reset (GstMPEGAudioParse * mp3parse) -{ - mp3parse->skip = 0; - mp3parse->resyncing = TRUE; - mp3parse->next_ts = GST_CLOCK_TIME_NONE; - mp3parse->cur_offset = -1; - - mp3parse->sync_offset = 0; - mp3parse->tracked_offset = 0; - mp3parse->pending_ts = GST_CLOCK_TIME_NONE; - mp3parse->pending_offset = -1; - - gst_adapter_clear (mp3parse->adapter); - - mp3parse->rate = mp3parse->channels = mp3parse->layer = -1; - mp3parse->version = 1; - mp3parse->max_bitreservoir = GST_CLOCK_TIME_NONE; - - mp3parse->avg_bitrate = 0; - mp3parse->bitrate_sum = 0; - mp3parse->last_posted_bitrate = 0; - mp3parse->frame_count = 0; - mp3parse->sent_codec_tag = FALSE; - - mp3parse->last_posted_crc = CRC_UNKNOWN; - mp3parse->last_posted_channel_mode = MP3_CHANNEL_MODE_UNKNOWN; - - mp3parse->xing_flags = 0; - mp3parse->xing_bitrate = 0; - mp3parse->xing_frames = 0; - mp3parse->xing_total_time = 0; - mp3parse->xing_bytes = 0; - mp3parse->xing_vbr_scale = 0; - memset (mp3parse->xing_seek_table, 0, 100); - memset (mp3parse->xing_seek_table_inverse, 0, 256); - - mp3parse->vbri_bitrate = 0; - mp3parse->vbri_frames = 0; - mp3parse->vbri_total_time = 0; - mp3parse->vbri_bytes = 0; - mp3parse->vbri_seek_points = 0; - g_free (mp3parse->vbri_seek_table); - mp3parse->vbri_seek_table = NULL; - - if (mp3parse->seek_table) { - g_list_foreach (mp3parse->seek_table, (GFunc) mpeg_audio_seek_entry_free, - NULL); - g_list_free (mp3parse->seek_table); - mp3parse->seek_table = NULL; - } - - g_mutex_lock (mp3parse->pending_seeks_lock); - if (mp3parse->pending_accurate_seeks) { - g_slist_foreach (mp3parse->pending_accurate_seeks, (GFunc) g_free, NULL); - g_slist_free (mp3parse->pending_accurate_seeks); - mp3parse->pending_accurate_seeks = NULL; - } - if (mp3parse->pending_nonaccurate_seeks) { - g_slist_foreach (mp3parse->pending_nonaccurate_seeks, (GFunc) g_free, NULL); - g_slist_free (mp3parse->pending_nonaccurate_seeks); - mp3parse->pending_nonaccurate_seeks = NULL; - } - g_mutex_unlock (mp3parse->pending_seeks_lock); - - if (mp3parse->pending_segment) { - GstEvent **eventp = &mp3parse->pending_segment; - - gst_event_replace (eventp, NULL); - } - - mp3parse->exact_position = FALSE; - gst_segment_init (&mp3parse->segment, GST_FORMAT_TIME); -} - -static void -gst_mp3parse_init (GstMPEGAudioParse * mp3parse, GstMPEGAudioParseClass * klass) -{ - mp3parse->sinkpad = - gst_pad_new_from_static_template (&mp3_sink_template, "sink"); - gst_pad_set_event_function (mp3parse->sinkpad, gst_mp3parse_sink_event); - gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain); - gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad); - - mp3parse->srcpad = - gst_pad_new_from_static_template (&mp3_src_template, "src"); - gst_pad_use_fixed_caps (mp3parse->srcpad); - gst_pad_set_event_function (mp3parse->srcpad, mp3parse_src_event); - gst_pad_set_query_function (mp3parse->srcpad, mp3parse_src_query); - gst_pad_set_query_type_function (mp3parse->srcpad, mp3parse_get_query_types); - gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad); - - mp3parse->adapter = gst_adapter_new (); - mp3parse->pending_seeks_lock = g_mutex_new (); - - gst_mp3parse_reset (mp3parse); -} - -static void -gst_mp3parse_dispose (GObject * object) -{ - GstMPEGAudioParse *mp3parse = GST_MP3PARSE (object); - - gst_mp3parse_reset (mp3parse); - - if (mp3parse->adapter) { - g_object_unref (mp3parse->adapter); - mp3parse->adapter = NULL; - } - g_mutex_free (mp3parse->pending_seeks_lock); - mp3parse->pending_seeks_lock = NULL; - - g_list_foreach (mp3parse->pending_events, (GFunc) gst_mini_object_unref, - NULL); - g_list_free (mp3parse->pending_events); - mp3parse->pending_events = NULL; - - G_OBJECT_CLASS (parent_class)->dispose (object); -} - -static gboolean -gst_mp3parse_sink_event (GstPad * pad, GstEvent * event) -{ - gboolean res = TRUE; - GstMPEGAudioParse *mp3parse; - GstEvent **eventp; - - mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_NEWSEGMENT: - { - gdouble rate, applied_rate; - GstFormat format; - gint64 start, stop, pos; - gboolean update; - MPEGAudioPendingAccurateSeek *seek = NULL; - GSList *node; - - gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate, - &format, &start, &stop, &pos); - - g_mutex_lock (mp3parse->pending_seeks_lock); - if (format == GST_FORMAT_BYTES && mp3parse->pending_accurate_seeks) { - - for (node = mp3parse->pending_accurate_seeks; node; node = node->next) { - MPEGAudioPendingAccurateSeek *tmp = node->data; - - if (tmp->upstream_start == pos) { - seek = tmp; - break; - } - } - if (seek) { - GstSegment *s = &seek->segment; - - event = - gst_event_new_new_segment_full (FALSE, s->rate, s->applied_rate, - GST_FORMAT_TIME, s->start, s->stop, s->last_stop); - - mp3parse->segment = seek->segment; - - mp3parse->resyncing = FALSE; - mp3parse->cur_offset = pos; - mp3parse->next_ts = seek->timestamp_start; - mp3parse->pending_ts = GST_CLOCK_TIME_NONE; - mp3parse->tracked_offset = 0; - mp3parse->sync_offset = 0; - - gst_event_parse_new_segment_full (event, &update, &rate, - &applied_rate, &format, &start, &stop, &pos); - - GST_DEBUG_OBJECT (mp3parse, - "Pushing accurate newseg rate %g, applied rate %g, " - "format %d, start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT - ", pos %" G_GINT64_FORMAT, rate, applied_rate, format, start, - stop, pos); - - g_free (seek); - mp3parse->pending_accurate_seeks = - g_slist_delete_link (mp3parse->pending_accurate_seeks, node); - - g_mutex_unlock (mp3parse->pending_seeks_lock); - res = gst_pad_push_event (mp3parse->srcpad, event); - - return res; - } else { - GST_WARNING_OBJECT (mp3parse, - "Accurate seek not possible, didn't get an appropiate upstream segment"); - } - } - g_mutex_unlock (mp3parse->pending_seeks_lock); - - mp3parse->exact_position = FALSE; - - if (format == GST_FORMAT_BYTES) { - GstClockTime seg_start, seg_stop, seg_pos; - - /* stop time is allowed to be open-ended, but not start & pos */ - if (!mp3parse_bytepos_to_time (mp3parse, stop, &seg_stop, FALSE)) - seg_stop = GST_CLOCK_TIME_NONE; - if (mp3parse_bytepos_to_time (mp3parse, start, &seg_start, FALSE) && - mp3parse_bytepos_to_time (mp3parse, pos, &seg_pos, FALSE)) { - gst_event_unref (event); - - /* search the pending nonaccurate seeks */ - g_mutex_lock (mp3parse->pending_seeks_lock); - seek = NULL; - for (node = mp3parse->pending_nonaccurate_seeks; node; - node = node->next) { - MPEGAudioPendingAccurateSeek *tmp = node->data; - - if (tmp->upstream_start == pos) { - seek = tmp; - break; - } - } - - if (seek) { - if (seek->segment.stop == -1) { - /* corrent the segment end, because non-accurate seeks might make - * our streaming end earlier (see bug #603695) */ - seg_stop = -1; - } - g_free (seek); - mp3parse->pending_nonaccurate_seeks = - g_slist_delete_link (mp3parse->pending_nonaccurate_seeks, node); - } - g_mutex_unlock (mp3parse->pending_seeks_lock); - - event = gst_event_new_new_segment_full (update, rate, applied_rate, - GST_FORMAT_TIME, seg_start, seg_stop, seg_pos); - format = GST_FORMAT_TIME; - GST_DEBUG_OBJECT (mp3parse, "Converted incoming segment to TIME. " - "start = %" GST_TIME_FORMAT ", stop = %" GST_TIME_FORMAT - ", pos = %" GST_TIME_FORMAT, GST_TIME_ARGS (seg_start), - GST_TIME_ARGS (seg_stop), GST_TIME_ARGS (seg_pos)); - } - } - - if (format != GST_FORMAT_TIME) { - /* Unknown incoming segment format. Output a default open-ended - * TIME segment */ - gst_event_unref (event); - event = gst_event_new_new_segment_full (update, rate, applied_rate, - GST_FORMAT_TIME, 0, GST_CLOCK_TIME_NONE, 0); - } - - mp3parse->resyncing = TRUE; - mp3parse->cur_offset = -1; - mp3parse->next_ts = GST_CLOCK_TIME_NONE; - mp3parse->pending_ts = GST_CLOCK_TIME_NONE; - mp3parse->tracked_offset = 0; - mp3parse->sync_offset = 0; - /* also clear leftover data if clearing so much state */ - gst_adapter_clear (mp3parse->adapter); - - gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate, - &format, &start, &stop, &pos); - GST_DEBUG_OBJECT (mp3parse, "Pushing newseg rate %g, applied rate %g, " - "format %d, start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT - ", pos %" G_GINT64_FORMAT, rate, applied_rate, format, start, stop, - pos); - - gst_segment_set_newsegment_full (&mp3parse->segment, update, rate, - applied_rate, format, start, stop, pos); - - /* save the segment for later, right before we push a new buffer so that - * the caps are fixed and the next linked element can receive the segment. */ - eventp = &mp3parse->pending_segment; - gst_event_replace (eventp, event); - gst_event_unref (event); - res = TRUE; - break; - } - case GST_EVENT_FLUSH_STOP: - /* Clear our adapter and set up for a new position */ - gst_adapter_clear (mp3parse->adapter); - eventp = &mp3parse->pending_segment; - gst_event_replace (eventp, NULL); - res = gst_pad_push_event (mp3parse->srcpad, event); - break; - case GST_EVENT_EOS: - /* If we haven't processed any frames yet, then make sure we process - at least whatever's in our adapter */ - if (mp3parse->frame_count == 0) { - gst_mp3parse_handle_data (mp3parse, TRUE); - - /* If we STILL have zero frames processed, fire an error */ - if (mp3parse->frame_count == 0) { - GST_ELEMENT_ERROR (mp3parse, STREAM, WRONG_TYPE, - ("No valid frames found before end of stream"), (NULL)); - } - } - /* fall through */ - default: - if (mp3parse->pending_segment && - (GST_EVENT_TYPE (event) != GST_EVENT_EOS) && - (GST_EVENT_TYPE (event) != GST_EVENT_FLUSH_START)) { - /* Cache all events except EOS and the ones above if we have - * a pending segment */ - mp3parse->pending_events = - g_list_append (mp3parse->pending_events, event); - } else { - res = gst_pad_push_event (mp3parse->srcpad, event); - } - break; - } - - gst_object_unref (mp3parse); - - return res; -} - -static void -gst_mp3parse_add_index_entry (GstMPEGAudioParse * mp3parse, guint64 offset, - GstClockTime ts) -{ - MPEGAudioSeekEntry *entry, *last; - - if (G_LIKELY (mp3parse->seek_table != NULL)) { - last = mp3parse->seek_table->data; - - if (last->byte >= offset) - return; - - if (GST_CLOCK_DIFF (last->timestamp, ts) < mp3parse->idx_interval) - return; - } - - entry = mpeg_audio_seek_entry_new (); - entry->byte = offset; - entry->timestamp = ts; - mp3parse->seek_table = g_list_prepend (mp3parse->seek_table, entry); - - GST_LOG_OBJECT (mp3parse, "Adding index entry %" GST_TIME_FORMAT " @ offset " - "0x%08" G_GINT64_MODIFIER "x", GST_TIME_ARGS (ts), offset); -} - -/* Prepare a buffer of the indicated size, timestamp it and output */ -static GstFlowReturn -gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size, - guint mode, guint crc) -{ - GstBuffer *outbuf; - guint bitrate; - GstFlowReturn ret = GST_FLOW_OK; - GstClockTime push_start; - GstTagList *taglist; - - outbuf = gst_adapter_take_buffer (mp3parse->adapter, size); - - GST_BUFFER_DURATION (outbuf) = - gst_util_uint64_scale (GST_SECOND, mp3parse->spf, mp3parse->rate); - - GST_BUFFER_OFFSET (outbuf) = mp3parse->cur_offset; - - /* Check if we have a pending timestamp from an incoming buffer to apply - * here */ - if (GST_CLOCK_TIME_IS_VALID (mp3parse->pending_ts)) { - if (mp3parse->tracked_offset >= mp3parse->pending_offset) { - /* If the incoming timestamp differs from our expected by more than - * half a frame, then take it instead of our calculated timestamp. - * This avoids creating imperfect streams just because of - * quantization in the container timestamping */ - GstClockTimeDiff diff = mp3parse->next_ts - mp3parse->pending_ts; - GstClockTimeDiff thresh = GST_BUFFER_DURATION (outbuf) / 2; - - if (diff < -thresh || diff > thresh) { - GST_DEBUG_OBJECT (mp3parse, "Updating next_ts from %" GST_TIME_FORMAT - " to pending ts %" GST_TIME_FORMAT - " at offset %" G_GINT64_FORMAT " (pending offset was %" - G_GINT64_FORMAT ")", GST_TIME_ARGS (mp3parse->next_ts), - GST_TIME_ARGS (mp3parse->pending_ts), mp3parse->tracked_offset, - mp3parse->pending_offset); - mp3parse->next_ts = mp3parse->pending_ts; - } - mp3parse->pending_ts = GST_CLOCK_TIME_NONE; - } - } - - /* Decide what timestamp we're going to apply */ - if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts)) { - GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->next_ts; - } else { - GstClockTime ts; - - /* No timestamp yet, convert our offset to a timestamp if we can, or - * start at 0 */ - if (mp3parse_bytepos_to_time (mp3parse, mp3parse->cur_offset, &ts, FALSE) && - GST_CLOCK_TIME_IS_VALID (ts)) - GST_BUFFER_TIMESTAMP (outbuf) = ts; - else { - GST_BUFFER_TIMESTAMP (outbuf) = 0; - } - } - - if (GST_BUFFER_TIMESTAMP (outbuf) == 0) - mp3parse->exact_position = TRUE; - - if (mp3parse->seekable && - mp3parse->exact_position && GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) && - mp3parse->cur_offset != GST_BUFFER_OFFSET_NONE) { - gst_mp3parse_add_index_entry (mp3parse, mp3parse->cur_offset, - GST_BUFFER_TIMESTAMP (outbuf)); - } - - /* Update our byte offset tracking */ - if (mp3parse->cur_offset != -1) { - mp3parse->cur_offset += size; - } - mp3parse->tracked_offset += size; - - if (GST_BUFFER_TIMESTAMP_IS_VALID (outbuf)) - mp3parse->next_ts = - GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf); - - gst_buffer_set_caps (outbuf, GST_PAD_CAPS (mp3parse->srcpad)); - - /* Post a bitrate tag if we need to before pushing the buffer */ - if (mp3parse->xing_bitrate != 0) - bitrate = mp3parse->xing_bitrate; - else if (mp3parse->vbri_bitrate != 0) - bitrate = mp3parse->vbri_bitrate; - else - bitrate = mp3parse->avg_bitrate; - - /* we will create a taglist (if any of the parameters has changed) - * to add the tags that changed */ - taglist = NULL; - if ((mp3parse->last_posted_bitrate / 10000) != (bitrate / 10000)) { - taglist = gst_tag_list_new (); - mp3parse->last_posted_bitrate = bitrate; - gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE, - mp3parse->last_posted_bitrate, NULL); - - /* Post a new duration message if the average bitrate changes that much - * so applications can update their cached values - */ - if ((mp3parse->xing_flags & XING_TOC_FLAG) == 0 - && mp3parse->vbri_total_time == 0) { - gst_element_post_message (GST_ELEMENT (mp3parse), - gst_message_new_duration (GST_OBJECT (mp3parse), GST_FORMAT_TIME, - -1)); - } - } - - if (mp3parse->last_posted_crc != crc) { - gboolean using_crc; - - if (!taglist) { - taglist = gst_tag_list_new (); - } - mp3parse->last_posted_crc = crc; - if (mp3parse->last_posted_crc == CRC_PROTECTED) { - using_crc = TRUE; - } else { - using_crc = FALSE; - } - gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CRC, - using_crc, NULL); - } - - if (mp3parse->last_posted_channel_mode != mode) { - if (!taglist) { - taglist = gst_tag_list_new (); - } - mp3parse->last_posted_channel_mode = mode; - - gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE, - gst_mp3_channel_mode_get_nick (mode), NULL); - } - - /* if the taglist exists, we need to send it */ - if (taglist) { - gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse), - mp3parse->srcpad, taglist); - } - - /* We start pushing 9 frames earlier (29 frames for MPEG2) than - * segment start to be able to decode the first frame we want. - * 9 (29) frames are the theoretical maximum of frames that contain - * data for the current frame (bit reservoir). - */ - if (mp3parse->segment.start == 0) { - push_start = 0; - } else if (GST_CLOCK_TIME_IS_VALID (mp3parse->max_bitreservoir)) { - if (GST_CLOCK_TIME_IS_VALID (mp3parse->segment.start) && - mp3parse->segment.start > mp3parse->max_bitreservoir) - push_start = mp3parse->segment.start - mp3parse->max_bitreservoir; - else - push_start = 0; - } else { - push_start = mp3parse->segment.start; - } - - if (G_UNLIKELY ((GST_CLOCK_TIME_IS_VALID (push_start) && - GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) && - GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf) - < push_start))) { - GST_DEBUG_OBJECT (mp3parse, - "Buffer before configured segment range %" GST_TIME_FORMAT - " to %" GST_TIME_FORMAT ", dropping, timestamp %" - GST_TIME_FORMAT " duration %" GST_TIME_FORMAT - ", offset 0x%08" G_GINT64_MODIFIER "x", GST_TIME_ARGS (push_start), - GST_TIME_ARGS (mp3parse->segment.stop), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), - GST_BUFFER_OFFSET (outbuf)); - - gst_buffer_unref (outbuf); - ret = GST_FLOW_OK; - } else if (G_UNLIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) && - GST_CLOCK_TIME_IS_VALID (mp3parse->segment.stop) && - GST_BUFFER_TIMESTAMP (outbuf) >= - mp3parse->segment.stop + GST_BUFFER_DURATION (outbuf))) { - /* Some mp3 streams have an offset in the timestamps, for which we have to - * push the frame *after* the end position in order for the decoder to be - * able to decode everything up until the segment.stop position. - * That is the reason of the calculated offset */ - GST_DEBUG_OBJECT (mp3parse, - "Buffer after configured segment range %" GST_TIME_FORMAT " to %" - GST_TIME_FORMAT ", returning GST_FLOW_UNEXPECTED, timestamp %" - GST_TIME_FORMAT " duration %" GST_TIME_FORMAT ", offset 0x%08" - G_GINT64_MODIFIER "x", GST_TIME_ARGS (push_start), - GST_TIME_ARGS (mp3parse->segment.stop), - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)), - GST_BUFFER_OFFSET (outbuf)); - - gst_buffer_unref (outbuf); - ret = GST_FLOW_UNEXPECTED; - } else { - GST_DEBUG_OBJECT (mp3parse, - "pushing buffer of %d bytes, timestamp %" GST_TIME_FORMAT - ", offset 0x%08" G_GINT64_MODIFIER "x", size, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)), - GST_BUFFER_OFFSET (outbuf)); - mp3parse->segment.last_stop = GST_BUFFER_TIMESTAMP (outbuf); - /* push any pending segment now */ - if (mp3parse->pending_segment) { - gst_pad_push_event (mp3parse->srcpad, mp3parse->pending_segment); - mp3parse->pending_segment = NULL; - } - if (mp3parse->pending_events) { - GList *l; - - for (l = mp3parse->pending_events; l != NULL; l = l->next) { - gst_pad_push_event (mp3parse->srcpad, GST_EVENT (l->data)); - } - g_list_free (mp3parse->pending_events); - mp3parse->pending_events = NULL; - } - - /* set discont if needed */ - if (mp3parse->discont) { - GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); - mp3parse->discont = FALSE; - } - - ret = gst_pad_push (mp3parse->srcpad, outbuf); - } - - return ret; -} - -static void -gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse) -{ - GstTagList *taglist; - gchar *codec; - const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */ - const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */ - const guint32 vbri_id = 0x56425249; /* 'VBRI' in hex */ - - gint offset; - - guint64 avail; - gint64 upstream_total_bytes = 0; - guint32 read_id; - const guint8 *data; - - /* Output codec tag */ - if (!mp3parse->sent_codec_tag) { - if (mp3parse->layer == 3) { - codec = g_strdup_printf ("MPEG %d Audio, Layer %d (MP3)", - mp3parse->version, mp3parse->layer); - } else { - codec = g_strdup_printf ("MPEG %d Audio, Layer %d", - mp3parse->version, mp3parse->layer); - } - - taglist = gst_tag_list_new (); - gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, - GST_TAG_AUDIO_CODEC, codec, NULL); - gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse), - mp3parse->srcpad, taglist); - g_free (codec); - - mp3parse->sent_codec_tag = TRUE; - } - /* end setting the tag */ - - /* Check first frame for Xing info */ - if (mp3parse->version == 1) { /* MPEG-1 file */ - if (mp3parse->channels == 1) - offset = 0x11; - else - offset = 0x20; - } else { /* MPEG-2 header */ - if (mp3parse->channels == 1) - offset = 0x09; - else - offset = 0x11; - } - /* Skip the 4 bytes of the MP3 header too */ - offset += 4; - - /* Check if we have enough data to read the Xing header */ - avail = gst_adapter_available (mp3parse->adapter); - - if (avail < offset + 8) - return; - - data = gst_adapter_peek (mp3parse->adapter, offset + 8); - if (data == NULL) - return; - /* The header starts at the provided offset */ - data += offset; - - /* obtain real upstream total bytes */ - mp3parse_total_bytes (mp3parse, &upstream_total_bytes); - - read_id = GST_READ_UINT32_BE (data); - if (read_id == xing_id || read_id == info_id) { - guint32 xing_flags; - guint bytes_needed = offset + 8; - gint64 total_bytes; - GstClockTime total_time; - - GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id); - - /* Read 4 base bytes of flags, big-endian */ - xing_flags = GST_READ_UINT32_BE (data + 4); - if (xing_flags & XING_FRAMES_FLAG) - bytes_needed += 4; - if (xing_flags & XING_BYTES_FLAG) - bytes_needed += 4; - if (xing_flags & XING_TOC_FLAG) - bytes_needed += 100; - if (xing_flags & XING_VBR_SCALE_FLAG) - bytes_needed += 4; - if (avail < bytes_needed) { - GST_DEBUG_OBJECT (mp3parse, - "Not enough data to read Xing header (need %d)", bytes_needed); - return; - } - - GST_DEBUG_OBJECT (mp3parse, "Reading Xing header"); - mp3parse->xing_flags = xing_flags; - data = gst_adapter_peek (mp3parse->adapter, bytes_needed); - data += offset + 8; - - if (xing_flags & XING_FRAMES_FLAG) { - mp3parse->xing_frames = GST_READ_UINT32_BE (data); - if (mp3parse->xing_frames == 0) { - GST_WARNING_OBJECT (mp3parse, - "Invalid number of frames in Xing header"); - mp3parse->xing_flags &= ~XING_FRAMES_FLAG; - } else { - mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND, - (guint64) (mp3parse->xing_frames) * (mp3parse->spf), - mp3parse->rate); - } - - data += 4; - } else { - mp3parse->xing_frames = 0; - mp3parse->xing_total_time = 0; - } - - if (xing_flags & XING_BYTES_FLAG) { - mp3parse->xing_bytes = GST_READ_UINT32_BE (data); - if (mp3parse->xing_bytes == 0) { - GST_WARNING_OBJECT (mp3parse, "Invalid number of bytes in Xing header"); - mp3parse->xing_flags &= ~XING_BYTES_FLAG; - } - - data += 4; - } else { - mp3parse->xing_bytes = 0; - } - - /* If we know the upstream size and duration, compute the - * total bitrate, rounded up to the nearest kbit/sec */ - if ((total_time = mp3parse->xing_total_time) && - (total_bytes = mp3parse->xing_bytes)) { - mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes, - 8 * GST_SECOND, total_time); - mp3parse->xing_bitrate += 500; - mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000; - } - - if (xing_flags & XING_TOC_FLAG) { - int i, percent = 0; - guchar *table = mp3parse->xing_seek_table; - guchar old = 0, new; - guint first; - - first = data[0]; - GST_DEBUG_OBJECT (mp3parse, - "Subtracting initial offset of %d bytes from Xing TOC", first); - - /* xing seek table: percent time -> 1/256 bytepos */ - for (i = 0; i < 100; i++) { - new = data[i] - first; - if (old > new) { - GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC"); - mp3parse->xing_flags &= ~XING_TOC_FLAG; - goto skip_toc; - } - mp3parse->xing_seek_table[i] = old = new; - } - - /* build inverse table: 1/256 bytepos -> 1/100 percent time */ - for (i = 0; i < 256; i++) { - while (percent < 99 && table[percent + 1] <= i) - percent++; - - if (table[percent] == i) { - mp3parse->xing_seek_table_inverse[i] = percent * 100; - } else if (table[percent] < i && percent < 99) { - gdouble fa, fb, fx; - gint a = percent, b = percent + 1; - - fa = table[a]; - fb = table[b]; - fx = (b - a) / (fb - fa) * (i - fa) + a; - mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100); - } else if (percent == 99) { - gdouble fa, fb, fx; - gint a = percent, b = 100; - - fa = table[a]; - fb = 256.0; - fx = (b - a) / (fb - fa) * (i - fa) + a; - mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100); - } - } - skip_toc: - data += 100; - } else { - memset (mp3parse->xing_seek_table, 0, 100); - memset (mp3parse->xing_seek_table_inverse, 0, 256); - } - - if (xing_flags & XING_VBR_SCALE_FLAG) { - mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data); - } else - mp3parse->xing_vbr_scale = 0; - - GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %" - GST_TIME_FORMAT ", %u bytes, vbr scale %u", mp3parse->xing_frames, - GST_TIME_ARGS (mp3parse->xing_total_time), mp3parse->xing_bytes, - mp3parse->xing_vbr_scale); - - /* check for truncated file */ - if (upstream_total_bytes && mp3parse->xing_bytes && - mp3parse->xing_bytes * 0.8 > upstream_total_bytes) { - GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; " - "invalidating Xing header duration and size"); - mp3parse->xing_flags &= ~XING_BYTES_FLAG; - mp3parse->xing_flags &= ~XING_FRAMES_FLAG; - } - } else if (read_id == vbri_id) { - gint64 total_bytes, total_frames; - GstClockTime total_time; - guint16 nseek_points; - - GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id); - if (avail < offset + 26) { - GST_DEBUG_OBJECT (mp3parse, - "Not enough data to read VBRI header (need %d)", offset + 26); - return; - } - - GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header"); - data = gst_adapter_peek (mp3parse->adapter, offset + 26); - data += offset + 4; - - if (GST_READ_UINT16_BE (data) != 0x0001) { - GST_WARNING_OBJECT (mp3parse, - "Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data)); - return; - } - data += 2; - - /* Skip encoder delay */ - data += 2; - - /* Skip quality */ - data += 2; - - total_bytes = GST_READ_UINT32_BE (data); - if (total_bytes != 0) - mp3parse->vbri_bytes = total_bytes; - data += 4; - - total_frames = GST_READ_UINT32_BE (data); - if (total_frames != 0) { - mp3parse->vbri_frames = total_frames; - mp3parse->vbri_total_time = gst_util_uint64_scale (GST_SECOND, - (guint64) (mp3parse->vbri_frames) * (mp3parse->spf), mp3parse->rate); - } - data += 4; - - /* If we know the upstream size and duration, compute the - * total bitrate, rounded up to the nearest kbit/sec */ - if ((total_time = mp3parse->vbri_total_time) && - (total_bytes = mp3parse->vbri_bytes)) { - mp3parse->vbri_bitrate = gst_util_uint64_scale (total_bytes, - 8 * GST_SECOND, total_time); - mp3parse->vbri_bitrate += 500; - mp3parse->vbri_bitrate -= mp3parse->vbri_bitrate % 1000; - } - - nseek_points = GST_READ_UINT16_BE (data); - data += 2; - - if (nseek_points > 0) { - guint scale, seek_bytes, seek_frames; - gint i; - - mp3parse->vbri_seek_points = nseek_points; - - scale = GST_READ_UINT16_BE (data); - data += 2; - - seek_bytes = GST_READ_UINT16_BE (data); - data += 2; - - seek_frames = GST_READ_UINT16_BE (data); - - if (scale == 0 || seek_bytes == 0 || seek_bytes > 4 || seek_frames == 0) { - GST_WARNING_OBJECT (mp3parse, "Unsupported VBRI seek table"); - goto out_vbri; - } - - if (avail < offset + 26 + nseek_points * seek_bytes) { - GST_WARNING_OBJECT (mp3parse, - "Not enough data to read VBRI seek table (need %d)", - offset + 26 + nseek_points * seek_bytes); - goto out_vbri; - } - - if (seek_frames * nseek_points < total_frames - seek_frames || - seek_frames * nseek_points > total_frames + seek_frames) { - GST_WARNING_OBJECT (mp3parse, - "VBRI seek table doesn't cover the complete file"); - goto out_vbri; - } - - data = - gst_adapter_peek (mp3parse->adapter, - offset + 26 + nseek_points * seek_bytes); - data += offset + 26; - - - /* VBRI seek table: frame/seek_frames -> byte */ - mp3parse->vbri_seek_table = g_new (guint32, nseek_points); - if (seek_bytes == 4) - for (i = 0; i < nseek_points; i++) { - mp3parse->vbri_seek_table[i] = GST_READ_UINT32_BE (data) * scale; - data += 4; - } else if (seek_bytes == 3) - for (i = 0; i < nseek_points; i++) { - mp3parse->vbri_seek_table[i] = GST_READ_UINT24_BE (data) * scale; - data += 3; - } else if (seek_bytes == 2) - for (i = 0; i < nseek_points; i++) { - mp3parse->vbri_seek_table[i] = GST_READ_UINT16_BE (data) * scale; - data += 2; - } else /* seek_bytes == 1 */ - for (i = 0; i < nseek_points; i++) { - mp3parse->vbri_seek_table[i] = GST_READ_UINT8 (data) * scale; - data += 1; - } - } - out_vbri: - - GST_DEBUG_OBJECT (mp3parse, "VBRI header reported %u frames, time %" - GST_TIME_FORMAT ", bytes %u", mp3parse->vbri_frames, - GST_TIME_ARGS (mp3parse->vbri_total_time), mp3parse->vbri_bytes); - - /* check for truncated file */ - if (upstream_total_bytes && mp3parse->vbri_bytes && - mp3parse->vbri_bytes * 0.8 > upstream_total_bytes) { - GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; " - "invalidating VBRI header duration and size"); - mp3parse->vbri_valid = FALSE; - } else { - mp3parse->vbri_valid = TRUE; - } - } else { - GST_DEBUG_OBJECT (mp3parse, - "Xing, LAME or VBRI header not found in first frame"); - } -} - -static void -gst_mp3parse_check_seekability (GstMPEGAudioParse * mp3parse) -{ - GstQuery *query; - gboolean seekable = FALSE; - gint64 start = -1, stop = -1; - guint idx_interval = 0; - - query = gst_query_new_seeking (GST_FORMAT_BYTES); - if (!gst_pad_peer_query (mp3parse->sinkpad, query)) { - GST_DEBUG_OBJECT (mp3parse, "seeking query failed"); - goto done; - } - - gst_query_parse_seeking (query, NULL, &seekable, &start, &stop); - - /* try harder to query upstream size if we didn't get it the first time */ - if (seekable && stop == -1) { - GstFormat fmt = GST_FORMAT_BYTES; - - GST_DEBUG_OBJECT (mp3parse, "doing duration query to fix up unset stop"); - gst_pad_query_peer_duration (mp3parse->sinkpad, &fmt, &stop); - } - - /* if upstream doesn't know the size, it's likely that it's not seekable in - * practice even if it technically may be seekable */ - if (seekable && (start != 0 || stop <= start)) { - GST_DEBUG_OBJECT (mp3parse, "seekable but unknown start/stop -> disable"); - seekable = FALSE; - } - - /* let's not put every single frame into our index */ - if (seekable) { - if (stop < 10 * 1024 * 1024) - idx_interval = 100; - else if (stop < 100 * 1024 * 1024) - idx_interval = 500; - else - idx_interval = 1000; - } - -done: - - GST_INFO_OBJECT (mp3parse, "seekable: %d (%" G_GUINT64_FORMAT " - %" - G_GUINT64_FORMAT ")", seekable, start, stop); - mp3parse->seekable = seekable; - - GST_INFO_OBJECT (mp3parse, "idx_interval: %ums", idx_interval); - mp3parse->idx_interval = idx_interval * GST_MSECOND; - - gst_query_unref (query); -} - -/* Flush some number of bytes and update tracked offsets */ -static void -gst_mp3parse_flush_bytes (GstMPEGAudioParse * mp3parse, int bytes) -{ - gst_adapter_flush (mp3parse->adapter, bytes); - if (mp3parse->cur_offset != -1) - mp3parse->cur_offset += bytes; - mp3parse->tracked_offset += bytes; -} - -/* Perform extended validation to check that subsequent headers match - the first header given here in important characteristics, to avoid - false sync. We look for a minimum of MIN_RESYNC_FRAMES consecutive - frames to match their major characteristics. - - If at_eos is set to TRUE, we just check that we don't find any invalid - frames in whatever data is available, rather than requiring a full - MIN_RESYNC_FRAMES of data. - - Returns TRUE if we've seen enough data to validate or reject the frame. - If TRUE is returned, then *valid contains TRUE if it validated, or false - if we decided it was false sync. - */ -static gboolean -gst_mp3parse_validate_extended (GstMPEGAudioParse * mp3parse, guint32 header, - int bpf, gboolean at_eos, gboolean * valid) -{ - guint32 next_header; - const guint8 *data; - guint available; - int frames_found = 1; - int offset = bpf; - - while (frames_found < MIN_RESYNC_FRAMES) { - /* Check if we have enough data for all these frames, plus the next - frame header. */ - available = gst_adapter_available (mp3parse->adapter); - if (available < offset + 4) { - if (at_eos) { - /* Running out of data at EOS is fine; just accept it */ - *valid = TRUE; - return TRUE; - } else { - return FALSE; - } - } - - data = gst_adapter_peek (mp3parse->adapter, offset + 4); - next_header = GST_READ_UINT32_BE (data + offset); - GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d", - offset, (unsigned int) header, (unsigned int) next_header, bpf); - -/* mask the bits which are allowed to differ between frames */ -#define HDRMASK ~((0xF << 12) /* bitrate */ | \ - (0x1 << 9) /* padding */ | \ - (0xf << 4) /* mode|mode extension */ | \ - (0xf)) /* copyright|emphasis */ - - if ((next_header & HDRMASK) != (header & HDRMASK)) { - /* If any of the unmasked bits don't match, then it's not valid */ - GST_DEBUG_OBJECT (mp3parse, "next header doesn't match " - "(header=%08X (%08X), header2=%08X (%08X), bpf=%d)", - (guint) header, (guint) header & HDRMASK, (guint) next_header, - (guint) next_header & HDRMASK, bpf); - *valid = FALSE; - return TRUE; - } else if ((((next_header >> 12) & 0xf) == 0) || - (((next_header >> 12) & 0xf) == 0xf)) { - /* The essential parts were the same, but the bitrate held an - invalid value - also reject */ - GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)"); - *valid = FALSE; - return TRUE; - } - - bpf = mp3_type_frame_length_from_header (mp3parse, next_header, - NULL, NULL, NULL, NULL, NULL, NULL, NULL); - - offset += bpf; - frames_found++; - } - - *valid = TRUE; - return TRUE; -} - -static GstFlowReturn -gst_mp3parse_handle_data (GstMPEGAudioParse * mp3parse, gboolean at_eos) -{ - GstFlowReturn flow = GST_FLOW_OK; - const guchar *data; - guint32 header; - int bpf; - guint available; - guint bitrate, layer, rate, channels, version, mode, crc; - gboolean caps_change; - - /* while we still have at least 4 bytes (for the header) available */ - while (gst_adapter_available (mp3parse->adapter) >= 4) { - /* Get the header bytes, check if they're potentially valid */ - data = gst_adapter_peek (mp3parse->adapter, 4); - header = GST_READ_UINT32_BE (data); - - if (!head_check (mp3parse, header)) { - /* Not a valid MP3 header; we start looking forward byte-by-byte trying to - find a place to resync */ - if (!mp3parse->resyncing) - mp3parse->sync_offset = mp3parse->tracked_offset; - mp3parse->resyncing = TRUE; - gst_mp3parse_flush_bytes (mp3parse, 1); - GST_DEBUG_OBJECT (mp3parse, "wrong header, skipping byte"); - continue; - } - - /* We have a potentially valid header. - If this is just a normal 'next frame', we go ahead and output it. - - However, sometimes, we do additional validation to ensure we haven't - got false sync (common with mp3 due to the short sync word). - The additional validation requires that we find several consecutive mp3 - frames with the same major parameters, or reach EOS with a smaller - number of valid-looking frames. - - We do this if: - - This is the very first frame we've processed - - We're resyncing after a non-accurate seek, or after losing sync - due to invalid data. - - The format of the stream changes in a major way (number of channels, - sample rate, layer, or mpeg version). - */ - available = gst_adapter_available (mp3parse->adapter); - - if (G_UNLIKELY (mp3parse->resyncing && - mp3parse->tracked_offset - mp3parse->sync_offset > 2 * 1024 * 1024)) - goto sync_failure; - - bpf = mp3_type_frame_length_from_header (mp3parse, header, - &version, &layer, &channels, &bitrate, &rate, &mode, &crc); - g_assert (bpf != 0); - - if (channels != mp3parse->channels || - rate != mp3parse->rate || layer != mp3parse->layer || - version != mp3parse->version) - caps_change = TRUE; - else - caps_change = FALSE; - - if (mp3parse->resyncing || caps_change) { - gboolean valid; - if (!gst_mp3parse_validate_extended (mp3parse, header, bpf, at_eos, - &valid)) { - /* Not enough data to validate; wait for more */ - break; - } - - if (!valid) { - /* Extended validation failed; we probably got false sync. - Continue searching from the next byte in the stream */ - if (!mp3parse->resyncing) - mp3parse->sync_offset = mp3parse->tracked_offset; - mp3parse->resyncing = TRUE; - gst_mp3parse_flush_bytes (mp3parse, 1); - continue; - } - } - - /* if we don't have the whole frame... */ - if (available < bpf) { - GST_DEBUG_OBJECT (mp3parse, "insufficient data available, need " - "%d bytes, have %d", bpf, available); - break; - } - - if (caps_change) { - GstCaps *caps; - - caps = mp3_caps_create (version, layer, channels, rate); - gst_pad_set_caps (mp3parse->srcpad, caps); - gst_caps_unref (caps); - - mp3parse->channels = channels; - mp3parse->rate = rate; - - mp3parse->layer = layer; - mp3parse->version = version; - - /* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */ - if (mp3parse->layer == 1) - mp3parse->spf = 384; - else if (mp3parse->layer == 2) - mp3parse->spf = 1152; - else if (mp3parse->version == 1) { - mp3parse->spf = 1152; - } else { - /* MPEG-2 or "2.5" */ - mp3parse->spf = 576; - } - - mp3parse->max_bitreservoir = gst_util_uint64_scale (GST_SECOND, - ((version == 1) ? 10 : 30) * mp3parse->spf, mp3parse->rate); - } - - mp3parse->bit_rate = bitrate; - - /* Check the first frame for a Xing header to get our total length */ - if (mp3parse->frame_count == 0) { - /* For the first frame in the file, look for a Xing frame after - * the header, and output a codec tag */ - gst_mp3parse_handle_first_frame (mp3parse); - - /* Check if we're seekable */ - gst_mp3parse_check_seekability (mp3parse); - } - - /* Update VBR stats */ - mp3parse->bitrate_sum += mp3parse->bit_rate; - mp3parse->frame_count++; - /* Compute the average bitrate, rounded up to the nearest 1000 bits */ - mp3parse->avg_bitrate = - (mp3parse->bitrate_sum / mp3parse->frame_count + 500); - mp3parse->avg_bitrate -= mp3parse->avg_bitrate % 1000; - - if (!mp3parse->skip) { - mp3parse->resyncing = FALSE; - flow = gst_mp3parse_emit_frame (mp3parse, bpf, mode, crc); - if (flow != GST_FLOW_OK) - break; - } else { - GST_DEBUG_OBJECT (mp3parse, "skipping buffer of %d bytes", bpf); - gst_mp3parse_flush_bytes (mp3parse, bpf); - mp3parse->skip--; - } - } - - return flow; - - /* ERRORS */ -sync_failure: - { - GST_ELEMENT_ERROR (mp3parse, STREAM, DECODE, - ("Failed to parse stream"), (NULL)); - return GST_FLOW_ERROR; - } -} - -static GstFlowReturn -gst_mp3parse_chain (GstPad * pad, GstBuffer * buf) -{ - GstMPEGAudioParse *mp3parse; - GstClockTime timestamp; - - mp3parse = GST_MP3PARSE (GST_PAD_PARENT (pad)); - - GST_LOG_OBJECT (mp3parse, "buffer of %d bytes", GST_BUFFER_SIZE (buf)); - - timestamp = GST_BUFFER_TIMESTAMP (buf); - - mp3parse->discont |= GST_BUFFER_IS_DISCONT (buf); - - /* If we don't yet have a next timestamp, save it and the incoming offset - * so we can apply it to the right outgoing buffer */ - if (GST_CLOCK_TIME_IS_VALID (timestamp)) { - gint64 avail = gst_adapter_available (mp3parse->adapter); - - mp3parse->pending_ts = timestamp; - mp3parse->pending_offset = mp3parse->tracked_offset + avail; - - /* If we have no data pending and the next timestamp is - * invalid we can use the upstream timestamp for the next frame. - * - * This will give us a timestamp if we're resyncing and upstream - * gave us -1 as offset. */ - if (avail == 0 && !GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts)) - mp3parse->next_ts = timestamp; - - GST_LOG_OBJECT (mp3parse, "Have pending ts %" GST_TIME_FORMAT - " to apply in %" G_GINT64_FORMAT " bytes (@ off %" G_GINT64_FORMAT ")", - GST_TIME_ARGS (mp3parse->pending_ts), avail, mp3parse->pending_offset); - } - - /* Update the cur_offset we'll apply to outgoing buffers */ - if (mp3parse->cur_offset == -1 && GST_BUFFER_OFFSET (buf) != -1) - mp3parse->cur_offset = GST_BUFFER_OFFSET (buf); - - /* And add the data to the pool */ - gst_adapter_push (mp3parse->adapter, buf); - - return gst_mp3parse_handle_data (mp3parse, FALSE); -} - -static gboolean -head_check (GstMPEGAudioParse * mp3parse, unsigned long head) -{ - GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head); - /* if it's not a valid sync */ - if ((head & 0xffe00000) != 0xffe00000) { - GST_WARNING_OBJECT (mp3parse, "invalid sync"); - return FALSE; - } - /* if it's an invalid MPEG version */ - if (((head >> 19) & 3) == 0x1) { - GST_WARNING_OBJECT (mp3parse, "invalid MPEG version: 0x%lx", - (head >> 19) & 3); - return FALSE; - } - /* if it's an invalid layer */ - if (!((head >> 17) & 3)) { - GST_WARNING_OBJECT (mp3parse, "invalid layer: 0x%lx", (head >> 17) & 3); - return FALSE; - } - /* if it's an invalid bitrate */ - if (((head >> 12) & 0xf) == 0x0) { - GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx." - "Free format files are not supported yet", (head >> 12) & 0xf); - return FALSE; - } - if (((head >> 12) & 0xf) == 0xf) { - GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf); - return FALSE; - } - /* if it's an invalid samplerate */ - if (((head >> 10) & 0x3) == 0x3) { - GST_WARNING_OBJECT (mp3parse, "invalid samplerate: 0x%lx", - (head >> 10) & 0x3); - return FALSE; - } - - if ((head & 0x3) == 0x2) { - /* Ignore this as there are some files with emphasis 0x2 that can - * be played fine. See BGO #537235 */ - GST_WARNING_OBJECT (mp3parse, "invalid emphasis: 0x%lx", head & 0x3); - } - - return TRUE; -} - -static void -gst_mp3parse_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstMPEGAudioParse *src; - - src = GST_MP3PARSE (object); - - switch (prop_id) { - case ARG_SKIP: - src->skip = g_value_get_int (value); - break; - default: - break; - } -} - -static void -gst_mp3parse_get_property (GObject * object, guint prop_id, GValue * value, - GParamSpec * pspec) -{ - GstMPEGAudioParse *src; - - src = GST_MP3PARSE (object); - - switch (prop_id) { - case ARG_SKIP: - g_value_set_int (value, src->skip); - break; - case ARG_BIT_RATE: - g_value_set_int (value, src->bit_rate * 1000); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static GstStateChangeReturn -gst_mp3parse_change_state (GstElement * element, GstStateChange transition) -{ - GstMPEGAudioParse *mp3parse; - GstStateChangeReturn result; - - mp3parse = GST_MP3PARSE (element); - - result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_mp3parse_reset (mp3parse); - break; - default: - break; - } - - return result; -} - -static gboolean -mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total) -{ - GstFormat fmt = GST_FORMAT_BYTES; - - if (gst_pad_query_peer_duration (mp3parse->sinkpad, &fmt, total)) - return TRUE; - - if (mp3parse->xing_flags & XING_BYTES_FLAG) { - *total = mp3parse->xing_bytes; - return TRUE; - } - - if (mp3parse->vbri_bytes != 0 && mp3parse->vbri_valid) { - *total = mp3parse->vbri_bytes; - return TRUE; - } - - return FALSE; -} - -static gboolean -mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total) -{ - gint64 total_bytes; - - *total = GST_CLOCK_TIME_NONE; - - if (mp3parse->xing_flags & XING_FRAMES_FLAG) { - *total = mp3parse->xing_total_time; - return TRUE; - } - - if (mp3parse->vbri_total_time != 0 && mp3parse->vbri_valid) { - *total = mp3parse->vbri_total_time; - return TRUE; - } - - /* Calculate time from the measured bitrate */ - if (!mp3parse_total_bytes (mp3parse, &total_bytes)) - return FALSE; - - if (total_bytes != -1 - && !mp3parse_bytepos_to_time (mp3parse, total_bytes, total, TRUE)) - return FALSE; - - return TRUE; -} - -/* Convert a timestamp to the file position required to start decoding that - * timestamp. For now, this just uses the avg bitrate. Later, use an - * incrementally accumulated seek table */ -static gboolean -mp3parse_time_to_bytepos (GstMPEGAudioParse * mp3parse, GstClockTime ts, - gint64 * bytepos) -{ - gint64 total_bytes; - GstClockTime total_time; - - /* -1 always maps to -1 */ - if (ts == -1) { - *bytepos = -1; - return TRUE; - } - - /* If XING seek table exists use this for time->byte conversion */ - if ((mp3parse->xing_flags & XING_TOC_FLAG) && - (total_bytes = mp3parse->xing_bytes) && - (total_time = mp3parse->xing_total_time)) { - gdouble fa, fb, fx; - gdouble percent = - CLAMP ((100.0 * gst_util_guint64_to_gdouble (ts)) / - gst_util_guint64_to_gdouble (total_time), 0.0, 100.0); - gint index = CLAMP (percent, 0, 99); - - fa = mp3parse->xing_seek_table[index]; - if (index < 99) - fb = mp3parse->xing_seek_table[index + 1]; - else - fb = 256.0; - - fx = fa + (fb - fa) * (percent - index); - - *bytepos = (1.0 / 256.0) * fx * total_bytes; - - return TRUE; - } - - if (mp3parse->vbri_seek_table && (total_bytes = mp3parse->vbri_bytes) && - (total_time = mp3parse->vbri_total_time)) { - gint i, j; - gdouble a, b, fa, fb; - - i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time); - i = CLAMP (i, 0, mp3parse->vbri_seek_points - 1); - - a = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time, - mp3parse->vbri_seek_points)); - fa = 0.0; - for (j = i; j >= 0; j--) - fa += mp3parse->vbri_seek_table[j]; - - if (i + 1 < mp3parse->vbri_seek_points) { - b = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time, - mp3parse->vbri_seek_points)); - fb = fa + mp3parse->vbri_seek_table[i + 1]; - } else { - b = gst_guint64_to_gdouble (total_time); - fb = total_bytes; - } - - *bytepos = fa + ((fb - fa) / (b - a)) * (gst_guint64_to_gdouble (ts) - a); - - return TRUE; - } - - if (mp3parse->avg_bitrate == 0) - goto no_bitrate; - - *bytepos = - gst_util_uint64_scale (ts, mp3parse->avg_bitrate, (8 * GST_SECOND)); - return TRUE; -no_bitrate: - GST_DEBUG_OBJECT (mp3parse, "Cannot seek yet - no average bitrate"); - return FALSE; -} - -static gboolean -mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse, - gint64 bytepos, GstClockTime * ts, gboolean from_total_time) -{ - gint64 total_bytes; - GstClockTime total_time; - - if (bytepos == -1) { - *ts = GST_CLOCK_TIME_NONE; - return TRUE; - } - - if (bytepos == 0) { - *ts = 0; - return TRUE; - } - - /* If XING seek table exists use this for byte->time conversion */ - if (!from_total_time && (mp3parse->xing_flags & XING_TOC_FLAG) && - (total_bytes = mp3parse->xing_bytes) && - (total_time = mp3parse->xing_total_time)) { - gdouble fa, fb, fx; - gdouble pos; - gint index; - - pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0); - index = CLAMP (pos, 0, 255); - fa = mp3parse->xing_seek_table_inverse[index]; - if (index < 255) - fb = mp3parse->xing_seek_table_inverse[index + 1]; - else - fb = 10000.0; - - fx = fa + (fb - fa) * (pos - index); - - *ts = (1.0 / 10000.0) * fx * gst_util_guint64_to_gdouble (total_time); - - return TRUE; - } - - if (!from_total_time && mp3parse->vbri_seek_table && - (total_bytes = mp3parse->vbri_bytes) && - (total_time = mp3parse->vbri_total_time)) { - gint i = 0; - guint64 sum = 0; - gdouble a, b, fa, fb; - - do { - sum += mp3parse->vbri_seek_table[i]; - i++; - } while (i + 1 < mp3parse->vbri_seek_points - && sum + mp3parse->vbri_seek_table[i] < bytepos); - i--; - - a = gst_guint64_to_gdouble (sum); - fa = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time, - mp3parse->vbri_seek_points)); - - if (i + 1 < mp3parse->vbri_seek_points) { - b = a + mp3parse->vbri_seek_table[i + 1]; - fb = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time, - mp3parse->vbri_seek_points)); - } else { - b = total_bytes; - fb = gst_guint64_to_gdouble (total_time); - } - - *ts = gst_gdouble_to_guint64 (fa + ((fb - fa) / (b - a)) * (bytepos - a)); - - return TRUE; - } - - /* Cannot convert anything except 0 if we don't have a bitrate yet */ - if (mp3parse->avg_bitrate == 0) - return FALSE; - - *ts = (GstClockTime) gst_util_uint64_scale (GST_SECOND, bytepos * 8, - mp3parse->avg_bitrate); - return TRUE; -} - -static gboolean -mp3parse_handle_seek (GstMPEGAudioParse * mp3parse, GstEvent * event) -{ - GstFormat format; - gdouble rate; - GstSeekFlags flags; - GstSeekType cur_type, stop_type; - gint64 cur, stop; - gint64 byte_cur, byte_stop; - MPEGAudioPendingAccurateSeek *seek; - GstClockTime start; - - gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur, - &stop_type, &stop); - - GST_DEBUG_OBJECT (mp3parse, "Performing seek to %" GST_TIME_FORMAT, - GST_TIME_ARGS (cur)); - - /* For any format other than TIME, see if upstream handles - * it directly or fail. For TIME, try upstream, but do it ourselves if - * it fails upstream */ - if (format != GST_FORMAT_TIME) { - gst_event_ref (event); - return gst_pad_push_event (mp3parse->sinkpad, event); - } else { - gst_event_ref (event); - if (gst_pad_push_event (mp3parse->sinkpad, event)) - return TRUE; - } - - seek = g_new0 (MPEGAudioPendingAccurateSeek, 1); - - seek->segment = mp3parse->segment; - - gst_segment_set_seek (&seek->segment, rate, GST_FORMAT_TIME, - flags, cur_type, cur, stop_type, stop, NULL); - - /* Handle TIME based seeks by converting to a BYTE position */ - - /* For accurate seeking get the frame 9 (MPEG1) or 29 (MPEG2) frames - * before the one we want to seek to and push them all to the decoder. - * - * This is necessary because of the bit reservoir. See - * http://www.mars.org/mailman/public/mad-dev/2002-May/000634.html - * - */ - - if (flags & GST_SEEK_FLAG_ACCURATE) { - if (!mp3parse->seek_table) { - byte_cur = 0; - byte_stop = -1; - start = 0; - } else { - MPEGAudioSeekEntry *entry = NULL, *start_entry = NULL, *stop_entry = NULL; - GList *start_node, *stop_node; - gint64 seek_ts = (cur > mp3parse->max_bitreservoir) ? - (cur - mp3parse->max_bitreservoir) : 0; - - for (start_node = mp3parse->seek_table; start_node; - start_node = start_node->next) { - entry = start_node->data; - - if (seek_ts >= entry->timestamp) { - start_entry = entry; - break; - } - } - - if (!start_entry) { - start_entry = mp3parse->seek_table->data; - start = start_entry->timestamp; - byte_cur = start_entry->byte; - } else { - start = start_entry->timestamp; - byte_cur = start_entry->byte; - } - - for (stop_node = mp3parse->seek_table; stop_node; - stop_node = stop_node->next) { - entry = stop_node->data; - - if (stop >= entry->timestamp) { - stop_node = stop_node->prev; - stop_entry = (stop_node) ? stop_node->data : NULL; - break; - } - } - - if (!stop_entry) { - byte_stop = -1; - } else { - byte_stop = stop_entry->byte; - } - - } - event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, - byte_cur, stop_type, byte_stop); - g_mutex_lock (mp3parse->pending_seeks_lock); - seek->upstream_start = byte_cur; - seek->timestamp_start = start; - mp3parse->pending_accurate_seeks = - g_slist_prepend (mp3parse->pending_accurate_seeks, seek); - g_mutex_unlock (mp3parse->pending_seeks_lock); - if (gst_pad_push_event (mp3parse->sinkpad, event)) { - mp3parse->exact_position = TRUE; - return TRUE; - } else { - mp3parse->exact_position = TRUE; - g_mutex_lock (mp3parse->pending_seeks_lock); - mp3parse->pending_accurate_seeks = - g_slist_remove (mp3parse->pending_accurate_seeks, seek); - g_mutex_unlock (mp3parse->pending_seeks_lock); - g_free (seek); - return FALSE; - } - } - - mp3parse->exact_position = FALSE; - - /* Convert the TIME to the appropriate BYTE position at which to resume - * decoding. */ - if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) cur, &byte_cur)) - goto no_pos; - if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) stop, &byte_stop)) - goto no_pos; - - GST_DEBUG_OBJECT (mp3parse, "Seeking to byte range %" G_GINT64_FORMAT - " to %" G_GINT64_FORMAT, byte_cur, byte_stop); - - /* Send BYTE based seek upstream */ - event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type, - byte_cur, stop_type, byte_stop); - - GST_LOG_OBJECT (mp3parse, "Storing pending seek"); - g_mutex_lock (mp3parse->pending_seeks_lock); - seek->upstream_start = byte_cur; - seek->timestamp_start = cur; - mp3parse->pending_nonaccurate_seeks = - g_slist_prepend (mp3parse->pending_nonaccurate_seeks, seek); - g_mutex_unlock (mp3parse->pending_seeks_lock); - if (gst_pad_push_event (mp3parse->sinkpad, event)) { - return TRUE; - } else { - g_mutex_lock (mp3parse->pending_seeks_lock); - mp3parse->pending_nonaccurate_seeks = - g_slist_remove (mp3parse->pending_nonaccurate_seeks, seek); - g_mutex_unlock (mp3parse->pending_seeks_lock); - g_free (seek); - return FALSE; - } - -no_pos: - GST_DEBUG_OBJECT (mp3parse, - "Could not determine byte position for desired time"); - return FALSE; -} - -static gboolean -mp3parse_src_event (GstPad * pad, GstEvent * event) -{ - GstMPEGAudioParse *mp3parse; - gboolean res = FALSE; - - mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_SEEK: - res = mp3parse_handle_seek (mp3parse, event); - gst_event_unref (event); - break; - default: - res = gst_pad_event_default (pad, event); - break; - } - - gst_object_unref (mp3parse); - return res; -} - -static gboolean -mp3parse_src_query (GstPad * pad, GstQuery * query) -{ - GstFormat format; - GstClockTime total; - GstMPEGAudioParse *mp3parse; - gboolean res = FALSE; - GstPad *peer; - - mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad)); - - GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query)); - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_POSITION: - gst_query_parse_position (query, &format, NULL); - - if (format == GST_FORMAT_BYTES || format == GST_FORMAT_DEFAULT) { - if (mp3parse->cur_offset != -1) { - gst_query_set_position (query, GST_FORMAT_BYTES, - mp3parse->cur_offset); - res = TRUE; - } - } else if (format == GST_FORMAT_TIME) { - if (mp3parse->next_ts == GST_CLOCK_TIME_NONE) - goto out; - gst_query_set_position (query, GST_FORMAT_TIME, mp3parse->next_ts); - res = TRUE; - } - - /* If no answer above, see if upstream knows */ - if (!res) { - if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) { - res = gst_pad_query (peer, query); - gst_object_unref (peer); - if (res) - goto out; - } - } - break; - case GST_QUERY_DURATION: - gst_query_parse_duration (query, &format, NULL); - - /* First, see if upstream knows */ - if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) { - res = gst_pad_query (peer, query); - gst_object_unref (peer); - if (res) - goto out; - } - - if (format == GST_FORMAT_TIME) { - if (!mp3parse_total_time (mp3parse, &total) || total == -1) - goto out; - gst_query_set_duration (query, format, total); - res = TRUE; - } - break; - case GST_QUERY_SEEKING: - gst_query_parse_seeking (query, &format, NULL, NULL, NULL); - - /* does upstream handle ? */ - if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) { - res = gst_pad_query (peer, query); - gst_object_unref (peer); - } - /* we may be able to help if in TIME */ - if (format == GST_FORMAT_TIME) { - gboolean seekable; - - gst_query_parse_seeking (query, &format, &seekable, NULL, NULL); - /* already OK if upstream takes care */ - if (!(res && seekable)) { - gint64 pos; - - seekable = TRUE; - if (!mp3parse_total_time (mp3parse, &total) || total == -1) { - seekable = FALSE; - } else if (!mp3parse_time_to_bytepos (mp3parse, 0, &pos)) { - seekable = FALSE; - } else { - GstQuery *q; - - q = gst_query_new_seeking (GST_FORMAT_BYTES); - if (!gst_pad_peer_query (mp3parse->sinkpad, q)) { - seekable = FALSE; - } else { - gst_query_parse_seeking (q, &format, &seekable, NULL, NULL); - } - gst_query_unref (q); - } - gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0, total); - res = TRUE; - } - } - break; - default: - res = gst_pad_query_default (pad, query); - break; - } - -out: - gst_object_unref (mp3parse); - return res; -} - -static const GstQueryType * -mp3parse_get_query_types (GstPad * pad G_GNUC_UNUSED) -{ - static const GstQueryType query_types[] = { - GST_QUERY_POSITION, - GST_QUERY_DURATION, - 0 - }; - - return query_types; -} diff --git a/gst/mpegaudioparse/gstmpegaudioparse.h b/gst/mpegaudioparse/gstmpegaudioparse.h deleted file mode 100644 index 3a2852d39d..0000000000 --- a/gst/mpegaudioparse/gstmpegaudioparse.h +++ /dev/null @@ -1,151 +0,0 @@ -/* GStreamer - * Copyright (C) <1999> Erik Walthinsen - * - * 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 __MP3PARSE_H__ -#define __MP3PARSE_H__ - - -#include -#include - -G_BEGIN_DECLS - -#define GST_TYPE_MP3PARSE \ - (gst_mp3parse_get_type()) -#define GST_MP3PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_MP3PARSE,GstMPEGAudioParse)) -#define GST_MP3PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_MP3PARSE,GstMPEGAudioParseClass)) -#define GST_IS_MP3PARSE(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_MP3PARSE)) -#define GST_IS_MP3PARSE_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_MP3PARSE)) - -typedef struct _GstMPEGAudioParse GstMPEGAudioParse; -typedef struct _GstMPEGAudioParseClass GstMPEGAudioParseClass; -typedef struct _MPEGAudioSeekEntry MPEGAudioSeekEntry; -typedef struct _MPEGAudioPendingAccurateSeek MPEGAudioPendingAccurateSeek; - - -struct _MPEGAudioSeekEntry { - gint64 byte; - GstClockTime timestamp; -}; - -struct _MPEGAudioPendingAccurateSeek { - GstSegment segment; - gint64 upstream_start; - GstClockTime timestamp_start; -}; - -struct _GstMPEGAudioParse { - GstElement element; - - GstPad *sinkpad, *srcpad; - - GstSegment segment; - GstClockTime next_ts; - gboolean discont; - - /* Offset as supplied by incoming buffers */ - gint64 cur_offset; - - /* Upcoming timestamp given on an incoming buffer and - * the offset at which it becomes active */ - GstClockTime pending_ts; - gint64 pending_offset; - /* Offset since the last newseg */ - gint64 tracked_offset; - /* tracked_offset when resyncing started */ - gint64 sync_offset; - - GstAdapter *adapter; - - guint skip; /* number of frames to skip */ - guint bit_rate; /* in kbps */ - gint channels, rate, layer, version; - GstClockTime max_bitreservoir; - gint spf; /* Samples per frame */ - - gboolean resyncing; /* True when attempting to resync (stricter checks are - performed) */ - gboolean sent_codec_tag; - - /* VBR tracking */ - guint avg_bitrate; - guint64 bitrate_sum; - guint frame_count; - guint last_posted_bitrate; - gint last_posted_crc; - guint last_posted_channel_mode; - - /* Xing info */ - guint32 xing_flags; - guint32 xing_frames; - GstClockTime xing_total_time; - guint32 xing_bytes; - /* percent -> filepos mapping */ - guchar xing_seek_table[100]; - /* filepos -> percent mapping */ - guint16 xing_seek_table_inverse[256]; - guint32 xing_vbr_scale; - guint xing_bitrate; - - /* VBRI info */ - guint32 vbri_frames; - GstClockTime vbri_total_time; - guint32 vbri_bytes; - guint vbri_bitrate; - guint vbri_seek_points; - guint32 *vbri_seek_table; - gboolean vbri_valid; - - /* Accurate seeking */ - GList *seek_table; - GMutex *pending_seeks_lock; - GSList *pending_accurate_seeks; - gboolean exact_position; - - GSList *pending_nonaccurate_seeks; - - /* Track whether we're seekable (in BYTES format, if upstream operates in - * TIME format, we don't care about seekability and assume upstream handles - * it). The seek table for accurate seeking is not maintained if we're not - * seekable. */ - gboolean seekable; - - /* minimum distance between two index entries */ - GstClockTimeDiff idx_interval; - - /* pending segment */ - GstEvent *pending_segment; - /* pending events */ - GList *pending_events; -}; - -struct _GstMPEGAudioParseClass { - GstElementClass parent_class; -}; - -GType gst_mp3parse_get_type(void); - -G_END_DECLS - -#endif /* __MP3PARSE_H__ */ diff --git a/gst/mpegaudioparse/mpegaudioparse.vcproj b/gst/mpegaudioparse/mpegaudioparse.vcproj deleted file mode 100644 index 198e78e4cf..0000000000 --- a/gst/mpegaudioparse/mpegaudioparse.vcproj +++ /dev/null @@ -1,148 +0,0 @@ - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - diff --git a/gst/xingmux/Makefile.am b/gst/xingmux/Makefile.am new file mode 100644 index 0000000000..5439a42c55 --- /dev/null +++ b/gst/xingmux/Makefile.am @@ -0,0 +1,25 @@ +# FIXME 0.11: element should move somewhere else really, such as +# gst-plugins-good/gst/tags/ or so +plugin_LTLIBRARIES = libgstxingmux.la + +libgstxingmux_la_SOURCES = plugin.c gstxingmux.c +libgstxingmux_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) +libgstxingmux_la_LIBADD = $(GST_BASE_LIBS) $(GST_LIBS) +libgstxingmux_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) +libgstxingmux_la_LIBTOOLFLAGS = --tag=disable-static + +noinst_HEADERS = gstxingmux.h + +Android.mk: Makefile.am $(BUILT_SOURCES) + androgenizer \ + -:PROJECT libgstxingmux -:SHARED libgstxingmux \ + -:TAGS eng debug \ + -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \ + -:SOURCES $(libgstxingmux_la_SOURCES) \ + -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstxingmux_la_CFLAGS) \ + -:LDFLAGS $(libgstxingmux_la_LDFLAGS) \ + $(libgstxingmux_la_LIBADD) \ + -ldl \ + -:PASSTHROUGH LOCAL_ARM_MODE:=arm \ + LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \ + > $@ diff --git a/gst/mpegaudioparse/gstxingmux.c b/gst/xingmux/gstxingmux.c similarity index 100% rename from gst/mpegaudioparse/gstxingmux.c rename to gst/xingmux/gstxingmux.c diff --git a/gst/mpegaudioparse/gstxingmux.h b/gst/xingmux/gstxingmux.h similarity index 100% rename from gst/mpegaudioparse/gstxingmux.h rename to gst/xingmux/gstxingmux.h diff --git a/gst/mpegaudioparse/plugin.c b/gst/xingmux/plugin.c similarity index 84% rename from gst/mpegaudioparse/plugin.c rename to gst/xingmux/plugin.c index 0e7d652a19..a871a4ab39 100644 --- a/gst/mpegaudioparse/plugin.c +++ b/gst/xingmux/plugin.c @@ -22,7 +22,6 @@ #endif #include -#include "gstmpegaudioparse.h" #include "gstxingmux.h" static gboolean @@ -31,15 +30,12 @@ plugin_init (GstPlugin * plugin) if (!gst_element_register (plugin, "xingmux", GST_RANK_NONE, GST_TYPE_XING_MUX)) return FALSE; - if (!gst_element_register (plugin, "mp3parse", GST_RANK_PRIMARY + 1, - GST_TYPE_MP3PARSE)) - return FALSE; return TRUE; } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, - "mpegaudioparse", - "MPEG-1 layer 1/2/3 audio stream elements", + "xingmux", + "Add XING tags to mpeg audio files", plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);