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gstreamer/gst/mpegaudioparse/gstmpegaudioparse.c
Jan Schmidt 18df8e3250 Move xingmux from -bad. 
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-ugly-plugins-docs.sgml:
* docs/plugins/gst-plugins-ugly-plugins-sections.txt:
* docs/plugins/inspect/plugin-mpegaudioparse.xml:
* gst/mpegaudioparse/Makefile.am:
* gst/mpegaudioparse/gstmpegaudioparse.c:
* gst/mpegaudioparse/gstxingmux.c:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
Move xingmux from -bad.
2008-02-08 00:36:51 +00:00

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/* 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
#define GST_READ_UINT24_BE(p) (p[2] | (p[1] << 8) | (p[0] << 16))
/* 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 gboolean
mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total);
/*static guint gst_mp3parse_signals[LAST_SIGNAL] = { 0 }; */
GST_BOILERPLATE (GstMPEGAudioParse, gst_mp3parse, GstElement, GST_TYPE_ELEMENT);
static const guint mp3types_bitrates[2][3][16] = {
{
{0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448,},
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384,},
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320,}
},
{
{0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256,},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160,}
},
};
static const guint mp3types_freqs[3][3] = { {44100, 48000, 32000},
{22050, 24000, 16000},
{11025, 12000, 8000}
};
static inline guint
mp3_type_frame_length_from_header (GstMPEGAudioParse * mp3parse, guint32 header,
guint * put_version, guint * put_layer, guint * put_channels,
guint * put_bitrate, guint * put_samplerate)
{
guint 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 samplerate)
{
GstCaps *new;
g_assert (layer);
g_assert (samplerate);
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_DEBUG_CATEGORY_INIT (mp3parse_debug, "mp3parse", 0, "MPEG Audio Parser");
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;
mp3parse->xing_frames = 0;
mp3parse->xing_total_time = 0;
mp3parse->xing_bytes = 0;
mp3parse->xing_vbr_scale = 0;
memset (mp3parse->xing_seek_table, 0, 100);
memset (mp3parse->xing_seek_table_inverse, 0, 256);
mp3parse->vbri_bitrate = 0;
mp3parse->vbri_frames = 0;
mp3parse->vbri_total_time = 0;
mp3parse->vbri_bytes = 0;
mp3parse->vbri_seek_points = 0;
g_free (mp3parse->vbri_seek_table);
mp3parse->vbri_seek_table = NULL;
if (mp3parse->seek_table) {
g_list_foreach (mp3parse->seek_table, (GFunc) g_free, NULL);
g_list_free (mp3parse->seek_table);
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);
g_slist_free (mp3parse->pending_accurate_seeks);
mp3parse->pending_accurate_seeks = NULL;
}
g_mutex_unlock (mp3parse->pending_accurate_seeks_lock);
if (mp3parse->pending_segment) {
GstEvent **eventp = &mp3parse->pending_segment;
gst_event_replace (eventp, NULL);
}
mp3parse->exact_position = FALSE;
gst_segment_init (&mp3parse->segment, GST_FORMAT_TIME);
}
static void
gst_mp3parse_init (GstMPEGAudioParse * mp3parse, GstMPEGAudioParseClass * klass)
{
mp3parse->sinkpad =
gst_pad_new_from_static_template (&mp3_sink_template, "sink");
gst_pad_set_event_function (mp3parse->sinkpad, gst_mp3parse_sink_event);
gst_pad_set_chain_function (mp3parse->sinkpad, gst_mp3parse_chain);
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->sinkpad);
mp3parse->srcpad =
gst_pad_new_from_static_template (&mp3_src_template, "src");
gst_pad_use_fixed_caps (mp3parse->srcpad);
gst_pad_set_event_function (mp3parse->srcpad, mp3parse_src_event);
gst_pad_set_query_function (mp3parse->srcpad, mp3parse_src_query);
gst_pad_set_query_type_function (mp3parse->srcpad, mp3parse_get_query_types);
gst_element_add_pad (GST_ELEMENT (mp3parse), mp3parse->srcpad);
mp3parse->adapter = gst_adapter_new ();
mp3parse->pending_accurate_seeks_lock = g_mutex_new ();
gst_mp3parse_reset (mp3parse);
}
static void
gst_mp3parse_dispose (GObject * object)
{
GstMPEGAudioParse *mp3parse = GST_MP3PARSE (object);
gst_mp3parse_reset (mp3parse);
if (mp3parse->adapter) {
g_object_unref (mp3parse->adapter);
mp3parse->adapter = NULL;
}
g_mutex_free (mp3parse->pending_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;
GstEvent **eventp;
mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:
{
gdouble rate, applied_rate;
GstFormat format;
gint64 start, stop, pos;
gboolean update;
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);
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);
/* save the segment for later, right before we push a new buffer so that
* the caps are fixed and the next linked element can receive the segment. */
eventp = &mp3parse->pending_segment;
gst_event_replace (eventp, event);
gst_event_unref (event);
res = TRUE;
break;
}
case GST_EVENT_FLUSH_STOP:
/* Clear our adapter and set up for a new position */
gst_adapter_clear (mp3parse->adapter);
eventp = &mp3parse->pending_segment;
gst_event_replace (eventp, NULL);
res = gst_pad_push_event (mp3parse->srcpad, event);
break;
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 if (mp3parse->vbri_bitrate != 0)
bitrate = mp3parse->vbri_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);
/* push any pending segment now */
if (mp3parse->pending_segment) {
gst_pad_push_event (mp3parse->srcpad, mp3parse->pending_segment);
mp3parse->pending_segment = NULL;
}
ret = gst_pad_push (mp3parse->srcpad, outbuf);
}
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 guint32 vbri_id = 0x56425249; /* 'VBRI' in hex */
gint 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)
offset = 0x11;
else
offset = 0x20;
} else { /* MPEG-2 header */
if (mp3parse->channels == 1)
offset = 0x09;
else
offset = 0x11;
}
/* Skip the 4 bytes of the MP3 header too */
offset += 4;
/* Check if we have enough data to read the Xing header */
avail = gst_adapter_available (mp3parse->adapter);
if (avail < offset + 8)
return;
data = gst_adapter_peek (mp3parse->adapter, offset + 8);
if (data == NULL)
return;
/* The header starts at the provided offset */
data += offset;
read_id = GST_READ_UINT32_BE (data);
if (read_id == xing_id || read_id == info_id) {
guint32 xing_flags;
guint bytes_needed = offset + 8;
gint64 total_bytes;
GstClockTime total_time;
GST_DEBUG_OBJECT (mp3parse, "Found Xing header marker 0x%x", xing_id);
/* Read 4 base bytes of flags, big-endian */
xing_flags = GST_READ_UINT32_BE (data + 4);
if (xing_flags & XING_FRAMES_FLAG)
bytes_needed += 4;
if (xing_flags & XING_BYTES_FLAG)
bytes_needed += 4;
if (xing_flags & XING_TOC_FLAG)
bytes_needed += 100;
if (xing_flags & XING_VBR_SCALE_FLAG)
bytes_needed += 4;
if (avail < bytes_needed) {
GST_DEBUG_OBJECT (mp3parse,
"Not enough data to read Xing header (need %d)", bytes_needed);
return;
}
GST_DEBUG_OBJECT (mp3parse, "Reading Xing header");
mp3parse->xing_flags = xing_flags;
data = gst_adapter_peek (mp3parse->adapter, bytes_needed);
data += offset + 8;
if (xing_flags & XING_FRAMES_FLAG) {
mp3parse->xing_frames = GST_READ_UINT32_BE (data);
if (mp3parse->xing_frames == 0) {
GST_WARNING_OBJECT (mp3parse,
"Invalid number of frames in Xing header");
mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
} else {
mp3parse->xing_total_time = gst_util_uint64_scale (GST_SECOND,
(guint64) (mp3parse->xing_frames) * (mp3parse->spf),
mp3parse->rate);
}
data += 4;
} else {
mp3parse->xing_frames = 0;
mp3parse->xing_total_time = 0;
}
if (xing_flags & XING_BYTES_FLAG) {
mp3parse->xing_bytes = GST_READ_UINT32_BE (data);
if (mp3parse->xing_bytes == 0) {
GST_WARNING_OBJECT (mp3parse, "Invalid number of bytes in Xing header");
mp3parse->xing_flags &= ~XING_BYTES_FLAG;
}
data += 4;
} else {
mp3parse->xing_bytes = 0;
}
/* If we know the upstream size and duration, compute the
* total bitrate, rounded up to the nearest kbit/sec */
if (mp3parse_total_time (mp3parse, &total_time) &&
mp3parse_total_bytes (mp3parse, &total_bytes)) {
mp3parse->xing_bitrate = gst_util_uint64_scale (total_bytes,
8 * GST_SECOND, total_time);
mp3parse->xing_bitrate += 500;
mp3parse->xing_bitrate -= mp3parse->xing_bitrate % 1000;
}
if (xing_flags & XING_TOC_FLAG) {
int i, percent = 0;
guchar *table = mp3parse->xing_seek_table;
guchar old = 0;
if (data[0] != 0) {
GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
mp3parse->xing_flags &= ~XING_TOC_FLAG;
goto skip_toc;
}
/* xing seek table: percent time -> 1/256 bytepos */
for (i = 0; i < 100; i++) {
mp3parse->xing_seek_table[i] = data[i];
if (old > data[i]) {
GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
mp3parse->xing_flags &= ~XING_TOC_FLAG;
goto skip_toc;
}
}
/* 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);
}
}
skip_toc:
data += 100;
} else {
memset (mp3parse->xing_seek_table, 0, 100);
memset (mp3parse->xing_seek_table_inverse, 0, 256);
}
if (xing_flags & XING_VBR_SCALE_FLAG) {
mp3parse->xing_vbr_scale = GST_READ_UINT32_BE (data);
data += 4;
} else
mp3parse->xing_vbr_scale = 0;
GST_DEBUG_OBJECT (mp3parse, "Xing header reported %u frames, time %"
GST_TIME_FORMAT ", %u bytes, vbr scale %u", mp3parse->xing_frames,
GST_TIME_ARGS (mp3parse->xing_total_time), mp3parse->xing_bytes,
mp3parse->xing_vbr_scale);
} else if (read_id == vbri_id) {
gint64 total_bytes, total_frames;
GstClockTime total_time;
guint16 nseek_points;
GST_DEBUG_OBJECT (mp3parse, "Found VBRI header marker 0x%x", vbri_id);
if (avail < offset + 26) {
GST_DEBUG_OBJECT (mp3parse,
"Not enough data to read VBRI header (need %d)", offset + 26);
return;
}
GST_DEBUG_OBJECT (mp3parse, "Reading VBRI header");
data = gst_adapter_peek (mp3parse->adapter, offset + 26);
data += offset + 4;
if (GST_READ_UINT16_BE (data) != 0x0001) {
GST_WARNING_OBJECT (mp3parse,
"Unsupported VBRI version 0x%x", GST_READ_UINT16_BE (data));
return;
}
data += 2;
/* Skip encoder delay */
data += 2;
/* Skip quality */
data += 2;
total_bytes = GST_READ_UINT32_BE (data);
if (total_bytes != 0)
mp3parse->vbri_bytes = total_bytes;
data += 4;
total_frames = GST_READ_UINT32_BE (data);
if (total_frames != 0) {
mp3parse->vbri_frames = total_frames;
mp3parse->vbri_total_time = gst_util_uint64_scale (GST_SECOND,
(guint64) (mp3parse->vbri_frames) * (mp3parse->spf), mp3parse->rate);
}
data += 4;
/* If we know the upstream size and duration, compute the
* total bitrate, rounded up to the nearest kbit/sec */
if (mp3parse_total_time (mp3parse, &total_time) &&
mp3parse_total_bytes (mp3parse, &total_bytes)) {
mp3parse->vbri_bitrate = gst_util_uint64_scale (total_bytes,
8 * GST_SECOND, total_time);
mp3parse->vbri_bitrate += 500;
mp3parse->vbri_bitrate -= mp3parse->vbri_bitrate % 1000;
}
nseek_points = GST_READ_UINT16_BE (data);
data += 2;
if (nseek_points > 0) {
guint scale, seek_bytes, seek_frames;
gint i;
mp3parse->vbri_seek_points = nseek_points;
scale = GST_READ_UINT16_BE (data);
data += 2;
seek_bytes = GST_READ_UINT16_BE (data);
data += 2;
seek_frames = GST_READ_UINT16_BE (data);
data += 2;
if (scale == 0 || seek_bytes == 0 || seek_bytes > 4 || seek_frames == 0) {
GST_WARNING_OBJECT (mp3parse, "Unsupported VBRI seek table");
goto out_vbri;
}
if (avail < offset + 26 + nseek_points * seek_bytes) {
GST_WARNING_OBJECT (mp3parse,
"Not enough data to read VBRI seek table (need %d)",
offset + 26 + nseek_points * seek_bytes);
goto out_vbri;
}
if (seek_frames * nseek_points < total_frames - seek_frames ||
seek_frames * nseek_points > total_frames + seek_frames) {
GST_WARNING_OBJECT (mp3parse,
"VBRI seek table doesn't cover the complete file");
goto out_vbri;
}
data =
gst_adapter_peek (mp3parse->adapter,
offset + 26 + nseek_points * seek_bytes);
data += offset + 26;
/* VBRI seek table: frame/seek_frames -> byte */
mp3parse->vbri_seek_table = g_new (guint32, nseek_points);
if (seek_bytes == 4)
for (i = 0; i < nseek_points; i++) {
mp3parse->vbri_seek_table[i] = GST_READ_UINT32_BE (data) * scale;
data += 4;
} else if (seek_bytes == 3)
for (i = 0; i < nseek_points; i++) {
mp3parse->vbri_seek_table[i] = GST_READ_UINT24_BE (data) * scale;
data += 3;
} else if (seek_bytes == 2)
for (i = 0; i < nseek_points; i++) {
mp3parse->vbri_seek_table[i] = GST_READ_UINT16_BE (data) * scale;
data += 2;
} else /* seek_bytes == 1 */
for (i = 0; i < nseek_points; i++) {
mp3parse->vbri_seek_table[i] = GST_READ_UINT8 (data) * scale;
data += 1;
}
}
out_vbri:
GST_DEBUG_OBJECT (mp3parse, "VBRI header reported %u frames, time %"
GST_TIME_FORMAT ", bytes %u", mp3parse->vbri_frames,
GST_TIME_ARGS (mp3parse->vbri_total_time), mp3parse->vbri_bytes);
} else {
GST_DEBUG_OBJECT (mp3parse,
"Xing, LAME or VBRI 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) {
GstCaps *caps;
caps = mp3_caps_create (layer, channels, rate);
gst_pad_set_caps (mp3parse->srcpad, caps);
gst_caps_unref (caps);
mp3parse->channels = channels;
mp3parse->rate = rate;
}
if (layer != mp3parse->layer || version != mp3parse->version) {
mp3parse->layer = layer;
mp3parse->version = version;
/* see http://www.codeproject.com/audio/MPEGAudioInfo.asp */
if (mp3parse->layer == 1)
mp3parse->spf = 384;
else if (mp3parse->layer == 2)
mp3parse->spf = 1152;
else if (mp3parse->version == 2) {
mp3parse->spf = 576;
} else
mp3parse->spf = 1152;
}
mp3parse->bit_rate = bitrate;
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;
}
if (mp3parse->vbri_bytes != 0) {
*total = mp3parse->vbri_bytes;
return TRUE;
}
return FALSE;
}
static gboolean
mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total)
{
gint64 total_bytes;
*total = GST_CLOCK_TIME_NONE;
if (mp3parse->xing_flags & XING_FRAMES_FLAG) {
*total = mp3parse->xing_total_time;
return TRUE;
}
if (mp3parse->vbri_total_time != 0) {
*total = mp3parse->vbri_total_time;
return TRUE;
}
/* Calculate time from the measured bitrate */
if (!mp3parse_total_bytes (mp3parse, &total_bytes))
return FALSE;
if (total_bytes != -1
&& !mp3parse_bytepos_to_time (mp3parse, total_bytes, total))
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 * gst_util_guint64_to_gdouble (ts)) /
gst_util_guint64_to_gdouble (total_time), 0.0, 100.0);
gint index = CLAMP (percent, 0, 99);
fa = mp3parse->xing_seek_table[index];
if (index < 99)
fb = mp3parse->xing_seek_table[index + 1];
else
fb = 256.0;
fx = fa + (fb - fa) * (percent - index);
*bytepos = (1.0 / 256.0) * fx * total_bytes;
return TRUE;
}
if (mp3parse->vbri_seek_table &&
mp3parse_total_bytes (mp3parse, &total_bytes) &&
mp3parse_total_time (mp3parse, &total_time)) {
gint i, j;
gdouble a, b, fa, fb;
i = gst_util_uint64_scale (ts, mp3parse->vbri_seek_points - 1, total_time);
i = CLAMP (i, 0, mp3parse->vbri_seek_points - 1);
a = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
mp3parse->vbri_seek_points));
fa = 0.0;
for (j = i; j >= 0; j--)
fa += mp3parse->vbri_seek_table[j];
if (i + 1 < mp3parse->vbri_seek_points) {
b = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
mp3parse->vbri_seek_points));
fb = fa + mp3parse->vbri_seek_table[i + 1];
} else {
b = gst_guint64_to_gdouble (total_time);
fb = total_bytes;
}
*bytepos = fa + ((fb - fa) / (b - a)) * (gst_guint64_to_gdouble (ts) - a);
return TRUE;
}
if (mp3parse->avg_bitrate == 0)
goto no_bitrate;
*bytepos =
gst_util_uint64_scale (ts, mp3parse->avg_bitrate, (8 * GST_SECOND));
return TRUE;
no_bitrate:
GST_DEBUG_OBJECT (mp3parse, "Cannot seek yet - no average bitrate");
return FALSE;
}
static gboolean
mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
gint64 bytepos, GstClockTime * ts)
{
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 * gst_util_guint64_to_gdouble (total_time);
return TRUE;
}
if (mp3parse->vbri_seek_table &&
mp3parse_total_bytes (mp3parse, &total_bytes) &&
mp3parse_total_time (mp3parse, &total_time)) {
gint i = 0;
guint64 sum = 0;
gdouble a, b, fa, fb;
do {
sum += mp3parse->vbri_seek_table[i];
i++;
} while (i + 1 < mp3parse->vbri_seek_points
&& sum + mp3parse->vbri_seek_table[i] < bytepos);
i--;
a = gst_guint64_to_gdouble (sum);
fa = gst_guint64_to_gdouble (gst_util_uint64_scale (i, total_time,
mp3parse->vbri_seek_points));
if (i + 1 < mp3parse->vbri_seek_points) {
b = a + mp3parse->vbri_seek_table[i + 1];
fb = gst_guint64_to_gdouble (gst_util_uint64_scale (i + 1, total_time,
mp3parse->vbri_seek_points));
} else {
b = total_bytes;
fb = gst_guint64_to_gdouble (total_time);
}
*ts = gst_gdouble_to_guint64 (fa + ((fb - fa) / (b - a)) * (bytepos - a));
return TRUE;
}
/* Cannot convert anything except 0 if we don't have a bitrate yet */
if (mp3parse->avg_bitrate == 0)
return FALSE;
*ts = (GstClockTime) gst_util_uint64_scale (GST_SECOND, bytepos * 8,
mp3parse->avg_bitrate);
return TRUE;
}
static gboolean
mp3parse_handle_seek (GstMPEGAudioParse * mp3parse, GstEvent * event)
{
GstFormat format;
gdouble rate;
GstSeekFlags flags;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
gint64 byte_cur, byte_stop;
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);
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;
}
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