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1000 lines
29 KiB
C
1000 lines
29 KiB
C
/* 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rademux
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* @title: rademux
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*
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* Demuxes/parses a RealAudio (.ra) file or stream into compressed audio.
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*
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* ## Example launch line
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* |[
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* gst-launch-1.0 filesrc location=interview.ra ! rademux ! avdec_real_288 ! audioconvert ! audioresample ! autoaudiosink
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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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* |[
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* gst-launch-1.0 souphttpsrc location=http://www.example.org/interview.ra ! rademux ! ac3parse ! a52dec ! audioconvert ! audioresample ! autoaudiosink
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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.
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*
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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 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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#define gst_real_audio_demux_parent_class parent_class
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G_DEFINE_TYPE (GstRealAudioDemux, gst_real_audio_demux, GST_TYPE_ELEMENT);
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GST_ELEMENT_REGISTER_DEFINE (rademux, "rademux",
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GST_RANK_SECONDARY, GST_TYPE_REAL_AUDIO_DEMUX);
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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,
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GstObject * parent, GstBuffer * buf);
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static gboolean gst_real_audio_demux_sink_event (GstPad * pad,
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GstObject * parent, GstEvent * ev);
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static gboolean gst_real_audio_demux_src_event (GstPad * pad,
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GstObject * parent, GstEvent * ev);
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static gboolean gst_real_audio_demux_src_query (GstPad * pad,
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GstObject * parent, 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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GstObject * parent);
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static gboolean gst_real_audio_demux_sink_activate_mode (GstPad * sinkpad,
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GstObject * parent, GstPadMode mode, gboolean active);
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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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gst_element_class_add_static_pad_template (gstelement_class, &sink_template);
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gst_element_class_add_static_pad_template (gstelement_class, &src_template);
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gst_element_class_set_static_metadata (gstelement_class, "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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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_change_state);
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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_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_unref (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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demux->have_group_id = FALSE;
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demux->group_id = G_MAXUINT;
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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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{
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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_activatemode_function (demux->sinkpad,
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_activate_mode));
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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, GstObject * parent)
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{
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GstQuery *query;
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gboolean pull_mode;
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query = gst_query_new_scheduling ();
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if (!gst_pad_peer_query (sinkpad, query)) {
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gst_query_unref (query);
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goto activate_push;
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}
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pull_mode = gst_query_has_scheduling_mode_with_flags (query,
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GST_PAD_MODE_PULL, GST_SCHEDULING_FLAG_SEEKABLE);
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gst_query_unref (query);
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if (!pull_mode)
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goto activate_push;
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GST_DEBUG_OBJECT (sinkpad, "activating pull");
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return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PULL, TRUE);
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activate_push:
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{
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GST_DEBUG_OBJECT (sinkpad, "activating push");
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return gst_pad_activate_mode (sinkpad, GST_PAD_MODE_PUSH, 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_mode (GstPad * sinkpad, GstObject * parent,
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GstPadMode mode, gboolean active)
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{
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gboolean res;
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GstRealAudioDemux *demux;
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demux = GST_REAL_AUDIO_DEMUX (parent);
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switch (mode) {
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case GST_PAD_MODE_PUSH:
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demux->seekable = FALSE;
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res = TRUE;
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break;
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case GST_PAD_MODE_PULL:
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if (active) {
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demux->seekable = TRUE;
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res = gst_pad_start_task (sinkpad,
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(GstTaskFunction) gst_real_audio_demux_loop, demux, NULL);
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} else {
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demux->seekable = FALSE;
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res = gst_pad_stop_task (sinkpad);
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}
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break;
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default:
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res = FALSE;
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break;
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}
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return res;
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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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guint8 data[6];
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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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gst_adapter_copy (demux->adapter, data, 0, 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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guint8 data[16];
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gst_adapter_copy (demux->adapter, data, 0, 16);
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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;
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}
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static GstFlowReturn
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gst_real_audio_demux_parse_header (GstRealAudioDemux * demux)
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{
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const guint8 *data;
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gchar *codec_name = NULL;
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GstCaps *caps = NULL;
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GstEvent *event;
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gchar *stream_id;
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guint avail;
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g_assert (demux->ra_version == 4 || demux->ra_version == 3);
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avail = gst_adapter_available (demux->adapter);
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if (avail < 16)
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return GST_FLOW_OK;
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if (!gst_real_audio_demux_get_data_offset_from_header (demux))
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return GST_FLOW_ERROR; /* shouldn't happen */
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GST_DEBUG_OBJECT (demux, "data_offset = %u", demux->data_offset);
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if (avail + 6 < demux->data_offset) {
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GST_DEBUG_OBJECT (demux, "Need %u bytes, but only %u available now",
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demux->data_offset - 6, avail);
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return GST_FLOW_OK;
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}
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data = gst_adapter_map (demux->adapter, demux->data_offset - 6);
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g_assert (data);
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switch (demux->ra_version) {
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case 3:
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demux->fourcc = GST_RM_AUD_14_4;
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demux->packet_size = 20;
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demux->sample_rate = 8000;
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demux->channels = 1;
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demux->sample_width = 16;
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demux->flavour = 1;
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demux->leaf_size = 0;
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demux->height = 0;
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break;
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case 4:
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demux->flavour = GST_READ_UINT16_BE (data + 16);
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/* demux->frame_size = GST_READ_UINT32_BE (data + 36); */
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demux->leaf_size = GST_READ_UINT16_BE (data + 38);
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demux->height = GST_READ_UINT16_BE (data + 34);
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demux->packet_size = GST_READ_UINT32_BE (data + 18);
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demux->sample_rate = GST_READ_UINT16_BE (data + 42);
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demux->sample_width = GST_READ_UINT16_BE (data + 46);
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demux->channels = GST_READ_UINT16_BE (data + 48);
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demux->fourcc = GST_READ_UINT32_LE (data + 56);
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demux->pending_tags = gst_rm_utils_read_tags (data + 63,
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demux->data_offset - 63, gst_rm_utils_read_string8);
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if (demux->pending_tags)
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gst_tag_list_set_scope (demux->pending_tags, GST_TAG_SCOPE_GLOBAL);
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break;
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default:
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g_assert_not_reached ();
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#if 0
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case 5:
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demux->flavour = GST_READ_UINT16_BE (data + 16);
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/* demux->frame_size = GST_READ_UINT32_BE (data + 36); */
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demux->leaf_size = GST_READ_UINT16_BE (data + 38);
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demux->height = GST_READ_UINT16_BE (data + 34);
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demux->sample_rate = GST_READ_UINT16_BE (data + 48);
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demux->sample_width = GST_READ_UINT16_BE (data + 52);
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demux->n_channels = GST_READ_UINT16_BE (data + 54);
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demux->fourcc = RMDEMUX_FOURCC_GET (data + 60);
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break;
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#endif
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}
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GST_INFO_OBJECT (demux, "packet_size = %u", demux->packet_size);
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GST_INFO_OBJECT (demux, "sample_rate = %u", demux->sample_rate);
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GST_INFO_OBJECT (demux, "sample_width = %u", demux->sample_width);
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GST_INFO_OBJECT (demux, "channels = %u", demux->channels);
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GST_INFO_OBJECT (demux, "fourcc = '%" GST_FOURCC_FORMAT "' (%08X)",
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GST_FOURCC_ARGS (demux->fourcc), demux->fourcc);
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switch (demux->fourcc) {
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case GST_RM_AUD_14_4:
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caps = gst_caps_new_simple ("audio/x-pn-realaudio", "raversion",
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G_TYPE_INT, 1, NULL);
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demux->byterate_num = 1000;
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demux->byterate_denom = 1;
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break;
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case GST_RM_AUD_28_8:
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/* FIXME: needs descrambling */
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caps = gst_caps_new_simple ("audio/x-pn-realaudio", "raversion",
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G_TYPE_INT, 2, NULL);
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break;
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case GST_RM_AUD_DNET:
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caps = gst_caps_new_simple ("audio/x-ac3", "rate", G_TYPE_INT,
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demux->sample_rate, NULL);
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if (demux->packet_size == 0 || demux->sample_rate == 0)
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goto broken_file;
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demux->byterate_num = demux->packet_size * demux->sample_rate;
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demux->byterate_denom = 1536;
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break;
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/* Sipro/ACELP.NET Voice Codec (MIME unknown) */
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case GST_RM_AUD_SIPR:
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caps = gst_caps_new_empty_simple ("audio/x-sipro");
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break;
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default:
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GST_WARNING_OBJECT (demux, "unknown fourcc %08X", demux->fourcc);
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break;
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}
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if (caps == NULL)
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goto unknown_fourcc;
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gst_caps_set_simple (caps,
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"flavor", G_TYPE_INT, demux->flavour,
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"rate", G_TYPE_INT, demux->sample_rate,
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"channels", G_TYPE_INT, demux->channels,
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"width", G_TYPE_INT, demux->sample_width,
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"leaf_size", G_TYPE_INT, demux->leaf_size,
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"packet_size", G_TYPE_INT, demux->packet_size,
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"height", G_TYPE_INT, demux->height, NULL);
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GST_INFO_OBJECT (demux, "Adding source pad, caps %" GST_PTR_FORMAT, caps);
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demux->srcpad = gst_pad_new_from_static_template (&src_template, "src");
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gst_pad_set_event_function (demux->srcpad,
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_src_event));
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gst_pad_set_query_function (demux->srcpad,
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GST_DEBUG_FUNCPTR (gst_real_audio_demux_src_query));
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gst_pad_set_active (demux->srcpad, TRUE);
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gst_pad_use_fixed_caps (demux->srcpad);
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stream_id =
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gst_pad_create_stream_id (demux->srcpad, GST_ELEMENT_CAST (demux), NULL);
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event = gst_pad_get_sticky_event (demux->sinkpad, GST_EVENT_STREAM_START, 0);
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if (event) {
|
|
if (gst_event_parse_group_id (event, &demux->group_id))
|
|
demux->have_group_id = TRUE;
|
|
else
|
|
demux->have_group_id = FALSE;
|
|
gst_event_unref (event);
|
|
} else if (!demux->have_group_id) {
|
|
demux->have_group_id = TRUE;
|
|
demux->group_id = gst_util_group_id_next ();
|
|
}
|
|
|
|
event = gst_event_new_stream_start (stream_id);
|
|
if (demux->have_group_id)
|
|
gst_event_set_group_id (event, demux->group_id);
|
|
|
|
gst_pad_push_event (demux->srcpad, event);
|
|
g_free (stream_id);
|
|
|
|
gst_pad_set_caps (demux->srcpad, caps);
|
|
codec_name = gst_pb_utils_get_codec_description (caps);
|
|
gst_caps_unref (caps);
|
|
|
|
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_peer_query_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_empty ();
|
|
gst_tag_list_set_scope (demux->pending_tags, GST_TAG_SCOPE_GLOBAL);
|
|
}
|
|
|
|
gst_tag_list_add (demux->pending_tags, GST_TAG_MERGE_REPLACE,
|
|
GST_TAG_AUDIO_CODEC, codec_name, NULL);
|
|
g_free (codec_name);
|
|
}
|
|
|
|
gst_adapter_unmap (demux->adapter);
|
|
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 '0x%" G_GINT32_MODIFIER "x'", 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;
|
|
GstBuffer *buf;
|
|
|
|
buf = gst_adapter_take_buffer (demux->adapter, unit_size);
|
|
avail -= unit_size;
|
|
|
|
if (demux->need_newsegment) {
|
|
gst_pad_push_event (demux->srcpad,
|
|
gst_event_new_segment (&demux->segment));
|
|
demux->need_newsegment = FALSE;
|
|
}
|
|
|
|
if (demux->pending_tags) {
|
|
gst_pad_push_event (demux->srcpad,
|
|
gst_event_new_tag (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;
|
|
|
|
demux->segment.position = 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_return_val_if_reached (GST_FLOW_ERROR);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_real_audio_demux_chain (GstPad * pad, GstObject * parent, GstBuffer * buf)
|
|
{
|
|
GstRealAudioDemux *demux;
|
|
|
|
demux = GST_REAL_AUDIO_DEMUX (parent);
|
|
|
|
return gst_real_audio_demux_handle_buffer (demux, buf);
|
|
}
|
|
|
|
static void
|
|
gst_real_audio_demux_loop (GstRealAudioDemux * demux)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstBuffer *buf;
|
|
guint bytes_needed;
|
|
|
|
/* 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_return_if_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;
|
|
|
|
buf = NULL;
|
|
ret = gst_pad_pull_range (demux->sinkpad, demux->offset, bytes_needed, &buf);
|
|
|
|
if (ret != GST_FLOW_OK)
|
|
goto pull_range_error;
|
|
|
|
if (gst_buffer_get_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.position != -1 &&
|
|
demux->segment.position > 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 %" G_GSIZE_FORMAT " bytes", bytes_needed,
|
|
gst_buffer_get_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));
|
|
gst_pad_push_event (demux->srcpad,
|
|
gst_event_new_segment_done (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, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRealAudioDemux *demux;
|
|
gboolean ret;
|
|
|
|
demux = GST_REAL_AUDIO_DEMUX (parent);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEGMENT:{
|
|
/* FIXME */
|
|
gst_event_unref (event);
|
|
demux->need_newsegment = TRUE;
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_event_default (pad, parent, event);
|
|
break;
|
|
}
|
|
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);
|
|
|
|
gst_segment_do_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 (TRUE));
|
|
gst_pad_push_event (demux->srcpad, gst_event_new_flush_stop (TRUE));
|
|
|
|
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.position));
|
|
}
|
|
|
|
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, NULL);
|
|
|
|
/* 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, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRealAudioDemux *demux;
|
|
gboolean ret = FALSE;
|
|
|
|
demux = GST_REAL_AUDIO_DEMUX (parent);
|
|
|
|
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, parent, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_real_audio_demux_src_query (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
GstRealAudioDemux *demux;
|
|
gboolean ret = FALSE;
|
|
|
|
demux = GST_REAL_AUDIO_DEMUX (parent);
|
|
|
|
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;
|
|
}
|
|
case GST_QUERY_SEGMENT:
|
|
{
|
|
GstFormat format;
|
|
gint64 start, stop;
|
|
|
|
format = demux->segment.format;
|
|
|
|
start =
|
|
gst_segment_to_stream_time (&demux->segment, format,
|
|
demux->segment.start);
|
|
if ((stop = demux->segment.stop) == -1)
|
|
stop = demux->segment.duration;
|
|
else
|
|
stop = gst_segment_to_stream_time (&demux->segment, format, stop);
|
|
|
|
gst_query_set_segment (query, demux->segment.rate, format, start, stop);
|
|
ret = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
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;
|
|
}
|