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gstreamer/gst/realmedia/rademux.c
Tim-Philipp Müller da9f20eceb Work around deprecated thread API in glib master 
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We can't change most of
these in 0.10 because they're part of our API and ABI.

Suppress deprecation warnings in selected files, mostly for
g_static_rec_mutex_*. StaticRecMutex is part of our API/ABI,
not much we can do here in 0.10.
2011-12-12 11:33:31 +00:00

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/* 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.
*/
/**
* SECTION:element-rademux
*
* Demuxes/parses a RealAudio (.ra) file or stream into compressed audio.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=interview.ra ! rademux ! ffdec_real_288 ! audioconvert ! audioresample ! alsasink
* ]| Read a RealAudio file and decode it and output it to the soundcard using
* the ALSA element. The .ra file is assumed to contain RealAudio version 2.
* |[
* gst-launch gnomevfssrc location=http://www.example.org/interview.ra ! rademux ! a52dec ! audioconvert ! audioresample ! alsasink
* ]| Stream RealAudio data containing AC3 (dnet) compressed audio and decode it
* and output it to the soundcard using the ALSA element.
* </refsect2>
*
* Last reviewed on 2006-10-24 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
/* FIXME 0.11: suppress warnings for deprecated API such as GStaticRecMutex
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include "rademux.h"
#include "rmdemux.h"
#include "rmutils.h"
#include <string.h>
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-pn-realaudio")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS_ANY);
GST_DEBUG_CATEGORY_STATIC (real_audio_demux_debug);
#define GST_CAT_DEFAULT real_audio_demux_debug
GST_BOILERPLATE (GstRealAudioDemux, gst_real_audio_demux, GstElement,
GST_TYPE_ELEMENT);
static GstStateChangeReturn gst_real_audio_demux_change_state (GstElement * e,
GstStateChange transition);
static GstFlowReturn gst_real_audio_demux_chain (GstPad * pad, GstBuffer * buf);
static gboolean gst_real_audio_demux_sink_event (GstPad * pad, GstEvent * ev);
static gboolean gst_real_audio_demux_src_event (GstPad * pad, GstEvent * ev);
static gboolean gst_real_audio_demux_src_query (GstPad * pad, GstQuery * query);
static void gst_real_audio_demux_loop (GstRealAudioDemux * demux);
static gboolean gst_real_audio_demux_sink_activate (GstPad * sinkpad);
static gboolean gst_real_audio_demux_sink_activate_push (GstPad * sinkpad,
gboolean active);
static gboolean gst_real_audio_demux_sink_activate_pull (GstPad * sinkpad,
gboolean active);
static void
gst_real_audio_demux_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_static_pad_template (element_class,
&sink_template);
gst_element_class_add_static_pad_template (element_class, &src_template);
gst_element_class_set_details_simple (element_class, "RealAudio Demuxer",
"Codec/Demuxer",
"Demultiplex a RealAudio file",
"Tim-Philipp Müller <tim centricular net>");
GST_DEBUG_CATEGORY_INIT (real_audio_demux_debug, "rademux",
0, "Demuxer for RealAudio streams");
}
static void
gst_real_audio_demux_finalize (GObject * obj)
{
GstRealAudioDemux *demux = GST_REAL_AUDIO_DEMUX (obj);
g_object_unref (demux->adapter);
G_OBJECT_CLASS (parent_class)->finalize (obj);
}
static void
gst_real_audio_demux_class_init (GstRealAudioDemuxClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
gobject_class->finalize = gst_real_audio_demux_finalize;
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_real_audio_demux_change_state);
}
static void
gst_real_audio_demux_reset (GstRealAudioDemux * demux)
{
gst_adapter_clear (demux->adapter);
if (demux->srcpad) {
GST_DEBUG_OBJECT (demux, "Removing source pad");
gst_element_remove_pad (GST_ELEMENT (demux), demux->srcpad);
demux->srcpad = NULL;
}
if (demux->pending_tags) {
gst_tag_list_free (demux->pending_tags);
demux->pending_tags = NULL;
}
demux->state = REAL_AUDIO_DEMUX_STATE_MARKER;
demux->ra_version = 0;
demux->data_offset = 0;
demux->packet_size = 0;
demux->sample_rate = 0;
demux->sample_width = 0;
demux->channels = 0;
demux->fourcc = 0;
demux->need_newsegment = TRUE;
demux->segment_running = FALSE;
demux->byterate_num = 0;
demux->byterate_denom = 0;
demux->duration = 0;
demux->upstream_size = 0;
demux->offset = 0;
gst_adapter_clear (demux->adapter);
}
static void
gst_real_audio_demux_init (GstRealAudioDemux * demux,
GstRealAudioDemuxClass * klass)
{
demux->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (demux->sinkpad,
GST_DEBUG_FUNCPTR (gst_real_audio_demux_chain));
gst_pad_set_event_function (demux->sinkpad,
GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_event));
gst_pad_set_activate_function (demux->sinkpad,
GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_activate));
gst_pad_set_activatepull_function (demux->sinkpad,
GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_activate_pull));
gst_pad_set_activatepush_function (demux->sinkpad,
GST_DEBUG_FUNCPTR (gst_real_audio_demux_sink_activate_push));
gst_element_add_pad (GST_ELEMENT (demux), demux->sinkpad);
demux->adapter = gst_adapter_new ();
gst_real_audio_demux_reset (demux);
}
static gboolean
gst_real_audio_demux_sink_activate (GstPad * sinkpad)
{
if (gst_pad_check_pull_range (sinkpad)) {
return gst_pad_activate_pull (sinkpad, TRUE);
} else {
return gst_pad_activate_push (sinkpad, TRUE);
}
}
static gboolean
gst_real_audio_demux_sink_activate_push (GstPad * sinkpad, gboolean active)
{
GstRealAudioDemux *demux;
demux = GST_REAL_AUDIO_DEMUX (GST_OBJECT_PARENT (sinkpad));
demux->seekable = FALSE;
return TRUE;
}
static gboolean
gst_real_audio_demux_sink_activate_pull (GstPad * sinkpad, gboolean active)
{
GstRealAudioDemux *demux;
demux = GST_REAL_AUDIO_DEMUX (GST_OBJECT_PARENT (sinkpad));
if (active) {
demux->seekable = TRUE;
return gst_pad_start_task (sinkpad,
(GstTaskFunction) gst_real_audio_demux_loop, demux);
} else {
demux->seekable = FALSE;
return gst_pad_stop_task (sinkpad);
}
}
static GstFlowReturn
gst_real_audio_demux_parse_marker (GstRealAudioDemux * demux)
{
const guint8 *data;
if (gst_adapter_available (demux->adapter) < 6) {
GST_LOG_OBJECT (demux, "need at least 6 bytes, waiting for more data");
return GST_FLOW_OK;
}
data = gst_adapter_peek (demux->adapter, 6);
if (memcmp (data, ".ra\375", 4) != 0)
goto wrong_format;
demux->ra_version = GST_READ_UINT16_BE (data + 4);
GST_DEBUG_OBJECT (demux, "ra_version = %u", demux->ra_version);
if (demux->ra_version != 4 && demux->ra_version != 3)
goto unsupported_ra_version;
gst_adapter_flush (demux->adapter, 6);
demux->state = REAL_AUDIO_DEMUX_STATE_HEADER;
return GST_FLOW_OK;
/* ERRORS */
wrong_format:
{
GST_ELEMENT_ERROR (GST_ELEMENT (demux), STREAM, WRONG_TYPE, (NULL), (NULL));
return GST_FLOW_ERROR;
}
unsupported_ra_version:
{
GST_ELEMENT_ERROR (GST_ELEMENT (demux), STREAM, DECODE,
("Cannot decode this RealAudio file, please file a bug"),
("ra_version = %u", demux->ra_version));
return GST_FLOW_ERROR;
}
}
static GstClockTime
gst_real_demux_get_timestamp_from_offset (GstRealAudioDemux * demux,
guint64 offset)
{
if (offset >= demux->data_offset && demux->byterate_num > 0 &&
demux->byterate_denom > 0) {
return gst_util_uint64_scale (offset - demux->data_offset,
demux->byterate_denom * GST_SECOND, demux->byterate_num);
} else if (offset == demux->data_offset) {
return (GstClockTime) 0;
} else {
return GST_CLOCK_TIME_NONE;
}
}
static gboolean
gst_real_audio_demux_get_data_offset_from_header (GstRealAudioDemux * demux)
{
const guint8 *data;
data = gst_adapter_peek (demux->adapter, 16);
g_assert (data != NULL);
switch (demux->ra_version) {
case 3:
demux->data_offset = GST_READ_UINT16_BE (data) + 8;
break;
case 4:
demux->data_offset = GST_READ_UINT32_BE (data + 12) + 16;
break;
default:
demux->data_offset = 0;
g_return_val_if_reached (FALSE);
}
return TRUE;
}
static GstFlowReturn
gst_real_audio_demux_parse_header (GstRealAudioDemux * demux)
{
const guint8 *data;
gchar *codec_name = NULL;
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);
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);
break;
case GST_RM_AUD_DNET:
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:
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);
codec_name = gst_pb_utils_get_codec_description (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);
g_free (codec_name);
}
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;
buf = gst_buffer_new_and_alloc (unit_size);
gst_buffer_set_caps (buf, GST_PAD_CAPS (demux->srcpad));
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_return_val_if_reached (GST_FLOW_ERROR);
}
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)
{
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;
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;
}
gboolean
gst_rademux_plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rademux",
GST_RANK_SECONDARY, GST_TYPE_REAL_AUDIO_DEMUX);
}
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