mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-13 03:46:34 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/mpegaudioparse/gstmpegaudioparse.c
Sebastian Dröge d1e3a616ca gst/mpegaudioparse/gstmpegaudioparse.*: Save some memory for each frame by only saving the start timestamp and start ... 
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (gst_mp3parse_reset),
(gst_mp3parse_emit_frame), (mp3parse_handle_seek):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Save some memory for each frame by only saving the start timestamp
and start byte position instead of additionally the stop timestamp
and stop byte position. This requires us to use a doubly-linked list
but still saves 8-12 bytes per frame.
2007-07-18 17:51:55 +00:00

1493 lines
46 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Copyright (C) <2006-2007> Jan Schmidt <thaytan@mad.scientist.com>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstmpegaudioparse.h"
GST_DEBUG_CATEGORY_STATIC (mp3parse_debug);
#define GST_CAT_DEFAULT mp3parse_debug
/* elementfactory information */
static GstElementDetails mp3parse_details = {
"MPEG1 Audio Parser",
"Codec/Parser/Audio",
"Parses and frames mpeg1 audio streams (levels 1-3), provides seek",
"Jan Schmidt <thaytan@mad.scientist.com>\n"
"Erik Walthinsen <omega@cse.ogi.edu>"
};
static GstStaticPadTemplate mp3_src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, "
"mpegversion = (int) 1, "
"layer = (int) [ 1, 3 ], "
"rate = (int) [ 8000, 48000 ], channels = (int) [ 1, 2 ],"
"parsed=(boolean) true")
);
static GstStaticPadTemplate mp3_sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, mpegversion = (int) 1, parsed=(boolean)false")
);
/* GstMPEGAudioParse signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
enum
{
ARG_0,
ARG_SKIP,
ARG_BIT_RATE
/* FILL ME */
};
static void gst_mp3parse_class_init (GstMPEGAudioParseClass * klass);
static void gst_mp3parse_base_init (gpointer klass);
static void gst_mp3parse_init (GstMPEGAudioParse * mp3parse,
GstMPEGAudioParseClass * klass);
static gboolean gst_mp3parse_sink_event (GstPad * pad, GstEvent * event);
static GstFlowReturn gst_mp3parse_chain (GstPad * pad, GstBuffer * buffer);
static gboolean mp3parse_src_query (GstPad * pad, GstQuery * query);
static const GstQueryType *mp3parse_get_query_types (GstPad * pad);
static gboolean mp3parse_src_event (GstPad * pad, GstEvent * event);
static int head_check (GstMPEGAudioParse * mp3parse, unsigned long head);
static void gst_mp3parse_dispose (GObject * object);
static void gst_mp3parse_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_mp3parse_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_mp3parse_change_state (GstElement * element,
GstStateChange transition);
static gboolean mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
gint64 bytepos, GstClockTime * ts);
static gboolean
mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total);
/*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
GST_BOILERPLATE (GstMPEGAudioParse, gst_mp3parse, GstElement, GST_TYPE_ELEMENT);
static guint mp3types_bitrates[2][3][16] = {
{
{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
},
{
{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
},
};
static guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
{22050, 24000, 16000},
{11025, 12000, 8000}
};
static inline guint
mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header,
guint * put_version, guint * put_layer, guint * put_channels,
guint * put_bitrate, guint * put_samplerate)
{
guint length;
gulong mode, samplerate, bitrate, layer, channels, padding;
gint lsf, mpg25;
if (header & (1 << 20)) {
lsf = (header & (1 << 19)) ? 0 : 1;
mpg25 = 0;
} else {
lsf = 1;
mpg25 = 1;
}
layer = 4 - ((header >> 17) & 0x3);
bitrate = (header >> 12) & 0xF;
bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
if (bitrate == 0)
return 0;
samplerate = (header >> 10) & 0x3;
samplerate = mp3types_freqs[lsf + mpg25][samplerate];
padding = (header >> 9) & 0x1;
mode = (header >> 6) & 0x3;
channels = (mode == 3) ? 1 : 2;
switch (layer) {
case 1:
length = 4 * ((bitrate * 12) / samplerate + padding);
break;
case 2:
length = (bitrate * 144) / samplerate + padding;
break;
default:
case 3:
length = (bitrate * 144) / (samplerate << lsf) + padding;
break;
}
GST_DEBUG_OBJECT (mp3parse, "Calculated mp3 frame length of %u bytes",
length);
GST_DEBUG_OBJECT (mp3parse, "samplerate = %lu, bitrate = %lu, layer = %lu, "
"channels = %lu", samplerate, bitrate, layer, channels);
if (put_version)
*put_version = lsf ? 2 : 1;
if (put_layer)
*put_layer = layer;
if (put_channels)
*put_channels = channels;
if (put_bitrate)
*put_bitrate = bitrate;
if (put_samplerate)
*put_samplerate = samplerate;
return length;
}
static GstCaps *
mp3_caps_create (guint layer, guint channels, guint bitrate, guint samplerate)
{
GstCaps *new;
g_assert (layer);
g_assert (samplerate);
g_assert (bitrate);
g_assert (channels);
new = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, 1,
"layer", G_TYPE_INT, layer,
"rate", G_TYPE_INT, samplerate,
"channels", G_TYPE_INT, channels, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
return new;
}
static void
gst_mp3parse_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&mp3_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&mp3_src_template));
gst_element_class_set_details (element_class, &mp3parse_details);
}
static void
gst_mp3parse_class_init (GstMPEGAudioParseClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_mp3parse_set_property;
gobject_class->get_property = gst_mp3parse_get_property;
gobject_class->dispose = gst_mp3parse_dispose;
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_SKIP,
g_param_spec_int ("skip", "skip", "skip",
G_MININT, G_MAXINT, 0, G_PARAM_READWRITE));
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BIT_RATE,
g_param_spec_int ("bitrate", "Bitrate", "Bit Rate",
G_MININT, G_MAXINT, 0, G_PARAM_READABLE));
gstelement_class->change_state = gst_mp3parse_change_state;
}
static void
gst_mp3parse_reset (GstMPEGAudioParse * mp3parse)
{
mp3parse->skip = 0;
mp3parse->resyncing = TRUE;
mp3parse->next_ts = GST_CLOCK_TIME_NONE;
mp3parse->cur_offset = -1;
mp3parse->tracked_offset = 0;
mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
mp3parse->pending_offset = -1;
gst_adapter_clear (mp3parse->adapter);
mp3parse->rate = mp3parse->channels = mp3parse->layer = -1;
mp3parse->version = 1;
mp3parse->max_bitreservoir = GST_CLOCK_TIME_NONE;
mp3parse->avg_bitrate = 0;
mp3parse->bitrate_sum = 0;
mp3parse->last_posted_bitrate = 0;
mp3parse->frame_count = 0;
mp3parse->sent_codec_tag = FALSE;
mp3parse->xing_flags = 0;
mp3parse->xing_bitrate = 0;
if (mp3parse->seek_table) {
g_list_foreach (mp3parse->seek_table, (GFunc) g_free, NULL);
mp3parse->seek_table = NULL;
}
g_mutex_lock (mp3parse->pending_accurate_seeks_lock);
if (mp3parse->pending_accurate_seeks) {
g_slist_foreach (mp3parse->pending_accurate_seeks, (GFunc) g_free, NULL);
mp3parse->pending_accurate_seeks = NULL;
}
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
mp3parse->exact_position = FALSE;
gst_segment_init (&mp3parse->segment, GST_FORMAT_TIME);
}
static void
gst_mp3parse_init (GstMPEGAudioParse * mp3parse, GstMPEGAudioParseClass * klass)
{
mp3parse->sinkpad =
gst_pad_new_from_static_template (&mp3_sink_template, "sink");
gst_pad_set_event_function (mp3parse->sinkpad, gst_mp3parse_sink_event);
gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
mp3parse->srcpad =
gst_pad_new_from_static_template (&mp3_src_template, "src");
gst_pad_use_fixed_caps (mp3parse->srcpad);
gst_pad_set_event_function (mp3parse->srcpad, mp3parse_src_event);
gst_pad_set_query_function (mp3parse->srcpad, mp3parse_src_query);
gst_pad_set_query_type_function (mp3parse->srcpad, mp3parse_get_query_types);
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
mp3parse->adapter = gst_adapter_new ();
mp3parse->pending_accurate_seeks_lock = g_mutex_new ();
gst_mp3parse_reset (mp3parse);
}
static void
gst_mp3parse_dispose (GObject * object)
{
GstMPEGAudioParse *mp3parse = GST_MP3PARSE (object);
if (mp3parse->adapter) {
g_object_unref (mp3parse->adapter);
mp3parse->adapter = NULL;
}
g_mutex_free (mp3parse->pending_accurate_seeks_lock);
mp3parse->pending_accurate_seeks_lock = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static gboolean
gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
{
gboolean res;
GstMPEGAudioParse *mp3parse;
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
gdouble rate, applied_rate;
GstFormat format;
gint64 start, stop, pos;
gboolean update;
gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
&format, &start, &stop, &pos);
g_mutex_lock (mp3parse->pending_accurate_seeks_lock);
if (format == GST_FORMAT_BYTES && mp3parse->pending_accurate_seeks) {
MPEGAudioPendingAccurateSeek *seek = NULL;
GSList *node;
for (node = mp3parse->pending_accurate_seeks; node; node = node->next) {
MPEGAudioPendingAccurateSeek *tmp = node->data;
if (tmp->upstream_start == pos) {
seek = tmp;
break;
}
}
if (seek) {
GstSegment *s = &seek->segment;
event =
gst_event_new_new_segment_full (FALSE, s->rate, s->applied_rate,
GST_FORMAT_TIME, s->start, s->stop, s->last_stop);
mp3parse->segment = seek->segment;
mp3parse->resyncing = FALSE;
mp3parse->cur_offset = pos;
mp3parse->next_ts = seek->timestamp_start;
mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
mp3parse->tracked_offset = 0;
gst_event_parse_new_segment_full (event, &update, &rate,
&applied_rate, &format, &start, &stop, &pos);
GST_DEBUG_OBJECT (mp3parse,
"Pushing accurate newseg rate %g, applied rate %g, "
"format %d, start %lld, stop %lld, pos %lld\n", rate,
applied_rate, format, start, stop, pos);
g_free (seek);
mp3parse->pending_accurate_seeks =
g_slist_delete_link (mp3parse->pending_accurate_seeks, node);
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
if (s->flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (mp3parse),
gst_message_new_segment_start (GST_OBJECT_CAST (mp3parse),
s->format, s->last_stop));
}
res = gst_pad_push_event (mp3parse->srcpad, event);
return res;
} else {
GST_WARNING_OBJECT (mp3parse,
"Accurate seek not possible, didn't get an appropiate upstream segment");
}
}
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
mp3parse->exact_position = FALSE;
if (format == GST_FORMAT_BYTES) {
GstClockTime seg_start, seg_stop, seg_pos;
/* stop time is allowed to be open-ended, but not start & pos */
if (!mp3parse_bytepos_to_time (mp3parse, stop, &seg_stop))
seg_stop = GST_CLOCK_TIME_NONE;
if (mp3parse_bytepos_to_time (mp3parse, start, &seg_start) &&
mp3parse_bytepos_to_time (mp3parse, pos, &seg_pos)) {
gst_event_unref (event);
event = gst_event_new_new_segment_full (update, rate, applied_rate,
GST_FORMAT_TIME, seg_start, seg_stop, seg_pos);
format = GST_FORMAT_TIME;
GST_DEBUG_OBJECT (mp3parse, "Converted incoming segment to TIME. "
"start = %" GST_TIME_FORMAT ", stop = %" GST_TIME_FORMAT
", pos = %" GST_TIME_FORMAT, GST_TIME_ARGS (seg_start),
GST_TIME_ARGS (seg_stop), GST_TIME_ARGS (seg_pos));
}
}
if (format != GST_FORMAT_TIME) {
/* Unknown incoming segment format. Output a default open-ended
* TIME segment */
gst_event_unref (event);
event = gst_event_new_new_segment_full (update, rate, applied_rate,
GST_FORMAT_TIME, 0, GST_CLOCK_TIME_NONE, 0);
}
mp3parse->resyncing = TRUE;
mp3parse->cur_offset = -1;
mp3parse->next_ts = GST_CLOCK_TIME_NONE;
mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
mp3parse->tracked_offset = 0;
gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
&format, &start, &stop, &pos);
GST_DEBUG_OBJECT (mp3parse, "Pushing newseg rate %g, applied rate %g, "
"format %d, start %lld, stop %lld, pos %lld\n",
rate, applied_rate, format, start, stop, pos);
gst_segment_set_newsegment_full (&mp3parse->segment, update, rate,
applied_rate, format, start, stop, pos);
res = gst_pad_push_event (mp3parse->srcpad, event);
break;
}
case GST_EVENT_FLUSH_STOP:
/* Clear our adapter and set up for a new position */
gst_adapter_clear (mp3parse->adapter);
res = gst_pad_push_event (mp3parse->srcpad, event);
break;
default:
res = gst_pad_push_event (mp3parse->srcpad, event);
break;
}
gst_object_unref (mp3parse);
return res;
}
static MPEGAudioSeekEntry *
mp3parse_seek_table_last_entry (GstMPEGAudioParse * mp3parse)
{
MPEGAudioSeekEntry *ret = NULL;
if (mp3parse->seek_table) {
ret = mp3parse->seek_table->data;
}
return ret;
}
/* Prepare a buffer of the indicated size, timestamp it and output */
static GstFlowReturn
gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size)
{
GstBuffer *outbuf;
guint bitrate;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime push_start;
outbuf = gst_adapter_take_buffer (mp3parse->adapter, size);
GST_BUFFER_DURATION (outbuf) =
gst_util_uint64_scale (GST_SECOND, mp3parse->spf, mp3parse->rate);
GST_BUFFER_OFFSET (outbuf) = mp3parse->cur_offset;
/* Check if we have a pending timestamp from an incoming buffer to apply
* here */
if (GST_CLOCK_TIME_IS_VALID (mp3parse->pending_ts)) {
if (mp3parse->tracked_offset >= mp3parse->pending_offset) {
/* If the incoming timestamp differs from our expected by more than 2
* 90khz MPEG ticks, then take it and, if needed, set the discont flag.
* This avoids creating imperfect streams just because of
* quantization in the MPEG clock sampling */
GstClockTimeDiff diff = mp3parse->next_ts - mp3parse->pending_ts;
if (diff < -2 * (GST_SECOND / 90000) || diff > 2 * (GST_SECOND / 90000)) {
GST_DEBUG_OBJECT (mp3parse, "Updating next_ts from %" GST_TIME_FORMAT
" to pending ts %" GST_TIME_FORMAT
" at offset %lld (pending offset was %lld)",
GST_TIME_ARGS (mp3parse->next_ts),
GST_TIME_ARGS (mp3parse->pending_ts), mp3parse->tracked_offset,
mp3parse->pending_offset);
/* Only set discont if we sent out some timestamps already and we're
* adjusting */
if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts))
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
mp3parse->next_ts = mp3parse->pending_ts;
}
mp3parse->pending_ts = GST_CLOCK_TIME_NONE;
}
}
/* Decide what timestamp we're going to apply */
if (GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts)) {
GST_BUFFER_TIMESTAMP (outbuf) = mp3parse->next_ts;
} else {
GstClockTime ts;
/* No timestamp yet, convert our offset to a timestamp if we can, or
* start at 0 */
if (mp3parse_bytepos_to_time (mp3parse, mp3parse->cur_offset, &ts))
GST_BUFFER_TIMESTAMP (outbuf) = ts;
else {
GST_BUFFER_TIMESTAMP (outbuf) = 0;
}
}
if (GST_BUFFER_TIMESTAMP (outbuf) == 0)
mp3parse->exact_position = TRUE;
if (mp3parse->exact_position && (!mp3parse->seek_table ||
(mp3parse_seek_table_last_entry (mp3parse))->byte <
GST_BUFFER_OFFSET (outbuf))) {
MPEGAudioSeekEntry *entry = g_new0 (MPEGAudioSeekEntry, 1);
entry->byte = mp3parse->cur_offset;
entry->timestamp = GST_BUFFER_TIMESTAMP (outbuf);
mp3parse->seek_table = g_list_prepend (mp3parse->seek_table, entry);
GST_DEBUG_OBJECT (mp3parse, "Adding index entry %" GST_TIME_FORMAT
" @ offset 0x%08" G_GINT64_MODIFIER "x",
GST_TIME_ARGS (entry->timestamp), entry->byte);
}
/* Update our byte offset tracking */
if (mp3parse->cur_offset != -1) {
mp3parse->cur_offset += size;
}
mp3parse->tracked_offset += size;
mp3parse->next_ts =
GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (mp3parse->srcpad));
/* Post a bitrate tag if we need to before pushing the buffer */
if (mp3parse->xing_bitrate != 0)
bitrate = mp3parse->xing_bitrate;
else
bitrate = mp3parse->avg_bitrate;
if ((mp3parse->last_posted_bitrate / 10000) != (bitrate / 10000)) {
GstTagList *taglist = gst_tag_list_new ();
mp3parse->last_posted_bitrate = bitrate;
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_BITRATE,
mp3parse->last_posted_bitrate, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
mp3parse->srcpad, taglist);
}
/* We start pushing 9 frames earlier (29 frames for MPEG2) than
* segment start to be able to decode the first frame we want.
* 9 (29) frames are the theoretical maximum of frames that contain
* data for the current frame (bit reservoir).
*/
if (mp3parse->segment.start == 0) {
push_start = 0;
} else if (GST_CLOCK_TIME_IS_VALID (mp3parse->max_bitreservoir)) {
if (mp3parse->segment.start > mp3parse->max_bitreservoir)
push_start = mp3parse->segment.start - mp3parse->max_bitreservoir;
else
push_start = 0;
} else {
push_start = mp3parse->segment.start;
}
if (G_UNLIKELY ((GST_CLOCK_TIME_IS_VALID (push_start) &&
GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf)
< push_start)
|| (GST_CLOCK_TIME_IS_VALID (mp3parse->segment.stop)
&& GST_BUFFER_TIMESTAMP (outbuf) >= mp3parse->segment.stop))) {
GST_DEBUG_OBJECT (mp3parse,
"Buffer outside of configured segment range %" GST_TIME_FORMAT
" to %" GST_TIME_FORMAT ", dropping, timestamp %"
GST_TIME_FORMAT ", offset 0x%08" G_GINT64_MODIFIER "x",
GST_TIME_ARGS (push_start), GST_TIME_ARGS (mp3parse->segment.stop),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_BUFFER_OFFSET (outbuf));
gst_buffer_unref (outbuf);
ret = GST_FLOW_OK;
} else {
GST_DEBUG_OBJECT (mp3parse,
"pushing buffer of %d bytes, timestamp %" GST_TIME_FORMAT
", offset 0x%08" G_GINT64_MODIFIER "x", size,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_BUFFER_OFFSET (outbuf));
mp3parse->segment.last_stop = GST_BUFFER_TIMESTAMP (outbuf);
ret = gst_pad_push (mp3parse->srcpad, outbuf);
}
if (ret == GST_FLOW_UNEXPECTED
&& mp3parse->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
GST_LOG_OBJECT (mp3parse, "Sending segment done");
if ((stop = mp3parse->segment.stop) == -1)
stop = mp3parse->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (mp3parse),
gst_message_new_segment_done (GST_OBJECT_CAST (mp3parse),
mp3parse->segment.format, stop));
} else if (GST_CLOCK_TIME_IS_VALID (mp3parse->segment.stop)
&& mp3parse->next_ts >= mp3parse->segment.stop) {
GST_DEBUG_OBJECT (mp3parse, "Sending EOS");
gst_pad_push_event (mp3parse->srcpad, gst_event_new_eos ());
}
return ret;
}
#define XING_FRAMES_FLAG 0x0001
#define XING_BYTES_FLAG 0x0002
#define XING_TOC_FLAG 0x0004
#define XING_VBR_SCALE_FLAG 0x0008
static void
gst_mp3parse_handle_first_frame (GstMPEGAudioParse * mp3parse)
{
GstTagList *taglist;
gchar *codec;
const guint32 xing_id = 0x58696e67; /* 'Xing' in hex */
const guint32 info_id = 0x496e666f; /* 'Info' in hex - found in LAME CBR files */
const guint XING_HDR_MIN = 8;
gint xing_offset;
guint64 avail;
guint32 read_id;
const guint8 *data;
/* Output codec tag */
if (!mp3parse->sent_codec_tag) {
if (mp3parse->layer == 3) {
codec = g_strdup_printf ("MPEG %d Audio, Layer %d (MP3)",
mp3parse->version, mp3parse->layer);
} else {
codec = g_strdup_printf ("MPEG %d Audio, Layer %d",
mp3parse->version, mp3parse->layer);
}
taglist = gst_tag_list_new ();
gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_AUDIO_CODEC, codec, NULL);
gst_element_found_tags_for_pad (GST_ELEMENT (mp3parse),
mp3parse->srcpad, taglist);
g_free (codec);
mp3parse->sent_codec_tag = TRUE;
}
/* end setting the tag */
/* Check first frame for Xing info */
if (mp3parse->version == 1) { /* MPEG-1 file */
if (mp3parse->channels == 1)
xing_offset = 0x11;
else
xing_offset = 0x20;
} else { /* MPEG-2 header */
if (mp3parse->channels == 1)
xing_offset = 0x09;
else
xing_offset = 0x11;
}
/* Skip the 4 bytes of the MP3 header too */
xing_offset += 4;
/* Check if we have enough data to read the Xing header */
avail = gst_adapter_available (mp3parse->adapter);
if (avail < xing_offset + XING_HDR_MIN)
return;
data = gst_adapter_peek (mp3parse->adapter, xing_offset + XING_HDR_MIN);
if (data == NULL)
return;
/* The header starts at the provided offset */
data += xing_offset;
read_id = GST_READ_UINT32_BE (data);
if (read_id == xing_id || read_id == info_id) {
guint32 xing_flags;
guint bytes_needed = xing_offset + XING_HDR_MIN;
GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
/* Read 4 base bytes of flags, big-endian */
xing_flags = GST_READ_UINT32_BE (data + 4);
if (xing_flags & XING_FRAMES_FLAG)
bytes_needed += 4;
if (xing_flags & XING_BYTES_FLAG)
bytes_needed += 4;
if (xing_flags & XING_TOC_FLAG)
bytes_needed += 100;
if (xing_flags & XING_VBR_SCALE_FLAG)
bytes_needed += 4;
if (avail < bytes_needed) {
GST_DEBUG_OBJECT (mp3parse,
"Not enough data to read Xing header (need %d)", bytes_needed);
return;
}
GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
mp3parse->xing_flags = xing_flags;
data = gst_adapter_peek (mp3parse->adapter, bytes_needed);
data += xing_offset + XING_HDR_MIN;
if (xing_flags & XING_FRAMES_FLAG) {
gint64 total_bytes;
mp3parse->xing_frames = GST_READ_UINT32_BE (data);
mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
(guint64) (mp3parse->xing_frames) * (mp3parse->spf), mp3parse->rate);
/* We know the total time. If we also know the upstream size, compute the
* total bitrate, rounded up to the nearest kbit/sec */
if (mp3parse_total_bytes (mp3parse, &total_bytes)) {
mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes,
8 * GST_SECOND, mp3parse->xing_total_time);
mp3parse->xing_bitrate += 500;
mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000;
}
data += 4;
} else {
mp3parse->xing_frames = 0;
mp3parse->xing_total_time = 0;
}
if (xing_flags & XING_BYTES_FLAG) {
mp3parse->xing_bytes = GST_READ_UINT32_BE (data);
data += 4;
} else
mp3parse->xing_bytes = 0;
if (xing_flags & XING_TOC_FLAG) {
int i, percent = 0;
guchar *table = mp3parse->xing_seek_table;
/* xing seek table: percent time -> 1/256 bytepos */
memcpy (mp3parse->xing_seek_table, data, 100);
/* build inverse table: 1/256 bytepos -> 1/100 percent time */
for (i = 0; i < 256; i++) {
while (percent < 99 && table[percent + 1] <= i)
percent++;
if (table[percent] == i) {
mp3parse->xing_seek_table_inverse[i] = percent * 100;
} else if (table[percent] < i && percent < 99) {
gdouble fa, fb, fx;
gint a = percent, b = percent + 1;
fa = table[a];
fb = table[b];
fx = (b - a) / (fb - fa) * (i - fa) + a;
mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
} else if (percent == 98 && table[percent + 1] <= i) {
gdouble fa, fb, fx;
gint a = percent + 1, b = 100;
fa = table[a];
fb = 256.0;
fx = (b - a) / (fb - fa) * (i - fa) + a;
mp3parse->xing_seek_table_inverse[i] = (guint16) (fx * 100);
}
}
data += 100;
} else {
memset (mp3parse->xing_seek_table, 0, 100);
memset (mp3parse->xing_seek_table_inverse, 0, 256);
}
if (xing_flags & XING_VBR_SCALE_FLAG) {
mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data);
data += 4;
} else
mp3parse->xing_vbr_scale = 0;
GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %"
G_GUINT64_FORMAT ", vbr scale %u", mp3parse->xing_frames,
mp3parse->xing_total_time, mp3parse->xing_vbr_scale);
} else {
GST_DEBUG_OBJECT (mp3parse, "Xing header not found in first frame");
}
}
static GstFlowReturn
gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
{
GstFlowReturn flow = GST_FLOW_OK;
GstMPEGAudioParse *mp3parse;
const guchar *data;
guint32 header;
int bpf;
guint available;
GstClockTime timestamp;
mp3parse = GST_MP3PARSE (GST_PAD_PARENT (pad));
GST_LOG_OBJECT (mp3parse, "buffer of %d bytes", GST_BUFFER_SIZE (buf));
timestamp = GST_BUFFER_TIMESTAMP (buf);
/* If we don't yet have a next timestamp, save it and the incoming offset
* so we can apply it to the right outgoing buffer */
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
gint64 avail = gst_adapter_available (mp3parse->adapter);
mp3parse->pending_ts = timestamp;
mp3parse->pending_offset = mp3parse->tracked_offset + avail;
GST_LOG_OBJECT (mp3parse, "Have pending ts %" GST_TIME_FORMAT
" to apply in %lld bytes (@ off %lld)\n",
GST_TIME_ARGS (mp3parse->pending_ts), avail, mp3parse->pending_offset);
}
/* Update the cur_offset we'll apply to outgoing buffers */
if (mp3parse->cur_offset == -1 && GST_BUFFER_OFFSET (buf) != -1)
mp3parse->cur_offset = GST_BUFFER_OFFSET (buf);
/* And add the data to the pool */
gst_adapter_push (mp3parse->adapter, buf);
/* while we still have at least 4 bytes (for the header) available */
while (gst_adapter_available (mp3parse->adapter) >= 4) {
/* search for a possible start byte */
data = gst_adapter_peek (mp3parse->adapter, 4);
if (*data != 0xff) {
/* It'd be nice to make this efficient, but it's ok for now; this is only
* when resyncing */
mp3parse->resyncing = TRUE;
gst_adapter_flush (mp3parse->adapter, 1);
if (mp3parse->cur_offset != -1)
mp3parse->cur_offset++;
mp3parse->tracked_offset++;
continue;
}
available = gst_adapter_available (mp3parse->adapter);
/* construct the header word */
header = GST_READ_UINT32_BE (data);
/* if it's a valid header, go ahead and send off the frame */
if (head_check (mp3parse, header)) {
guint bitrate = 0, layer = 0, rate = 0, channels = 0, version = 0;
if (!(bpf = mp3_type_frame_length_from_header (mp3parse, header,
&version, &layer, &channels, &bitrate, &rate)))
goto header_error;
/*************************************************************************
* robust seek support
* - This performs additional frame validation if the resyncing flag is set
* (indicating a discontinuous stream).
* - The current frame header is not accepted as valid unless the NEXT
* frame header has the same values for most fields. This significantly
* increases the probability that we aren't processing random data.
* - It is not clear if this is sufficient for robust seeking of Layer III
* streams which utilize the concept of a "bit reservoir" by borrowing
* bitrate from previous frames. In this case, seeking may be more
* complicated because the frames are not independently coded.
*************************************************************************/
if (mp3parse->resyncing) {
guint32 header2;
const guint8 *data2;
/* wait until we have the the entire current frame as well as the next
* frame header */
if (available < bpf + 4)
break;
data2 = gst_adapter_peek (mp3parse->adapter, bpf + 4);
header2 = GST_READ_UINT32_BE (data2 + bpf);
GST_DEBUG_OBJECT (mp3parse, "header=%08X, header2=%08X, bpf=%d",
(unsigned int) header, (unsigned int) header2, bpf);
/* mask the bits which are allowed to differ between frames */
#define HDRMASK ~((0xF << 12) /* bitrate */ | \
(0x1 << 9) /* padding */ | \
(0x3 << 4)) /* mode extension */
/* require 2 matching headers in a row */
if ((header2 & HDRMASK) != (header & HDRMASK)) {
GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
"(header=%08X, header2=%08X, bpf=%d)",
(unsigned int) header, (unsigned int) header2, bpf);
/* This frame is invalid. Start looking for a valid frame at the
* next position in the stream */
mp3parse->resyncing = TRUE;
gst_adapter_flush (mp3parse->adapter, 1);
if (mp3parse->cur_offset != -1)
mp3parse->cur_offset++;
mp3parse->tracked_offset++;
continue;
}
}
/* if we don't have the whole frame... */
if (available < bpf) {
GST_DEBUG_OBJECT (mp3parse, "insufficient data available, need "
"%d bytes, have %d", bpf, available);
break;
}
if (channels != mp3parse->channels ||
rate != mp3parse->rate ||
layer != mp3parse->layer || bitrate != mp3parse->bit_rate) {
GstCaps *caps;
caps = mp3_caps_create (layer, channels, bitrate, rate);
gst_pad_set_caps (mp3parse->srcpad, caps);
gst_caps_unref (caps);
mp3parse->channels = channels;
mp3parse->layer = layer;
mp3parse->rate = rate;
mp3parse->bit_rate = bitrate;
mp3parse->version = version;
/* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
if (mp3parse->layer == 1)
mp3parse->spf = 384;
else if (mp3parse->layer == 2)
mp3parse->spf = 1152;
else if (mp3parse->version == 2) {
mp3parse->spf = 576;
} else
mp3parse->spf = 1152;
}
mp3parse->max_bitreservoir = gst_util_uint64_scale (GST_SECOND,
((version == 1) ? 10 : 30) * mp3parse->spf, mp3parse->rate);
/* Check the first frame for a Xing header to get our total length */
if (mp3parse->frame_count == 0) {
/* For the first frame in the file, look for a Xing frame after
* the header, and output a codec tag */
gst_mp3parse_handle_first_frame (mp3parse);
}
/* Update VBR stats */
mp3parse->bitrate_sum += mp3parse->bit_rate;
mp3parse->frame_count++;
/* Compute the average bitrate, rounded up to the nearest 1000 bits */
mp3parse->avg_bitrate =
(mp3parse->bitrate_sum / mp3parse->frame_count + 500);
mp3parse->avg_bitrate -= mp3parse->avg_bitrate % 1000;
if (!mp3parse->skip) {
mp3parse->resyncing = FALSE;
flow = gst_mp3parse_emit_frame (mp3parse, bpf);
} else {
GST_DEBUG_OBJECT (mp3parse, "skipping buffer of %d bytes", bpf);
gst_adapter_flush (mp3parse->adapter, bpf);
if (mp3parse->cur_offset != -1)
mp3parse->cur_offset += bpf;
mp3parse->tracked_offset += bpf;
mp3parse->skip--;
}
} else {
mp3parse->resyncing = TRUE;
gst_adapter_flush (mp3parse->adapter, 1);
if (mp3parse->cur_offset != -1)
mp3parse->cur_offset++;
mp3parse->tracked_offset++;
GST_DEBUG_OBJECT (mp3parse, "wrong header, skipping byte");
}
if (GST_FLOW_IS_FATAL (flow))
break;
}
return flow;
header_error:
GST_ELEMENT_ERROR (mp3parse, STREAM, DECODE,
("Invalid MP3 header found"), (NULL));
return GST_FLOW_ERROR;
}
static gboolean
head_check (GstMPEGAudioParse * mp3parse, unsigned long head)
{
GST_DEBUG_OBJECT (mp3parse, "checking mp3 header 0x%08lx", head);
/* if it's not a valid sync */
if ((head & 0xffe00000) != 0xffe00000) {
GST_DEBUG_OBJECT (mp3parse, "invalid sync");
return FALSE;
}
/* if it's an invalid MPEG version */
if (((head >> 19) & 3) == 0x1) {
GST_DEBUG_OBJECT (mp3parse, "invalid MPEG version");
return FALSE;
}
/* if it's an invalid layer */
if (!((head >> 17) & 3)) {
GST_DEBUG_OBJECT (mp3parse, "invalid layer");
return FALSE;
}
/* if it's an invalid bitrate */
if (((head >> 12) & 0xf) == 0x0) {
GST_DEBUG_OBJECT (mp3parse, "invalid bitrate");
return FALSE;
}
if (((head >> 12) & 0xf) == 0xf) {
GST_DEBUG_OBJECT (mp3parse, "invalid bitrate");
return FALSE;
}
/* if it's an invalid samplerate */
if (((head >> 10) & 0x3) == 0x3) {
GST_DEBUG_OBJECT (mp3parse, "invalid samplerate");
return FALSE;
}
if ((head & 0xffff0000) == 0xfffe0000) {
GST_DEBUG_OBJECT (mp3parse, "invalid sync");
return FALSE;
}
if (head & 0x00000002) {
GST_DEBUG_OBJECT (mp3parse, "invalid emphasis");
return FALSE;
}
return TRUE;
}
static void
gst_mp3parse_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstMPEGAudioParse *src;
g_return_if_fail (GST_IS_MP3PARSE (object));
src = GST_MP3PARSE (object);
switch (prop_id) {
case ARG_SKIP:
src->skip = g_value_get_int (value);
break;
default:
break;
}
}
static void
gst_mp3parse_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstMPEGAudioParse *src;
g_return_if_fail (GST_IS_MP3PARSE (object));
src = GST_MP3PARSE (object);
switch (prop_id) {
case ARG_SKIP:
g_value_set_int (value, src->skip);
break;
case ARG_BIT_RATE:
g_value_set_int (value, src->bit_rate * 1000);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStateChangeReturn
gst_mp3parse_change_state (GstElement * element, GstStateChange transition)
{
GstMPEGAudioParse *mp3parse;
GstStateChangeReturn result;
mp3parse = GST_MP3PARSE (element);
result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_mp3parse_reset (mp3parse);
break;
default:
break;
}
return result;
}
static gboolean
mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total)
{
GstFormat fmt = GST_FORMAT_BYTES;
if (gst_pad_query_peer_duration (mp3parse->sinkpad, &fmt, total))
return TRUE;
if (mp3parse->xing_flags & XING_BYTES_FLAG) {
*total = mp3parse->xing_bytes;
return TRUE;
}
return FALSE;
}
static gboolean
mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total)
{
gint64 total_bytes;
*total = GST_CLOCK_TIME_NONE;
if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
*total = mp3parse->xing_total_time;
return TRUE;
}
/* Calculate time from the measured bitrate */
if (!mp3parse_total_bytes (mp3parse, &total_bytes))
return FALSE;
if (total_bytes != -1
&& !mp3parse_bytepos_to_time (mp3parse, total_bytes, total))
return FALSE;
return TRUE;
}
/* Convert a timestamp to the file position required to start decoding that
* timestamp. For now, this just uses the avg bitrate. Later, use an
* incrementally accumulated seek table */
static gboolean
mp3parse_time_to_bytepos (GstMPEGAudioParse * mp3parse, GstClockTime ts,
gint64 * bytepos)
{
gint64 total_bytes;
GstClockTime total_time;
/* -1 always maps to -1 */
if (ts == -1) {
*bytepos = -1;
return TRUE;
}
/* If XING seek table exists use this for time->byte conversion */
if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
mp3parse_total_bytes (mp3parse, &total_bytes) &&
mp3parse_total_time (mp3parse, &total_time)) {
gdouble fa, fb, fx;
gdouble percent = CLAMP ((100.0 * ts) / total_time, 0.0, 100.0);
gint index = CLAMP (percent, 0, 99);
fa = mp3parse->xing_seek_table[index];
if (index < 99)
fb = mp3parse->xing_seek_table[index + 1];
else
fb = 256.0;
fx = fa + (fb - fa) * (percent - index);
*bytepos = (1.0 / 256.0) * fx * total_bytes;
return TRUE;
}
if (mp3parse->avg_bitrate == 0)
goto no_bitrate;
*bytepos =
gst_util_uint64_scale (ts, mp3parse->avg_bitrate, (8 * GST_SECOND));
return TRUE;
no_bitrate:
GST_DEBUG_OBJECT (mp3parse, "Cannot seek yet - no average bitrate");
return FALSE;
}
static gboolean
mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
gint64 bytepos, GstClockTime * ts)
{
gint64 total_bytes;
GstClockTime total_time;
if (bytepos == -1) {
*ts = GST_CLOCK_TIME_NONE;
return TRUE;
}
if (bytepos == 0) {
*ts = 0;
return TRUE;
}
/* If XING seek table exists use this for byte->time conversion */
if ((mp3parse->xing_flags & XING_TOC_FLAG) &&
mp3parse_total_bytes (mp3parse, &total_bytes) &&
mp3parse_total_time (mp3parse, &total_time)) {
gdouble fa, fb, fx;
gdouble pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
gint index = CLAMP (pos, 0, 255);
fa = mp3parse->xing_seek_table_inverse[index];
if (index < 255)
fb = mp3parse->xing_seek_table_inverse[index + 1];
else
fb = 10000.0;
fx = fa + (fb - fa) * (pos - index);
*ts = (1.0 / 10000.0) * fx * total_time;
return TRUE;
}
/* Cannot convert anything except 0 if we don't have a bitrate yet */
if (mp3parse->avg_bitrate == 0)
return FALSE;
*ts = (GstClockTime) gst_util_uint64_scale (GST_SECOND, bytepos * 8,
mp3parse->avg_bitrate);
return TRUE;
}
static gboolean
mp3parse_handle_seek (GstMPEGAudioParse * mp3parse, GstEvent * event)
{
GstFormat format;
gdouble rate;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gint64 byte_cur, byte_stop;
gst_event_parse_seek (event, &rate, &format, &flags, &cur_type, &cur,
&stop_type, &stop);
GST_DEBUG_OBJECT (mp3parse, "Performing seek to %" GST_TIME_FORMAT,
GST_TIME_ARGS (cur));
/* For any format other than TIME, see if upstream handles
* it directly or fail. For TIME, try upstream, but do it ourselves if
* it fails upstream */
if (format != GST_FORMAT_TIME) {
gst_event_ref (event);
return gst_pad_push_event (mp3parse->sinkpad, event);
} else {
gst_event_ref (event);
if (gst_pad_push_event (mp3parse->sinkpad, event))
return TRUE;
}
/* Handle TIME based seeks by converting to a BYTE position */
/* For accurate seeking get the frame 9 (MPEG1) or 29 (MPEG2) frames
* before the one we want to seek to and push them all to the decoder.
*
* This is necessary because of the bit reservoir. See
* http://www.mars.org/mailman/public/mad-dev/2002-May/000634.html
*
*/
if (flags & GST_SEEK_FLAG_ACCURATE) {
MPEGAudioPendingAccurateSeek *seek =
g_new0 (MPEGAudioPendingAccurateSeek, 1);
GstClockTime start;
seek->segment = mp3parse->segment;
gst_segment_set_seek (&seek->segment, rate, GST_FORMAT_TIME,
flags, cur_type, cur, stop_type, stop, NULL);
if (!mp3parse->seek_table) {
byte_cur = 0;
byte_stop = -1;
start = 0;
} else {
MPEGAudioSeekEntry *entry = NULL, *start_entry = NULL, *stop_entry = NULL;
GList *start_node, *stop_node;
for (start_node = mp3parse->seek_table; start_node;
start_node = start_node->next) {
entry = start_node->data;
if (cur - mp3parse->max_bitreservoir >= entry->timestamp) {
start_entry = entry;
break;
}
}
if (!start_entry) {
start_entry = mp3parse->seek_table->data;
start = start_entry->timestamp;
byte_cur = start_entry->byte;
} else {
start = start_entry->timestamp;
byte_cur = start_entry->byte;
}
for (stop_node = mp3parse->seek_table; stop_node;
stop_node = stop_node->next) {
entry = stop_node->data;
if (stop >= entry->timestamp) {
stop_node = stop_node->prev;
stop_entry = (stop_node) ? stop_node->data : NULL;
break;
}
}
if (!stop_entry) {
byte_stop = -1;
} else {
byte_stop = stop_entry->byte;
}
}
g_mutex_lock (mp3parse->pending_accurate_seeks_lock);
event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type,
byte_cur, stop_type, byte_stop);
if (gst_pad_push_event (mp3parse->sinkpad, event)) {
mp3parse->exact_position = TRUE;
seek->upstream_start = byte_cur;
seek->timestamp_start = start;
mp3parse->pending_accurate_seeks =
g_slist_prepend (mp3parse->pending_accurate_seeks, seek);
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
return TRUE;
} else {
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
mp3parse->exact_position = TRUE;
g_free (seek);
return TRUE;
}
}
mp3parse->exact_position = FALSE;
/* Convert the TIME to the appropriate BYTE position at which to resume
* decoding. */
if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) cur, &byte_cur))
goto no_pos;
if (!mp3parse_time_to_bytepos (mp3parse, (GstClockTime) stop, &byte_stop))
goto no_pos;
GST_DEBUG_OBJECT (mp3parse, "Seeking to byte range %" G_GINT64_FORMAT
" to %" G_GINT64_FORMAT, byte_cur, byte_stop);
/* Send BYTE based seek upstream */
event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type,
byte_cur, stop_type, byte_stop);
if (flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (mp3parse),
gst_message_new_segment_start (GST_OBJECT_CAST (mp3parse),
GST_FORMAT_TIME, cur));
}
return gst_pad_push_event (mp3parse->sinkpad, event);
no_pos:
GST_DEBUG_OBJECT (mp3parse,
"Could not determine byte position for desired time");
return FALSE;
}
static gboolean
mp3parse_src_event (GstPad * pad, GstEvent * event)
{
GstMPEGAudioParse *mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
gboolean res = FALSE;
g_return_val_if_fail (mp3parse != NULL, FALSE);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
res = mp3parse_handle_seek (mp3parse, event);
gst_event_unref (event);
break;
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (mp3parse);
return res;
}
static gboolean
mp3parse_src_query (GstPad * pad, GstQuery * query)
{
GstFormat format;
GstClockTime total;
GstMPEGAudioParse *mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
gboolean res = FALSE;
GstPad *peer;
g_return_val_if_fail (mp3parse != NULL, FALSE);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
gst_query_parse_position (query, &format, NULL);
if (format == GST_FORMAT_BYTES || format == GST_FORMAT_DEFAULT) {
if (mp3parse->cur_offset != -1) {
gst_query_set_position (query, GST_FORMAT_BYTES,
mp3parse->cur_offset);
res = TRUE;
}
} else if (format == GST_FORMAT_TIME) {
if (mp3parse->next_ts == GST_CLOCK_TIME_NONE)
goto out;
gst_query_set_position (query, GST_FORMAT_TIME, mp3parse->next_ts);
res = TRUE;
}
/* If no answer above, see if upstream knows */
if (!res) {
if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
res = gst_pad_query (peer, query);
gst_object_unref (peer);
if (res)
goto out;
}
}
break;
case GST_QUERY_DURATION:
gst_query_parse_duration (query, &format, NULL);
/* First, see if upstream knows */
if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
res = gst_pad_query (peer, query);
gst_object_unref (peer);
if (res)
goto out;
}
if (format == GST_FORMAT_TIME) {
if (!mp3parse_total_time (mp3parse, &total) || total == -1)
goto out;
gst_query_set_duration (query, format, total);
res = TRUE;
}
break;
default:
res = gst_pad_query_default (pad, query);
break;
}
out:
gst_object_unref (mp3parse);
return res;
}
static const GstQueryType *
mp3parse_get_query_types (GstPad * pad ATTR_UNUSED)
{
static const GstQueryType query_types[] = {
GST_QUERY_POSITION,
GST_QUERY_DURATION,
0
};
return query_types;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (mp3parse_debug, "mp3parse", 0, "MP3 Parser");
return gst_element_register (plugin, "mp3parse",
GST_RANK_PRIMARY + 1, GST_TYPE_MP3PARSE);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"mpegaudioparse",
"MPEG-1 layer 1/2/3 audio parser",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
Reference in a new issue View git blame Copy permalink