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gst/realmedia/: Add demuxer for RealAudio files (#349779).
Original commit message from CVS: * gst/realmedia/Makefile.am: * gst/realmedia/rademux.c: * gst/realmedia/rademux.h: * gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream), (gst_rmdemux_descramble_dnet_audio), (gst_rmdemux_plugin_init): * gst/realmedia/rmutils.c: (gst_rm_utils_descramble_dnet_buffer): * gst/realmedia/rmutils.h: Add demuxer for RealAudio files (#349779).
This commit is contained in:
Tim-Philipp Müller
parent
6dcec924db
commit
a638e98a98
8 changed files with 1117 additions and 15 deletions
11
ChangeLog
11
ChangeLog
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@ -1,3 +1,14 @@
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2007-01-11 Tim-Philipp Müller <tim at centricular dot net>
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* gst/realmedia/Makefile.am:
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* gst/realmedia/rademux.c:
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* gst/realmedia/rademux.h:
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* gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream),
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(gst_rmdemux_descramble_dnet_audio), (gst_rmdemux_plugin_init):
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* gst/realmedia/rmutils.c: (gst_rm_utils_descramble_dnet_buffer):
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* gst/realmedia/rmutils.h:
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Add demuxer for RealAudio files (#349779).
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2007-01-07 Sébastien Moutte <sebastien@moutte.net>
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2007-01-07 Sébastien Moutte <sebastien@moutte.net>
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* Makefile.am:
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* Makefile.am:
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2
common
2
common
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@ -1 +1 @@
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Subproject commit 64f924f6f2ff6275b06facb4c2adbc7c05f70641
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Subproject commit 8ba5dffb5ee7e7daea1030f6b34bfef10f9801a3
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@ -1,9 +1,9 @@
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plugin_LTLIBRARIES = libgstrmdemux.la
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plugin_LTLIBRARIES = libgstrmdemux.la
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libgstrmdemux_la_SOURCES = rmdemux.c rmutils.c rdtdepay.c
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libgstrmdemux_la_SOURCES = rademux.c rmdemux.c rmutils.c rdtdepay.c
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libgstrmdemux_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
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libgstrmdemux_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
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libgstrmdemux_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
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libgstrmdemux_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
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libgstrmdemux_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstrmdemux_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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noinst_HEADERS = rmdemux.h rmutils.h rdtdepay.h
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noinst_HEADERS = rademux.h rmdemux.h rmutils.h rdtdepay.h
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973
gst/realmedia/rademux.c
Normal file
973
gst/realmedia/rademux.c
Normal file
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@ -0,0 +1,973 @@
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/* GStreamer RealAudio demuxer
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* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-rademux
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*
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* <refsect2>
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* <para>
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* Demuxes/parses a RealAudio (.ra) file or stream into compressed audio.
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* </para>
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* <title>Example launch line</title>
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* <para>
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* <programlisting>
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* gst-launch filesrc location=interview.ra ! rademux ! ffdec_real_288 ! audioconvert ! audioresample ! alsasink
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* </programlisting>
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* Read a RealAudio file and decode it and output it to the soundcard using
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* the ALSA element. The .ra file is assumed to contain RealAudio version 2.
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* </para>
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* <para>
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* <programlisting>
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* gst-launch gnomevfssrc location=http://www.example.org/interview.ra ! rademux ! a52dec ! audioconvert ! audioresample ! alsasink
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* </programlisting>
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* Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
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* and output it to the soundcard using the ALSA element.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2006-10-24 (0.10.5)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "rademux.h"
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#include "rmdemux.h"
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#include "rmutils.h"
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#include <string.h>
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static GstElementDetails real_audio_demux_details = {
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"RealAudio Demuxer",
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"Codec/Demuxer",
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"Demultiplex a RealAudio file",
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"Tim-Philipp Müller <tim centricular net>"
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};
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-pn-realaudio")
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);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS_ANY);
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GST_DEBUG_CATEGORY_STATIC (real_audio_demux_debug);
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#define GST_CAT_DEFAULT real_audio_demux_debug
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GST_BOILERPLATE (GstRealAudioDemux, gst_real_audio_demux, GstElement,
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GST_TYPE_ELEMENT);
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static GstStateChangeReturn gst_real_audio_demux_change_state (GstElement * e,
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GstStateChange transition);
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static GstFlowReturn gst_real_audio_demux_chain (GstPad * pad, GstBuffer * buf);
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static gboolean gst_real_audio_demux_sink_event (GstPad * pad, GstEvent * ev);
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static gboolean gst_real_audio_demux_src_event (GstPad * pad, GstEvent * ev);
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static gboolean gst_real_audio_demux_src_query (GstPad * pad, GstQuery * query);
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static void gst_real_audio_demux_loop (GstRealAudioDemux * demux);
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static gboolean gst_real_audio_demux_sink_activate (GstPad * sinkpad);
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static gboolean gst_real_audio_demux_sink_activate_push (GstPad * sinkpad,
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gboolean active);
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static gboolean gst_real_audio_demux_sink_activate_pull (GstPad * sinkpad,
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gboolean active);
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static void
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gst_real_audio_demux_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details (element_class, &real_audio_demux_details);
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GST_DEBUG_CATEGORY_INIT (real_audio_demux_debug, "rademux",
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0, "Demuxer for RealAudio streams");
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}
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static void
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gst_real_audio_demux_finalize (GObject * obj)
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{
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GstRealAudioDemux *demux = GST_REAL_AUDIO_DEMUX (obj);
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g_object_unref (demux->adapter);
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G_OBJECT_CLASS (parent_class)->finalize (obj);
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}
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static void
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gst_real_audio_demux_class_init (GstRealAudioDemuxClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *gstelement_class = (GstElementClass *) klass;
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gobject_class->finalize = gst_real_audio_demux_finalize;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_change_state);
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}
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static void
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gst_real_audio_demux_reset (GstRealAudioDemux * demux)
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{
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gst_adapter_clear (demux->adapter);
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if (demux->srcpad) {
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GST_DEBUG_OBJECT (demux, "Removing source pad");
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gst_element_remove_pad (GST_ELEMENT (demux), demux->srcpad);
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demux->srcpad = NULL;
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}
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if (demux->pending_tags) {
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gst_tag_list_free (demux->pending_tags);
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demux->pending_tags = NULL;
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}
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demux->state = REAL_AUDIO_DEMUX_STATE_MARKER;
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demux->ra_version = 0;
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demux->data_offset = 0;
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demux->packet_size = 0;
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demux->sample_rate = 0;
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demux->sample_width = 0;
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demux->channels = 0;
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demux->fourcc = 0;
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demux->need_newsegment = TRUE;
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demux->segment_running = FALSE;
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demux->byterate_num = 0;
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demux->byterate_denom = 0;
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demux->duration = 0;
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demux->upstream_size = 0;
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demux->offset = 0;
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gst_adapter_clear (demux->adapter);
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}
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static void
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gst_real_audio_demux_init (GstRealAudioDemux * demux,
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GstRealAudioDemuxClass * klass)
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{
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demux->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
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gst_pad_set_chain_function (demux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_chain));
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gst_pad_set_event_function (demux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_event));
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gst_pad_set_activate_function (demux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_activate));
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gst_pad_set_activatepull_function (demux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_activate_pull));
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gst_pad_set_activatepush_function (demux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_activate_push));
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gst_element_add_pad (GST_ELEMENT (demux), demux->sinkpad);
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demux->adapter = gst_adapter_new ();
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gst_real_audio_demux_reset (demux);
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}
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static gboolean
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gst_real_audio_demux_sink_activate (GstPad * sinkpad)
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{
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if (gst_pad_check_pull_range (sinkpad)) {
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return gst_pad_activate_pull (sinkpad, TRUE);
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} else {
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return gst_pad_activate_push (sinkpad, TRUE);
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}
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}
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static gboolean
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gst_real_audio_demux_sink_activate_push (GstPad * sinkpad, gboolean active)
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{
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GstRealAudioDemux *demux;
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demux = GST_REAL_AUDIO_DEMUX (GST_OBJECT_PARENT (sinkpad));
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demux->seekable = FALSE;
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return TRUE;
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}
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static gboolean
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gst_real_audio_demux_sink_activate_pull (GstPad * sinkpad, gboolean active)
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{
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GstRealAudioDemux *demux;
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demux = GST_REAL_AUDIO_DEMUX (GST_OBJECT_PARENT (sinkpad));
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if (active) {
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demux->seekable = TRUE;
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return gst_pad_start_task (sinkpad,
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(GstTaskFunction) gst_real_audio_demux_loop, demux);
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} else {
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demux->seekable = FALSE;
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return gst_pad_stop_task (sinkpad);
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}
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}
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static GstFlowReturn
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gst_real_audio_demux_parse_marker (GstRealAudioDemux * demux)
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{
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const guint8 *data;
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if (gst_adapter_available (demux->adapter) < 6) {
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GST_LOG_OBJECT (demux, "need at least 6 bytes, waiting for more data");
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return GST_FLOW_OK;
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}
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data = gst_adapter_peek (demux->adapter, 6);
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if (memcmp (data, ".ra\375", 4) != 0)
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goto wrong_format;
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demux->ra_version = GST_READ_UINT16_BE (data + 4);
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GST_DEBUG_OBJECT (demux, "ra_version = %u", demux->ra_version);
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if (demux->ra_version != 4 && demux->ra_version != 3)
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goto unsupported_ra_version;
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gst_adapter_flush (demux->adapter, 6);
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demux->state = REAL_AUDIO_DEMUX_STATE_HEADER;
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return GST_FLOW_OK;
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/* ERRORS */
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wrong_format:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (demux), STREAM, WRONG_TYPE, (NULL), (NULL));
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return GST_FLOW_ERROR;
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}
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unsupported_ra_version:
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{
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GST_ELEMENT_ERROR (GST_ELEMENT (demux), STREAM, DECODE,
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("Cannot decode this RealAudio file, please file a bug"),
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("ra_version = %u", demux->ra_version));
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return GST_FLOW_ERROR;
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}
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}
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static GstClockTime
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gst_real_demux_get_timestamp_from_offset (GstRealAudioDemux * demux,
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guint64 offset)
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{
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if (offset >= demux->data_offset && demux->byterate_num > 0 &&
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demux->byterate_denom > 0) {
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return gst_util_uint64_scale (offset - demux->data_offset,
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demux->byterate_denom * GST_SECOND, demux->byterate_num);
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} else if (offset == demux->data_offset) {
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return (GstClockTime) 0;
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} else {
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return GST_CLOCK_TIME_NONE;
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}
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}
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static gboolean
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gst_real_audio_demux_get_data_offset_from_header (GstRealAudioDemux * demux)
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{
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const guint8 *data;
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data = gst_adapter_peek (demux->adapter, 16);
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g_assert (data != NULL);
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switch (demux->ra_version) {
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case 3:
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demux->data_offset = GST_READ_UINT16_BE (data) + 8;
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break;
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case 4:
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demux->data_offset = GST_READ_UINT32_BE (data + 12) + 16;
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break;
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default:
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demux->data_offset = 0;
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g_return_val_if_reached (FALSE);
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}
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|
|
||||||
|
return TRUE;
|
||||||
|
}
|
||||||
|
|
||||||
|
static GstFlowReturn
|
||||||
|
gst_real_audio_demux_parse_header (GstRealAudioDemux * demux)
|
||||||
|
{
|
||||||
|
const gchar *codec_name = NULL;
|
||||||
|
const guint8 *data;
|
||||||
|
GstCaps *caps = NULL;
|
||||||
|
guint avail;
|
||||||
|
|
||||||
|
g_assert (demux->ra_version == 4 || demux->ra_version == 3);
|
||||||
|
|
||||||
|
avail = gst_adapter_available (demux->adapter);
|
||||||
|
if (avail < 16)
|
||||||
|
return GST_FLOW_OK;
|
||||||
|
|
||||||
|
if (!gst_real_audio_demux_get_data_offset_from_header (demux))
|
||||||
|
return GST_FLOW_ERROR; /* shouldn't happen */
|
||||||
|
|
||||||
|
GST_DEBUG_OBJECT (demux, "data_offset = %u", demux->data_offset);
|
||||||
|
|
||||||
|
if (avail + 6 < demux->data_offset) {
|
||||||
|
GST_DEBUG_OBJECT (demux, "Need %u bytes, but only %u available now",
|
||||||
|
demux->data_offset - 6, avail);
|
||||||
|
return GST_FLOW_OK;
|
||||||
|
}
|
||||||
|
|
||||||
|
data = gst_adapter_peek (demux->adapter, demux->data_offset - 6);
|
||||||
|
g_assert (data);
|
||||||
|
|
||||||
|
switch (demux->ra_version) {
|
||||||
|
case 3:
|
||||||
|
demux->fourcc = GST_RM_AUD_14_4;
|
||||||
|
demux->packet_size = 20;
|
||||||
|
demux->sample_rate = 8000;
|
||||||
|
demux->channels = 1;
|
||||||
|
demux->sample_width = 16;
|
||||||
|
demux->flavour = 1;
|
||||||
|
demux->leaf_size = 0;
|
||||||
|
demux->height = 0;
|
||||||
|
break;
|
||||||
|
case 4:
|
||||||
|
demux->flavour = GST_READ_UINT16_BE (data + 16);
|
||||||
|
/* demux->frame_size = GST_READ_UINT32_BE (data + 36); */
|
||||||
|
demux->leaf_size = GST_READ_UINT16_BE (data + 38);
|
||||||
|
demux->height = GST_READ_UINT16_BE (data + 34);
|
||||||
|
demux->packet_size = GST_READ_UINT32_BE (data + 18);
|
||||||
|
demux->sample_rate = GST_READ_UINT16_BE (data + 42);
|
||||||
|
demux->sample_width = GST_READ_UINT16_BE (data + 46);
|
||||||
|
demux->channels = GST_READ_UINT16_BE (data + 48);
|
||||||
|
demux->fourcc = GST_READ_UINT32_LE (data + 56);
|
||||||
|
demux->pending_tags = gst_rm_utils_read_tags (data + 63,
|
||||||
|
demux->data_offset - 63, gst_rm_utils_read_string8);
|
||||||
|
break;
|
||||||
|
default:
|
||||||
|
g_assert_not_reached ();
|
||||||
|
#if 0
|
||||||
|
case 5:
|
||||||
|
demux->flavour = GST_READ_UINT16_BE (data + 16);
|
||||||
|
/* demux->frame_size = GST_READ_UINT32_BE (data + 36); */
|
||||||
|
demux->leaf_size = GST_READ_UINT16_BE (data + 38);
|
||||||
|
demux->height = GST_READ_UINT16_BE (data + 34);
|
||||||
|
|
||||||
|
demux->sample_rate = GST_READ_UINT16_BE (data + 48);
|
||||||
|
demux->sample_width = GST_READ_UINT16_BE (data + 52);
|
||||||
|
demux->n_channels = GST_READ_UINT16_BE (data + 54);
|
||||||
|
demux->fourcc = RMDEMUX_FOURCC_GET (data + 60);
|
||||||
|
break;
|
||||||
|
#endif
|
||||||
|
}
|
||||||
|
|
||||||
|
GST_INFO_OBJECT (demux, "packet_size = %u", demux->packet_size);
|
||||||
|
GST_INFO_OBJECT (demux, "sample_rate = %u", demux->sample_rate);
|
||||||
|
GST_INFO_OBJECT (demux, "sample_width = %u", demux->sample_width);
|
||||||
|
GST_INFO_OBJECT (demux, "channels = %u", demux->channels);
|
||||||
|
GST_INFO_OBJECT (demux, "fourcc = '%" GST_FOURCC_FORMAT "' (%08X)",
|
||||||
|
GST_FOURCC_ARGS (demux->fourcc), demux->fourcc);
|
||||||
|
|
||||||
|
switch (demux->fourcc) {
|
||||||
|
case GST_RM_AUD_14_4:
|
||||||
|
caps = gst_caps_new_simple ("audio/x-pn-realaudio", "raversion",
|
||||||
|
G_TYPE_INT, 1, NULL);
|
||||||
|
codec_name = "Real Audio 14.4kbps";
|
||||||
|
demux->byterate_num = 1000;
|
||||||
|
demux->byterate_denom = 1;
|
||||||
|
break;
|
||||||
|
|
||||||
|
case GST_RM_AUD_28_8:
|
||||||
|
/* FIXME: needs descrambling */
|
||||||
|
caps = gst_caps_new_simple ("audio/x-pn-realaudio", "raversion",
|
||||||
|
G_TYPE_INT, 2, NULL);
|
||||||
|
codec_name = "Real Audio 28.8kbps";
|
||||||
|
break;
|
||||||
|
|
||||||
|
case GST_RM_AUD_DNET:
|
||||||
|
codec_name = "AC-3 audio";
|
||||||
|
caps = gst_caps_new_simple ("audio/x-ac3", "rate", G_TYPE_INT,
|
||||||
|
demux->sample_rate, NULL);
|
||||||
|
if (demux->packet_size == 0 || demux->sample_rate == 0)
|
||||||
|
goto broken_file;
|
||||||
|
demux->byterate_num = demux->packet_size * demux->sample_rate;
|
||||||
|
demux->byterate_denom = 1536;
|
||||||
|
break;
|
||||||
|
|
||||||
|
/* Sipro/ACELP.NET Voice Codec (MIME unknown) */
|
||||||
|
case GST_RM_AUD_SIPR:
|
||||||
|
codec_name = "Sipro Voice";
|
||||||
|
caps = gst_caps_new_simple ("audio/x-sipro", NULL);
|
||||||
|
break;
|
||||||
|
|
||||||
|
default:
|
||||||
|
GST_WARNING_OBJECT (demux, "unknown fourcc %08X", demux->fourcc);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
if (caps == NULL)
|
||||||
|
goto unknown_fourcc;
|
||||||
|
|
||||||
|
gst_caps_set_simple (caps,
|
||||||
|
"flavor", G_TYPE_INT, demux->flavour,
|
||||||
|
"rate", G_TYPE_INT, demux->sample_rate,
|
||||||
|
"channels", G_TYPE_INT, demux->channels,
|
||||||
|
"width", G_TYPE_INT, demux->sample_width,
|
||||||
|
"leaf_size", G_TYPE_INT, demux->leaf_size,
|
||||||
|
"packet_size", G_TYPE_INT, demux->packet_size,
|
||||||
|
"height", G_TYPE_INT, demux->height, NULL);
|
||||||
|
|
||||||
|
GST_INFO_OBJECT (demux, "Adding source pad, caps %" GST_PTR_FORMAT, caps);
|
||||||
|
demux->srcpad = gst_pad_new_from_static_template (&src_template, "src");
|
||||||
|
gst_pad_use_fixed_caps (demux->srcpad);
|
||||||
|
gst_pad_set_caps (demux->srcpad, caps);
|
||||||
|
gst_caps_unref (caps);
|
||||||
|
gst_pad_set_event_function (demux->srcpad,
|
||||||
|
GST_DEBUG_FUNCPTR (gst_real_audio_demux_src_event));
|
||||||
|
gst_pad_set_query_function (demux->srcpad,
|
||||||
|
GST_DEBUG_FUNCPTR (gst_real_audio_demux_src_query));
|
||||||
|
gst_pad_set_active (demux->srcpad, TRUE);
|
||||||
|
gst_element_add_pad (GST_ELEMENT (demux), demux->srcpad);
|
||||||
|
|
||||||
|
if (demux->byterate_num > 0 && demux->byterate_denom > 0) {
|
||||||
|
GstFormat bformat = GST_FORMAT_BYTES;
|
||||||
|
gint64 size_bytes = 0;
|
||||||
|
|
||||||
|
GST_INFO_OBJECT (demux, "byte rate = %u/%u = %u bytes/sec",
|
||||||
|
demux->byterate_num, demux->byterate_denom,
|
||||||
|
demux->byterate_num / demux->byterate_denom);
|
||||||
|
|
||||||
|
if (gst_pad_query_peer_duration (demux->sinkpad, &bformat, &size_bytes)) {
|
||||||
|
demux->duration =
|
||||||
|
gst_real_demux_get_timestamp_from_offset (demux, size_bytes);
|
||||||
|
demux->upstream_size = size_bytes;
|
||||||
|
GST_INFO_OBJECT (demux, "upstream_size = %" G_GUINT64_FORMAT,
|
||||||
|
demux->upstream_size);
|
||||||
|
GST_INFO_OBJECT (demux, "duration = %" GST_TIME_FORMAT,
|
||||||
|
GST_TIME_ARGS (demux->duration));
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
demux->need_newsegment = TRUE;
|
||||||
|
|
||||||
|
if (codec_name) {
|
||||||
|
if (demux->pending_tags == NULL)
|
||||||
|
demux->pending_tags = gst_tag_list_new ();
|
||||||
|
|
||||||
|
gst_tag_list_add (demux->pending_tags, GST_TAG_MERGE_REPLACE,
|
||||||
|
GST_TAG_AUDIO_CODEC, codec_name, NULL);
|
||||||
|
}
|
||||||
|
|
||||||
|
gst_adapter_flush (demux->adapter, demux->data_offset - 6);
|
||||||
|
|
||||||
|
demux->state = REAL_AUDIO_DEMUX_STATE_DATA;
|
||||||
|
demux->need_newsegment = TRUE;
|
||||||
|
|
||||||
|
return GST_FLOW_OK;
|
||||||
|
|
||||||
|
/* ERRORS */
|
||||||
|
unknown_fourcc:
|
||||||
|
{
|
||||||
|
GST_ELEMENT_ERROR (GST_ELEMENT (demux), STREAM, DECODE, (NULL),
|
||||||
|
("Unknown fourcc '%" GST_FOURCC_FORMAT "'",
|
||||||
|
GST_FOURCC_ARGS (demux->fourcc)));
|
||||||
|
return GST_FLOW_ERROR;
|
||||||
|
}
|
||||||
|
broken_file:
|
||||||
|
{
|
||||||
|
GST_ELEMENT_ERROR (GST_ELEMENT (demux), STREAM, DECODE, (NULL),
|
||||||
|
("Broken file - invalid sample_rate or other header value"));
|
||||||
|
return GST_FLOW_ERROR;
|
||||||
|
}
|
||||||
|
|
||||||
|
}
|
||||||
|
|
||||||
|
static GstFlowReturn
|
||||||
|
gst_real_audio_demux_parse_data (GstRealAudioDemux * demux)
|
||||||
|
{
|
||||||
|
GstFlowReturn ret = GST_FLOW_OK;
|
||||||
|
guint avail, unit_size;
|
||||||
|
|
||||||
|
avail = gst_adapter_available (demux->adapter);
|
||||||
|
|
||||||
|
if (demux->packet_size > 0)
|
||||||
|
unit_size = demux->packet_size;
|
||||||
|
else
|
||||||
|
unit_size = avail & 0xfffffff0; /* round down to next multiple of 16 */
|
||||||
|
|
||||||
|
GST_LOG_OBJECT (demux, "available = %u, unit_size = %u", avail, unit_size);
|
||||||
|
|
||||||
|
while (ret == GST_FLOW_OK && unit_size > 0 && avail >= unit_size) {
|
||||||
|
GstClockTime ts;
|
||||||
|
const guint8 *data;
|
||||||
|
GstBuffer *buf = NULL;
|
||||||
|
|
||||||
|
ret = gst_pad_alloc_buffer_and_set_caps (demux->srcpad,
|
||||||
|
GST_BUFFER_OFFSET_NONE, unit_size, GST_PAD_CAPS (demux->srcpad), &buf);
|
||||||
|
|
||||||
|
if (ret != GST_FLOW_OK) {
|
||||||
|
GST_DEBUG_OBJECT (demux, "pad_alloc flow: %s", gst_flow_get_name (ret));
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
data = gst_adapter_peek (demux->adapter, unit_size);
|
||||||
|
memcpy (GST_BUFFER_DATA (buf), data, unit_size);
|
||||||
|
gst_adapter_flush (demux->adapter, unit_size);
|
||||||
|
avail -= unit_size;
|
||||||
|
|
||||||
|
if (demux->need_newsegment) {
|
||||||
|
gst_pad_push_event (demux->srcpad,
|
||||||
|
gst_event_new_new_segment_full (FALSE, demux->segment.rate,
|
||||||
|
demux->segment.applied_rate, GST_FORMAT_TIME,
|
||||||
|
demux->segment.start, demux->segment.stop, demux->segment.time));
|
||||||
|
demux->need_newsegment = FALSE;
|
||||||
|
}
|
||||||
|
|
||||||
|
if (demux->pending_tags) {
|
||||||
|
gst_element_found_tags_for_pad (GST_ELEMENT (demux), demux->srcpad,
|
||||||
|
demux->pending_tags);
|
||||||
|
demux->pending_tags = NULL;
|
||||||
|
}
|
||||||
|
|
||||||
|
if (demux->fourcc == GST_RM_AUD_DNET) {
|
||||||
|
buf = gst_rm_utils_descramble_dnet_buffer (buf);
|
||||||
|
}
|
||||||
|
|
||||||
|
ts = gst_real_demux_get_timestamp_from_offset (demux, demux->offset);
|
||||||
|
GST_BUFFER_TIMESTAMP (buf) = ts;
|
||||||
|
|
||||||
|
gst_segment_set_last_stop (&demux->segment, GST_FORMAT_TIME, ts);
|
||||||
|
|
||||||
|
ret = gst_pad_push (demux->srcpad, buf);
|
||||||
|
}
|
||||||
|
|
||||||
|
return ret;
|
||||||
|
}
|
||||||
|
|
||||||
|
static GstFlowReturn
|
||||||
|
gst_real_audio_demux_handle_buffer (GstRealAudioDemux * demux, GstBuffer * buf)
|
||||||
|
{
|
||||||
|
GstFlowReturn ret;
|
||||||
|
|
||||||
|
gst_adapter_push (demux->adapter, buf);
|
||||||
|
buf = NULL;
|
||||||
|
|
||||||
|
switch (demux->state) {
|
||||||
|
case REAL_AUDIO_DEMUX_STATE_MARKER:{
|
||||||
|
ret = gst_real_audio_demux_parse_marker (demux);
|
||||||
|
if (ret != GST_FLOW_OK || demux->state != REAL_AUDIO_DEMUX_STATE_HEADER)
|
||||||
|
break;
|
||||||
|
/* otherwise fall through */
|
||||||
|
}
|
||||||
|
case REAL_AUDIO_DEMUX_STATE_HEADER:{
|
||||||
|
ret = gst_real_audio_demux_parse_header (demux);
|
||||||
|
if (ret != GST_FLOW_OK || demux->state != REAL_AUDIO_DEMUX_STATE_DATA)
|
||||||
|
break;
|
||||||
|
/* otherwise fall through */
|
||||||
|
}
|
||||||
|
case REAL_AUDIO_DEMUX_STATE_DATA:{
|
||||||
|
ret = gst_real_audio_demux_parse_data (demux);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
default:
|
||||||
|
g_assert_not_reached ();
|
||||||
|
}
|
||||||
|
|
||||||
|
return ret;
|
||||||
|
}
|
||||||
|
|
||||||
|
static GstFlowReturn
|
||||||
|
gst_real_audio_demux_chain (GstPad * pad, GstBuffer * buf)
|
||||||
|
{
|
||||||
|
GstRealAudioDemux *demux;
|
||||||
|
|
||||||
|
demux = GST_REAL_AUDIO_DEMUX (GST_PAD_PARENT (pad));
|
||||||
|
|
||||||
|
return gst_real_audio_demux_handle_buffer (demux, buf);
|
||||||
|
}
|
||||||
|
|
||||||
|
static void
|
||||||
|
gst_real_audio_demux_loop (GstRealAudioDemux * demux)
|
||||||
|
{
|
||||||
|
GstRealAudioDemuxState old_state;
|
||||||
|
GstFlowReturn ret;
|
||||||
|
GstBuffer *buf;
|
||||||
|
guint bytes_needed;
|
||||||
|
|
||||||
|
old_state = demux->state;
|
||||||
|
|
||||||
|
/* check how much data we need */
|
||||||
|
switch (demux->state) {
|
||||||
|
case REAL_AUDIO_DEMUX_STATE_MARKER:
|
||||||
|
bytes_needed = 6 + 16; /* 16 are beginning of header */
|
||||||
|
break;
|
||||||
|
case REAL_AUDIO_DEMUX_STATE_HEADER:
|
||||||
|
if (!gst_real_audio_demux_get_data_offset_from_header (demux))
|
||||||
|
goto parse_header_error;
|
||||||
|
bytes_needed = demux->data_offset - (6 + 16);
|
||||||
|
break;
|
||||||
|
case REAL_AUDIO_DEMUX_STATE_DATA:
|
||||||
|
if (demux->packet_size > 0) {
|
||||||
|
/* TODO: should probably take into account width/height as well? */
|
||||||
|
bytes_needed = demux->packet_size;
|
||||||
|
} else {
|
||||||
|
bytes_needed = 1024;
|
||||||
|
}
|
||||||
|
break;
|
||||||
|
default:
|
||||||
|
g_assert_not_reached ();
|
||||||
|
}
|
||||||
|
|
||||||
|
/* now get the data */
|
||||||
|
GST_LOG_OBJECT (demux, "getting data: %5u bytes @ %8" G_GINT64_MODIFIER "u",
|
||||||
|
bytes_needed, demux->offset);
|
||||||
|
|
||||||
|
if (demux->upstream_size > 0 && demux->offset >= demux->upstream_size)
|
||||||
|
goto eos;
|
||||||
|
|
||||||
|
ret = gst_pad_pull_range (demux->sinkpad, demux->offset, bytes_needed, &buf);
|
||||||
|
|
||||||
|
if (ret != GST_FLOW_OK)
|
||||||
|
goto pull_range_error;
|
||||||
|
|
||||||
|
if (GST_BUFFER_SIZE (buf) != bytes_needed)
|
||||||
|
goto pull_range_short_read;
|
||||||
|
|
||||||
|
ret = gst_real_audio_demux_handle_buffer (demux, buf);
|
||||||
|
if (ret != GST_FLOW_OK)
|
||||||
|
goto handle_flow_error;
|
||||||
|
|
||||||
|
/* TODO: increase this in chain function too (for timestamps)? */
|
||||||
|
demux->offset += bytes_needed;
|
||||||
|
|
||||||
|
/* check for the end of the segment */
|
||||||
|
if (demux->segment.stop != -1 && demux->segment.last_stop != -1 &&
|
||||||
|
demux->segment.last_stop > demux->segment.stop) {
|
||||||
|
GST_DEBUG_OBJECT (demux, "reached end of segment");
|
||||||
|
goto eos;
|
||||||
|
}
|
||||||
|
|
||||||
|
return;
|
||||||
|
|
||||||
|
/* ERRORS */
|
||||||
|
parse_header_error:
|
||||||
|
{
|
||||||
|
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL), (NULL));
|
||||||
|
goto pause_task;
|
||||||
|
}
|
||||||
|
handle_flow_error:
|
||||||
|
{
|
||||||
|
GST_WARNING_OBJECT (demux, "handle_buf flow: %s", gst_flow_get_name (ret));
|
||||||
|
goto pause_task;
|
||||||
|
}
|
||||||
|
pull_range_error:
|
||||||
|
{
|
||||||
|
GST_WARNING_OBJECT (demux, "pull range flow: %s", gst_flow_get_name (ret));
|
||||||
|
goto pause_task;
|
||||||
|
}
|
||||||
|
pull_range_short_read:
|
||||||
|
{
|
||||||
|
GST_WARNING_OBJECT (demux, "pull range short read: wanted %u bytes, but "
|
||||||
|
"got only %u bytes", bytes_needed, GST_BUFFER_SIZE (buf));
|
||||||
|
gst_buffer_unref (buf);
|
||||||
|
goto eos;
|
||||||
|
}
|
||||||
|
eos:
|
||||||
|
{
|
||||||
|
if (demux->state != REAL_AUDIO_DEMUX_STATE_DATA) {
|
||||||
|
GST_WARNING_OBJECT (demux, "reached EOS before finished parsing header");
|
||||||
|
goto parse_header_error;
|
||||||
|
}
|
||||||
|
GST_INFO_OBJECT (demux, "EOS");
|
||||||
|
if ((demux->segment.flags & GST_SEEK_FLAG_SEGMENT) != 0) {
|
||||||
|
gint64 stop;
|
||||||
|
|
||||||
|
/* for segment playback we need to post when (in stream time)
|
||||||
|
* we stopped, this is either stop (when set) or the duration. */
|
||||||
|
if ((stop = demux->segment.stop) == -1)
|
||||||
|
stop = demux->segment.duration;
|
||||||
|
|
||||||
|
GST_DEBUG_OBJECT (demux, "sending segment done, at end of segment");
|
||||||
|
gst_element_post_message (GST_ELEMENT (demux),
|
||||||
|
gst_message_new_segment_done (GST_OBJECT (demux), GST_FORMAT_TIME,
|
||||||
|
stop));
|
||||||
|
} else {
|
||||||
|
/* normal playback, send EOS event downstream */
|
||||||
|
GST_DEBUG_OBJECT (demux, "sending EOS event, at end of stream");
|
||||||
|
gst_pad_push_event (demux->srcpad, gst_event_new_eos ());
|
||||||
|
}
|
||||||
|
goto pause_task;
|
||||||
|
}
|
||||||
|
pause_task:
|
||||||
|
{
|
||||||
|
demux->segment_running = FALSE;
|
||||||
|
gst_pad_pause_task (demux->sinkpad);
|
||||||
|
GST_DEBUG_OBJECT (demux, "pausing task");
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
static gboolean
|
||||||
|
gst_real_audio_demux_sink_event (GstPad * pad, GstEvent * event)
|
||||||
|
{
|
||||||
|
GstRealAudioDemux *demux;
|
||||||
|
gboolean ret;
|
||||||
|
|
||||||
|
demux = GST_REAL_AUDIO_DEMUX (gst_pad_get_parent (pad));
|
||||||
|
|
||||||
|
switch (GST_EVENT_TYPE (event)) {
|
||||||
|
case GST_EVENT_NEWSEGMENT:{
|
||||||
|
/* FIXME */
|
||||||
|
gst_event_unref (event);
|
||||||
|
demux->need_newsegment = TRUE;
|
||||||
|
ret = TRUE;
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
default:
|
||||||
|
ret = gst_pad_event_default (pad, event);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
gst_object_unref (demux);
|
||||||
|
return ret;
|
||||||
|
}
|
||||||
|
|
||||||
|
static gboolean
|
||||||
|
gst_real_audio_demux_handle_seek (GstRealAudioDemux * demux, GstEvent * event)
|
||||||
|
{
|
||||||
|
GstFormat format;
|
||||||
|
GstSeekFlags flags;
|
||||||
|
GstSeekType cur_type, stop_type;
|
||||||
|
gboolean flush, update;
|
||||||
|
gdouble rate;
|
||||||
|
guint64 seek_pos;
|
||||||
|
gint64 cur, stop;
|
||||||
|
|
||||||
|
if (!demux->seekable)
|
||||||
|
goto not_seekable;
|
||||||
|
|
||||||
|
if (demux->byterate_num == 0 || demux->byterate_denom == 0)
|
||||||
|
goto no_bitrate;
|
||||||
|
|
||||||
|
gst_event_parse_seek (event, &rate, &format, &flags,
|
||||||
|
&cur_type, &cur, &stop_type, &stop);
|
||||||
|
|
||||||
|
if (format != GST_FORMAT_TIME)
|
||||||
|
goto only_time_format_supported;
|
||||||
|
|
||||||
|
if (rate <= 0.0)
|
||||||
|
goto cannot_do_backwards_playback;
|
||||||
|
|
||||||
|
flush = ((flags & GST_SEEK_FLAG_FLUSH) != 0);
|
||||||
|
|
||||||
|
GST_DEBUG_OBJECT (demux, "flush=%d, rate=%g", flush, rate);
|
||||||
|
|
||||||
|
/* unlock streaming thread and make streaming stop */
|
||||||
|
if (flush) {
|
||||||
|
gst_pad_push_event (demux->sinkpad, gst_event_new_flush_start ());
|
||||||
|
gst_pad_push_event (demux->srcpad, gst_event_new_flush_start ());
|
||||||
|
} else {
|
||||||
|
gst_pad_pause_task (demux->sinkpad);
|
||||||
|
}
|
||||||
|
|
||||||
|
GST_PAD_STREAM_LOCK (demux->sinkpad);
|
||||||
|
|
||||||
|
if (demux->segment_running && !flush) {
|
||||||
|
GstEvent *newseg;
|
||||||
|
|
||||||
|
newseg = gst_event_new_new_segment_full (TRUE, demux->segment.rate,
|
||||||
|
demux->segment.applied_rate, GST_FORMAT_TIME, demux->segment.start,
|
||||||
|
demux->segment.last_stop, demux->segment.time);
|
||||||
|
|
||||||
|
GST_DEBUG_OBJECT (demux, "sending NEWSEGMENT event to close the current "
|
||||||
|
"segment: %" GST_PTR_FORMAT, newseg);
|
||||||
|
|
||||||
|
gst_pad_push_event (demux->srcpad, newseg);
|
||||||
|
}
|
||||||
|
|
||||||
|
gst_segment_set_seek (&demux->segment, rate, format, flags,
|
||||||
|
cur_type, cur, stop_type, stop, &update);
|
||||||
|
|
||||||
|
GST_DEBUG_OBJECT (demux, "segment: %" GST_SEGMENT_FORMAT, &demux->segment);
|
||||||
|
|
||||||
|
seek_pos = gst_util_uint64_scale (demux->segment.start,
|
||||||
|
demux->byterate_num, demux->byterate_denom * GST_SECOND);
|
||||||
|
if (demux->packet_size > 0) {
|
||||||
|
seek_pos -= seek_pos % demux->packet_size;
|
||||||
|
}
|
||||||
|
seek_pos += demux->data_offset;
|
||||||
|
|
||||||
|
GST_DEBUG_OBJECT (demux, "seek_pos = %" G_GUINT64_FORMAT, seek_pos);
|
||||||
|
|
||||||
|
/* stop flushing */
|
||||||
|
gst_pad_push_event (demux->sinkpad, gst_event_new_flush_stop ());
|
||||||
|
gst_pad_push_event (demux->srcpad, gst_event_new_flush_stop ());
|
||||||
|
|
||||||
|
demux->offset = seek_pos;
|
||||||
|
demux->need_newsegment = TRUE;
|
||||||
|
|
||||||
|
/* notify start of new segment */
|
||||||
|
if (demux->segment.flags & GST_SEEK_FLAG_SEGMENT) {
|
||||||
|
gst_element_post_message (GST_ELEMENT (demux),
|
||||||
|
gst_message_new_segment_start (GST_OBJECT (demux),
|
||||||
|
GST_FORMAT_TIME, demux->segment.last_stop));
|
||||||
|
}
|
||||||
|
|
||||||
|
demux->segment_running = TRUE;
|
||||||
|
/* restart our task since it might have been stopped when we did the flush */
|
||||||
|
gst_pad_start_task (demux->sinkpad,
|
||||||
|
(GstTaskFunction) gst_real_audio_demux_loop, demux);
|
||||||
|
|
||||||
|
/* streaming can continue now */
|
||||||
|
GST_PAD_STREAM_UNLOCK (demux->sinkpad);
|
||||||
|
|
||||||
|
return TRUE;
|
||||||
|
|
||||||
|
/* ERRORS */
|
||||||
|
not_seekable:
|
||||||
|
{
|
||||||
|
GST_DEBUG_OBJECT (demux, "seek failed: cannot seek in streaming mode");
|
||||||
|
return FALSE;
|
||||||
|
}
|
||||||
|
no_bitrate:
|
||||||
|
{
|
||||||
|
GST_DEBUG_OBJECT (demux, "seek failed: bitrate unknown");
|
||||||
|
return FALSE;
|
||||||
|
}
|
||||||
|
only_time_format_supported:
|
||||||
|
{
|
||||||
|
GST_DEBUG_OBJECT (demux, "can only seek in TIME format");
|
||||||
|
return FALSE;
|
||||||
|
}
|
||||||
|
cannot_do_backwards_playback:
|
||||||
|
{
|
||||||
|
GST_DEBUG_OBJECT (demux, "can only seek with positive rate, not %lf", rate);
|
||||||
|
return FALSE;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
static gboolean
|
||||||
|
gst_real_audio_demux_src_event (GstPad * pad, GstEvent * event)
|
||||||
|
{
|
||||||
|
GstRealAudioDemux *demux;
|
||||||
|
gboolean ret = FALSE;
|
||||||
|
|
||||||
|
demux = GST_REAL_AUDIO_DEMUX (gst_pad_get_parent (pad));
|
||||||
|
|
||||||
|
switch (GST_EVENT_TYPE (event)) {
|
||||||
|
case GST_EVENT_QOS:
|
||||||
|
gst_event_unref (event);
|
||||||
|
break;
|
||||||
|
case GST_EVENT_SEEK:
|
||||||
|
ret = gst_real_audio_demux_handle_seek (demux, event);
|
||||||
|
gst_event_unref (event);
|
||||||
|
break;
|
||||||
|
default:
|
||||||
|
ret = gst_pad_event_default (pad, event);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
gst_object_unref (demux);
|
||||||
|
return ret;
|
||||||
|
}
|
||||||
|
|
||||||
|
static gboolean
|
||||||
|
gst_real_audio_demux_src_query (GstPad * pad, GstQuery * query)
|
||||||
|
{
|
||||||
|
GstRealAudioDemux *demux;
|
||||||
|
gboolean ret = FALSE;
|
||||||
|
|
||||||
|
demux = GST_REAL_AUDIO_DEMUX (gst_pad_get_parent (pad));
|
||||||
|
|
||||||
|
switch (GST_QUERY_TYPE (query)) {
|
||||||
|
case GST_QUERY_DURATION:{
|
||||||
|
GstFormat format;
|
||||||
|
|
||||||
|
gst_query_parse_duration (query, &format, NULL);
|
||||||
|
if (format == GST_FORMAT_TIME && demux->duration > 0) {
|
||||||
|
gst_query_set_duration (query, GST_FORMAT_TIME, demux->duration);
|
||||||
|
ret = TRUE;
|
||||||
|
} else if (format == GST_FORMAT_BYTES && demux->upstream_size > 0) {
|
||||||
|
gst_query_set_duration (query, GST_FORMAT_BYTES,
|
||||||
|
demux->upstream_size - demux->data_offset);
|
||||||
|
ret = TRUE;
|
||||||
|
}
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
case GST_QUERY_SEEKING:{
|
||||||
|
GstFormat format;
|
||||||
|
gboolean seekable;
|
||||||
|
|
||||||
|
gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
|
||||||
|
seekable = (format == GST_FORMAT_TIME && demux->seekable);
|
||||||
|
gst_query_set_seeking (query, format, seekable, 0,
|
||||||
|
(format == GST_FORMAT_TIME) ? demux->duration : -1);
|
||||||
|
ret = TRUE;
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
default:
|
||||||
|
ret = gst_pad_query_default (pad, query);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
gst_object_unref (demux);
|
||||||
|
return ret;
|
||||||
|
}
|
||||||
|
|
||||||
|
static GstStateChangeReturn
|
||||||
|
gst_real_audio_demux_change_state (GstElement * element,
|
||||||
|
GstStateChange transition)
|
||||||
|
{
|
||||||
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
||||||
|
GstRealAudioDemux *demux = GST_REAL_AUDIO_DEMUX (element);
|
||||||
|
|
||||||
|
switch (transition) {
|
||||||
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
||||||
|
break;
|
||||||
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||||
|
demux->state = REAL_AUDIO_DEMUX_STATE_MARKER;
|
||||||
|
demux->segment_running = FALSE;
|
||||||
|
gst_segment_init (&demux->segment, GST_FORMAT_TIME);
|
||||||
|
gst_adapter_clear (demux->adapter);
|
||||||
|
break;
|
||||||
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||||
|
break;
|
||||||
|
default:
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
||||||
|
|
||||||
|
switch (transition) {
|
||||||
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||||
|
break;
|
||||||
|
case GST_STATE_CHANGE_PAUSED_TO_READY:{
|
||||||
|
gst_real_audio_demux_reset (demux);
|
||||||
|
gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
||||||
|
break;
|
||||||
|
default:
|
||||||
|
break;
|
||||||
|
}
|
||||||
|
|
||||||
|
return ret;
|
||||||
|
}
|
||||||
99
gst/realmedia/rademux.h
Normal file
99
gst/realmedia/rademux.h
Normal file
|
|
@ -0,0 +1,99 @@
|
||||||
|
/* GStreamer RealAudio demuxer
|
||||||
|
* Copyright (C) 2006 Tim-Philipp Müller <tim centricular net>
|
||||||
|
*
|
||||||
|
* 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 __GST_REAL_AUDIO_DEMUX_H__
|
||||||
|
#define __GST_REAL_AUDIO_DEMUX_H__
|
||||||
|
|
||||||
|
#include <gst/gst.h>
|
||||||
|
#include <gst/base/gstadapter.h>
|
||||||
|
|
||||||
|
G_BEGIN_DECLS
|
||||||
|
|
||||||
|
#define GST_TYPE_REAL_AUDIO_DEMUX \
|
||||||
|
(gst_real_audio_demux_get_type())
|
||||||
|
#define GST_REAL_AUDIO_DEMUX(obj) \
|
||||||
|
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_REAL_AUDIO_DEMUX,GstRealAudioDemux))
|
||||||
|
#define GST_REAL_AUDIO_DEMUX_CLASS(klass) \
|
||||||
|
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_REAL_AUDIO_DEMUX,GstRealAudioDemuxClass))
|
||||||
|
#define GST_IS_REAL_AUDIO_DEMUX(obj) \
|
||||||
|
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_REAL_AUDIO_DEMUX))
|
||||||
|
#define GST_IS_REAL_AUDIO_DEMUX_CLASS(klass) \
|
||||||
|
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_REAL_AUDIO_DEMUX))
|
||||||
|
|
||||||
|
typedef enum
|
||||||
|
{
|
||||||
|
REAL_AUDIO_DEMUX_STATE_MARKER,
|
||||||
|
REAL_AUDIO_DEMUX_STATE_HEADER,
|
||||||
|
REAL_AUDIO_DEMUX_STATE_DATA
|
||||||
|
} GstRealAudioDemuxState;
|
||||||
|
|
||||||
|
typedef struct _GstRealAudioDemux GstRealAudioDemux;
|
||||||
|
typedef struct _GstRealAudioDemuxClass GstRealAudioDemuxClass;
|
||||||
|
|
||||||
|
struct _GstRealAudioDemux {
|
||||||
|
GstElement element;
|
||||||
|
|
||||||
|
GstPad *sinkpad;
|
||||||
|
GstPad *srcpad;
|
||||||
|
|
||||||
|
GstAdapter *adapter;
|
||||||
|
GstRealAudioDemuxState state;
|
||||||
|
|
||||||
|
guint ra_version;
|
||||||
|
guint data_offset;
|
||||||
|
|
||||||
|
guint packet_size;
|
||||||
|
guint leaf_size;
|
||||||
|
guint height;
|
||||||
|
guint flavour;
|
||||||
|
|
||||||
|
guint sample_rate;
|
||||||
|
guint sample_width;
|
||||||
|
guint channels;
|
||||||
|
guint32 fourcc;
|
||||||
|
|
||||||
|
gboolean segment_running;
|
||||||
|
|
||||||
|
gboolean need_newsegment;
|
||||||
|
GstTagList *pending_tags;
|
||||||
|
|
||||||
|
guint byterate_num; /* bytes per second */
|
||||||
|
guint byterate_denom;
|
||||||
|
|
||||||
|
gint64 duration;
|
||||||
|
gint64 upstream_size;
|
||||||
|
|
||||||
|
guint64 offset; /* current read byte offset for
|
||||||
|
* pull_range-based mode */
|
||||||
|
|
||||||
|
/* playback start/stop positions */
|
||||||
|
GstSegment segment;
|
||||||
|
|
||||||
|
gboolean seekable;
|
||||||
|
};
|
||||||
|
|
||||||
|
struct _GstRealAudioDemuxClass {
|
||||||
|
GstElementClass element_class;
|
||||||
|
};
|
||||||
|
|
||||||
|
GType gst_real_audio_demux_get_type (void);
|
||||||
|
|
||||||
|
G_END_DECLS
|
||||||
|
|
||||||
|
#endif /* __GST_REAL_AUDIO_DEMUX_H__ */
|
||||||
|
|
@ -26,9 +26,12 @@
|
||||||
#ifdef HAVE_CONFIG_H
|
#ifdef HAVE_CONFIG_H
|
||||||
# include "config.h"
|
# include "config.h"
|
||||||
#endif
|
#endif
|
||||||
|
|
||||||
#include "rmdemux.h"
|
#include "rmdemux.h"
|
||||||
#include "rdtdepay.h"
|
#include "rdtdepay.h"
|
||||||
#include "rmutils.h"
|
#include "rmutils.h"
|
||||||
|
#include "rademux.h"
|
||||||
|
|
||||||
#include <string.h>
|
#include <string.h>
|
||||||
#include <ctype.h>
|
#include <ctype.h>
|
||||||
#include <zlib.h>
|
#include <zlib.h>
|
||||||
|
|
@ -1405,6 +1408,9 @@ gst_rmdemux_add_stream (GstRMDemux * rmdemux, GstRMDemuxStream * stream)
|
||||||
rmdemux->n_streams++;
|
rmdemux->n_streams++;
|
||||||
GST_LOG_OBJECT (rmdemux, "n_streams is now %d", rmdemux->n_streams);
|
GST_LOG_OBJECT (rmdemux, "n_streams is now %d", rmdemux->n_streams);
|
||||||
|
|
||||||
|
GST_LOG ("stream->pad = %p, stream_caps = %" GST_PTR_FORMAT, stream->pad,
|
||||||
|
stream_caps);
|
||||||
|
|
||||||
if (stream->pad && stream_caps) {
|
if (stream->pad && stream_caps) {
|
||||||
|
|
||||||
GST_LOG_OBJECT (rmdemux, "%d bytes of extra data for stream %s",
|
GST_LOG_OBJECT (rmdemux, "%d bytes of extra data for stream %s",
|
||||||
|
|
@ -1893,21 +1899,12 @@ gst_rmdemux_descramble_dnet_audio (GstRMDemux * rmdemux,
|
||||||
GstRMDemuxStream * stream)
|
GstRMDemuxStream * stream)
|
||||||
{
|
{
|
||||||
GstBuffer *buf;
|
GstBuffer *buf;
|
||||||
guint8 *data, *end;
|
|
||||||
|
|
||||||
buf = g_ptr_array_index (stream->subpackets, 0);
|
buf = g_ptr_array_index (stream->subpackets, 0);
|
||||||
g_ptr_array_index (stream->subpackets, 0) = NULL;
|
g_ptr_array_index (stream->subpackets, 0) = NULL;
|
||||||
g_ptr_array_set_size (stream->subpackets, 0);
|
g_ptr_array_set_size (stream->subpackets, 0);
|
||||||
|
|
||||||
/* descramble is a bit of a misnomer, it's just byte-order swapped AC3 .. */
|
buf = gst_rm_utils_descramble_dnet_buffer (buf);
|
||||||
data = GST_BUFFER_DATA (buf);
|
|
||||||
end = GST_BUFFER_DATA (buf) + GST_BUFFER_SIZE (buf);
|
|
||||||
while ((data + 1) < end) {
|
|
||||||
/* byte-swap in an alignment-safe way */
|
|
||||||
GST_WRITE_UINT16_BE (data, GST_READ_UINT16_LE (data));
|
|
||||||
data += sizeof (guint16);
|
|
||||||
}
|
|
||||||
|
|
||||||
return gst_pad_push (stream->pad, buf);
|
return gst_pad_push (stream->pad, buf);
|
||||||
}
|
}
|
||||||
|
|
||||||
|
|
@ -2037,11 +2034,13 @@ alloc_failed:
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
gboolean
|
static gboolean
|
||||||
gst_rmdemux_plugin_init (GstPlugin * plugin)
|
gst_rmdemux_plugin_init (GstPlugin * plugin)
|
||||||
{
|
{
|
||||||
return gst_element_register (plugin, "rmdemux",
|
return gst_element_register (plugin, "rmdemux",
|
||||||
GST_RANK_PRIMARY, GST_TYPE_RMDEMUX);
|
GST_RANK_PRIMARY, GST_TYPE_RMDEMUX) &&
|
||||||
|
gst_element_register (plugin, "rademux",
|
||||||
|
GST_RANK_SECONDARY, GST_TYPE_REAL_AUDIO_DEMUX);
|
||||||
}
|
}
|
||||||
|
|
||||||
|
|
||||||
|
|
|
||||||
|
|
@ -120,3 +120,21 @@ gst_rm_utils_read_tags (const guint8 * data, guint datalen,
|
||||||
gst_tag_list_free (tags);
|
gst_tag_list_free (tags);
|
||||||
return NULL;
|
return NULL;
|
||||||
}
|
}
|
||||||
|
|
||||||
|
GstBuffer *
|
||||||
|
gst_rm_utils_descramble_dnet_buffer (GstBuffer * buf)
|
||||||
|
{
|
||||||
|
guint8 *data, *end;
|
||||||
|
|
||||||
|
buf = gst_buffer_make_writable (buf);
|
||||||
|
|
||||||
|
/* dnet = byte-order swapped AC3 */
|
||||||
|
data = GST_BUFFER_DATA (buf);
|
||||||
|
end = GST_BUFFER_DATA (buf) + GST_BUFFER_SIZE (buf);
|
||||||
|
while ((data + 1) < end) {
|
||||||
|
/* byte-swap in an alignment-safe way */
|
||||||
|
GST_WRITE_UINT16_BE (data, GST_READ_UINT16_LE (data));
|
||||||
|
data += sizeof (guint16);
|
||||||
|
}
|
||||||
|
return buf;
|
||||||
|
}
|
||||||
|
|
|
||||||
|
|
@ -38,6 +38,8 @@ GstTagList *gst_rm_utils_read_tags (const guint8 * data,
|
||||||
guint datalen,
|
guint datalen,
|
||||||
GstRmUtilsStringReadFunc func);
|
GstRmUtilsStringReadFunc func);
|
||||||
|
|
||||||
|
GstBuffer *gst_rm_utils_descramble_dnet_buffer (GstBuffer * buf);
|
||||||
|
|
||||||
G_END_DECLS
|
G_END_DECLS
|
||||||
|
|
||||||
#endif /* __GST_RM_UTILS_H__ */
|
#endif /* __GST_RM_UTILS_H__ */
|
||||||
|
|
|
||||||
Loading…
Reference in a new issue