--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_good/gst/mpegaudioparse/gstmpegaudioparse.c Fri May 14 18:43:44 2010 -0500
@@ -0,0 +1,2199 @@
+/* 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 MP3_CHANNEL_MODE_UNKNOWN -1
+#define MP3_CHANNEL_MODE_STEREO 0
+#define MP3_CHANNEL_MODE_JOINT_STEREO 1
+#define MP3_CHANNEL_MODE_DUAL_CHANNEL 2
+#define MP3_CHANNEL_MODE_MONO 3
+
+#define CRC_UNKNOWN -1
+#define CRC_PROTECTED 0
+#define CRC_NOT_PROTECTED 1
+
+#define XING_FRAMES_FLAG 0x0001
+#define XING_BYTES_FLAG 0x0002
+#define XING_TOC_FLAG 0x0004
+#define XING_VBR_SCALE_FLAG 0x0008
+
+#ifndef GST_READ_UINT24_BE
+#define GST_READ_UINT24_BE(p) (p[2] | (p[1] << 8) | (p[0] << 16))
+#endif
+
+/* Minimum number of consecutive, valid-looking frames to consider
+ for resyncing */
+#define MIN_RESYNC_FRAMES 3
+
+static inline MPEGAudioSeekEntry *
+mpeg_audio_seek_entry_new ()
+{
+ return g_slice_new (MPEGAudioSeekEntry);
+}
+
+static inline void
+mpeg_audio_seek_entry_free (MPEGAudioSeekEntry * entry)
+{
+ g_slice_free (MPEGAudioSeekEntry, entry);
+}
+
+/* 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 GstFlowReturn
+gst_mp3parse_handle_data (GstMPEGAudioParse * mp3parse, gboolean at_eos);
+
+static gboolean mp3parse_bytepos_to_time (GstMPEGAudioParse * mp3parse,
+ gint64 bytepos, GstClockTime * ts, gboolean from_total_time);
+static gboolean
+mp3parse_total_bytes (GstMPEGAudioParse * mp3parse, gint64 * total);
+static gboolean
+mp3parse_total_time (GstMPEGAudioParse * mp3parse, GstClockTime * total);
+
+GST_BOILERPLATE (GstMPEGAudioParse, gst_mp3parse, GstElement, GST_TYPE_ELEMENT);
+
+#define GST_TYPE_MP3_CHANNEL_MODE (gst_mp3_channel_mode_get_type())
+
+static const GEnumValue mp3_channel_mode[] = {
+ {MP3_CHANNEL_MODE_UNKNOWN, "Unknown", "unknown"},
+ {MP3_CHANNEL_MODE_MONO, "Mono", "mono"},
+ {MP3_CHANNEL_MODE_DUAL_CHANNEL, "Dual Channel", "dual-channel"},
+ {MP3_CHANNEL_MODE_JOINT_STEREO, "Joint Stereo", "joint-stereo"},
+ {MP3_CHANNEL_MODE_STEREO, "Stereo", "stereo"},
+ {0, NULL, NULL},
+};
+
+static GType
+gst_mp3_channel_mode_get_type (void)
+{
+ static GType mp3_channel_mode_type = 0;
+
+ if (!mp3_channel_mode_type) {
+ mp3_channel_mode_type =
+ g_enum_register_static ("GstMp3ChannelMode", mp3_channel_mode);
+ }
+ return mp3_channel_mode_type;
+}
+
+static const gchar *
+gst_mp3_channel_mode_get_nick (gint mode)
+{
+ guint i;
+ for (i = 0; i < G_N_ELEMENTS (mp3_channel_mode); i++) {
+ if (mp3_channel_mode[i].value == mode)
+ return mp3_channel_mode[i].value_nick;
+ }
+ return NULL;
+}
+
+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 * put_mode,
+ guint * put_crc)
+{
+ guint length;
+ gulong mode, samplerate, bitrate, layer, channels, padding, crc;
+ gulong version;
+ gint lsf, mpg25;
+
+ if (header & (1 << 20)) {
+ lsf = (header & (1 << 19)) ? 0 : 1;
+ mpg25 = 0;
+ } else {
+ lsf = 1;
+ mpg25 = 1;
+ }
+
+ version = 1 + lsf + mpg25;
+
+ layer = 4 - ((header >> 17) & 0x3);
+
+ crc = (header >> 16) & 0x1;
+
+ bitrate = (header >> 12) & 0xF;
+ bitrate = mp3types_bitrates[lsf][layer - 1][bitrate] * 1000;
+ /* The caller has ensured we have a valid header, so bitrate can't be
+ zero here. */
+ g_assert (bitrate != 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, version = %lu, "
+ "layer = %lu, channels = %lu, mode = %s", samplerate, bitrate, version,
+ layer, channels, gst_mp3_channel_mode_get_nick (mode));
+
+ if (put_version)
+ *put_version = version;
+ if (put_layer)
+ *put_layer = layer;
+ if (put_channels)
+ *put_channels = channels;
+ if (put_bitrate)
+ *put_bitrate = bitrate;
+ if (put_samplerate)
+ *put_samplerate = samplerate;
+ if (put_mode)
+ *put_mode = mode;
+ if (put_crc)
+ *put_crc = crc;
+
+ return length;
+}
+
+static GstCaps *
+mp3_caps_create (guint version, guint layer, guint channels, guint samplerate)
+{
+ GstCaps *new;
+
+ g_assert (version);
+ g_assert (layer);
+ g_assert (samplerate);
+ g_assert (channels);
+
+ new = gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, 1,
+ "mpegaudioversion", G_TYPE_INT, version,
+ "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;
+
+/* register tags */
+#define GST_TAG_CRC "has-crc"
+#define GST_TAG_MODE "channel-mode"
+
+ gst_tag_register (GST_TAG_CRC, GST_TAG_FLAG_META, G_TYPE_BOOLEAN,
+ "has crc", "Using CRC", NULL);
+ gst_tag_register (GST_TAG_MODE, GST_TAG_FLAG_ENCODED, G_TYPE_STRING,
+ "channel mode", "MPEG audio channel mode", NULL);
+
+ g_type_class_ref (GST_TYPE_MP3_CHANNEL_MODE);
+}
+
+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->sync_offset = 0;
+ 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->last_posted_crc = CRC_UNKNOWN;
+ mp3parse->last_posted_channel_mode = MP3_CHANNEL_MODE_UNKNOWN;
+
+ 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) mpeg_audio_seek_entry_free,
+ NULL);
+ g_list_free (mp3parse->seek_table);
+ mp3parse->seek_table = NULL;
+ }
+
+ g_mutex_lock (mp3parse->pending_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;
+ }
+ if (mp3parse->pending_nonaccurate_seeks) {
+ g_slist_foreach (mp3parse->pending_nonaccurate_seeks, (GFunc) g_free, NULL);
+ g_slist_free (mp3parse->pending_nonaccurate_seeks);
+ mp3parse->pending_nonaccurate_seeks = NULL;
+ }
+ g_mutex_unlock (mp3parse->pending_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_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_seeks_lock);
+ mp3parse->pending_seeks_lock = NULL;
+
+ g_list_foreach (mp3parse->pending_events, (GFunc) gst_mini_object_unref,
+ NULL);
+ g_list_free (mp3parse->pending_events);
+ mp3parse->pending_events = NULL;
+
+ G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static gboolean
+gst_mp3parse_sink_event (GstPad * pad, GstEvent * event)
+{
+ gboolean res = TRUE;
+ 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;
+ MPEGAudioPendingAccurateSeek *seek = NULL;
+ GSList *node;
+
+ gst_event_parse_new_segment_full (event, &update, &rate, &applied_rate,
+ &format, &start, &stop, &pos);
+
+ g_mutex_lock (mp3parse->pending_seeks_lock);
+ if (format == GST_FORMAT_BYTES && mp3parse->pending_accurate_seeks) {
+
+ 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;
+ mp3parse->sync_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 %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT
+ ", pos %" G_GINT64_FORMAT, 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_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_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, FALSE))
+ seg_stop = GST_CLOCK_TIME_NONE;
+ if (mp3parse_bytepos_to_time (mp3parse, start, &seg_start, FALSE) &&
+ mp3parse_bytepos_to_time (mp3parse, pos, &seg_pos, FALSE)) {
+ gst_event_unref (event);
+
+ /* search the pending nonaccurate seeks */
+ g_mutex_lock (mp3parse->pending_seeks_lock);
+ seek = NULL;
+ for (node = mp3parse->pending_nonaccurate_seeks; node;
+ node = node->next) {
+ MPEGAudioPendingAccurateSeek *tmp = node->data;
+
+ if (tmp->upstream_start == pos) {
+ seek = tmp;
+ break;
+ }
+ }
+
+ if (seek) {
+ if (seek->segment.stop == -1) {
+ /* corrent the segment end, because non-accurate seeks might make
+ * our streaming end earlier (see bug #603695) */
+ seg_stop = -1;
+ }
+ g_free (seek);
+ mp3parse->pending_nonaccurate_seeks =
+ g_slist_delete_link (mp3parse->pending_nonaccurate_seeks, node);
+ }
+ g_mutex_unlock (mp3parse->pending_seeks_lock);
+
+ 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;
+ mp3parse->sync_offset = 0;
+ /* also clear leftover data if clearing so much state */
+ gst_adapter_clear (mp3parse->adapter);
+
+ 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 %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT
+ ", pos %" G_GINT64_FORMAT, 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;
+ case GST_EVENT_EOS:
+ /* If we haven't processed any frames yet, then make sure we process
+ at least whatever's in our adapter */
+ if (mp3parse->frame_count == 0) {
+ gst_mp3parse_handle_data (mp3parse, TRUE);
+
+ /* If we STILL have zero frames processed, fire an error */
+ if (mp3parse->frame_count == 0) {
+ GST_ELEMENT_ERROR (mp3parse, STREAM, WRONG_TYPE,
+ ("No valid frames found before end of stream"), (NULL));
+ }
+ }
+ /* fall through */
+ default:
+ if (mp3parse->pending_segment &&
+ (GST_EVENT_TYPE (event) != GST_EVENT_EOS) &&
+ (GST_EVENT_TYPE (event) != GST_EVENT_FLUSH_START)) {
+ /* Cache all events except EOS and the ones above if we have
+ * a pending segment */
+ mp3parse->pending_events =
+ g_list_append (mp3parse->pending_events, event);
+ } else {
+ res = gst_pad_push_event (mp3parse->srcpad, event);
+ }
+ break;
+ }
+
+ gst_object_unref (mp3parse);
+
+ return res;
+}
+
+static void
+gst_mp3parse_add_index_entry (GstMPEGAudioParse * mp3parse, guint64 offset,
+ GstClockTime ts)
+{
+ MPEGAudioSeekEntry *entry, *last;
+
+ if (G_LIKELY (mp3parse->seek_table != NULL)) {
+ last = mp3parse->seek_table->data;
+
+ if (last->byte >= offset)
+ return;
+
+ if (GST_CLOCK_DIFF (last->timestamp, ts) < mp3parse->idx_interval)
+ return;
+ }
+
+ entry = mpeg_audio_seek_entry_new ();
+ entry->byte = offset;
+ entry->timestamp = ts;
+ mp3parse->seek_table = g_list_prepend (mp3parse->seek_table, entry);
+
+ GST_LOG_OBJECT (mp3parse, "Adding index entry %" GST_TIME_FORMAT " @ offset "
+ "0x%08" G_GINT64_MODIFIER "x", GST_TIME_ARGS (ts), offset);
+}
+
+/* Prepare a buffer of the indicated size, timestamp it and output */
+static GstFlowReturn
+gst_mp3parse_emit_frame (GstMPEGAudioParse * mp3parse, guint size,
+ guint mode, guint crc)
+{
+ GstBuffer *outbuf;
+ guint bitrate;
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstClockTime push_start;
+ GstTagList *taglist;
+
+ 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
+ * half a frame, then take it instead of our calculated timestamp.
+ * This avoids creating imperfect streams just because of
+ * quantization in the container timestamping */
+ GstClockTimeDiff diff = mp3parse->next_ts - mp3parse->pending_ts;
+ GstClockTimeDiff thresh = GST_BUFFER_DURATION (outbuf) / 2;
+
+ if (diff < -thresh || diff > thresh) {
+ GST_DEBUG_OBJECT (mp3parse, "Updating next_ts from %" GST_TIME_FORMAT
+ " to pending ts %" GST_TIME_FORMAT
+ " at offset %" G_GINT64_FORMAT " (pending offset was %"
+ G_GINT64_FORMAT ")", GST_TIME_ARGS (mp3parse->next_ts),
+ GST_TIME_ARGS (mp3parse->pending_ts), mp3parse->tracked_offset,
+ mp3parse->pending_offset);
+ 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, FALSE) &&
+ GST_CLOCK_TIME_IS_VALID (ts))
+ GST_BUFFER_TIMESTAMP (outbuf) = ts;
+ else {
+ GST_BUFFER_TIMESTAMP (outbuf) = 0;
+ }
+ }
+
+ if (GST_BUFFER_TIMESTAMP (outbuf) == 0)
+ mp3parse->exact_position = TRUE;
+
+ if (mp3parse->seekable &&
+ mp3parse->exact_position && GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) &&
+ mp3parse->cur_offset != GST_BUFFER_OFFSET_NONE) {
+ gst_mp3parse_add_index_entry (mp3parse, mp3parse->cur_offset,
+ GST_BUFFER_TIMESTAMP (outbuf));
+ }
+
+ /* Update our byte offset tracking */
+ if (mp3parse->cur_offset != -1) {
+ mp3parse->cur_offset += size;
+ }
+ mp3parse->tracked_offset += size;
+
+ if (GST_BUFFER_TIMESTAMP_IS_VALID (outbuf))
+ 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;
+
+ /* we will create a taglist (if any of the parameters has changed)
+ * to add the tags that changed */
+ taglist = NULL;
+ if ((mp3parse->last_posted_bitrate / 10000) != (bitrate / 10000)) {
+ 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);
+
+ /* Post a new duration message if the average bitrate changes that much
+ * so applications can update their cached values
+ */
+ if ((mp3parse->xing_flags & XING_TOC_FLAG) == 0
+ && mp3parse->vbri_total_time == 0) {
+ gst_element_post_message (GST_ELEMENT (mp3parse),
+ gst_message_new_duration (GST_OBJECT (mp3parse), GST_FORMAT_TIME,
+ -1));
+ }
+ }
+
+ if (mp3parse->last_posted_crc != crc) {
+ gboolean using_crc;
+
+ if (!taglist) {
+ taglist = gst_tag_list_new ();
+ }
+ mp3parse->last_posted_crc = crc;
+ if (mp3parse->last_posted_crc == CRC_PROTECTED) {
+ using_crc = TRUE;
+ } else {
+ using_crc = FALSE;
+ }
+ gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_CRC,
+ using_crc, NULL);
+ }
+
+ if (mp3parse->last_posted_channel_mode != mode) {
+ if (!taglist) {
+ taglist = gst_tag_list_new ();
+ }
+ mp3parse->last_posted_channel_mode = mode;
+
+ gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE, GST_TAG_MODE,
+ gst_mp3_channel_mode_get_nick (mode), NULL);
+ }
+
+ /* if the taglist exists, we need to send it */
+ if (taglist) {
+ 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 (GST_CLOCK_TIME_IS_VALID (mp3parse->segment.start) &&
+ 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_IS_VALID (outbuf) &&
+ GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf)
+ < push_start))) {
+ GST_DEBUG_OBJECT (mp3parse,
+ "Buffer before configured segment range %" GST_TIME_FORMAT
+ " to %" GST_TIME_FORMAT ", dropping, timestamp %"
+ GST_TIME_FORMAT " duration %" 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_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
+ GST_BUFFER_OFFSET (outbuf));
+
+ gst_buffer_unref (outbuf);
+ ret = GST_FLOW_OK;
+ } else if (G_UNLIKELY (GST_BUFFER_TIMESTAMP_IS_VALID (outbuf) &&
+ GST_CLOCK_TIME_IS_VALID (mp3parse->segment.stop) &&
+ GST_BUFFER_TIMESTAMP (outbuf) >=
+ mp3parse->segment.stop + GST_BUFFER_DURATION (outbuf))) {
+ /* Some mp3 streams have an offset in the timestamps, for which we have to
+ * push the frame *after* the end position in order for the decoder to be
+ * able to decode everything up until the segment.stop position.
+ * That is the reason of the calculated offset */
+ GST_DEBUG_OBJECT (mp3parse,
+ "Buffer after configured segment range %" GST_TIME_FORMAT " to %"
+ GST_TIME_FORMAT ", returning GST_FLOW_UNEXPECTED, timestamp %"
+ GST_TIME_FORMAT " duration %" 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_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
+ GST_BUFFER_OFFSET (outbuf));
+
+ gst_buffer_unref (outbuf);
+ ret = GST_FLOW_UNEXPECTED;
+ } 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;
+ }
+ if (mp3parse->pending_events) {
+ GList *l;
+
+ for (l = mp3parse->pending_events; l != NULL; l = l->next) {
+ gst_pad_push_event (mp3parse->srcpad, GST_EVENT (l->data));
+ }
+ g_list_free (mp3parse->pending_events);
+ mp3parse->pending_events = NULL;
+ }
+
+ /* set discont if needed */
+ if (mp3parse->discont) {
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ mp3parse->discont = FALSE;
+ }
+
+ ret = gst_pad_push (mp3parse->srcpad, outbuf);
+ }
+
+ return ret;
+}
+
+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;
+ gint64 upstream_total_bytes = 0;
+ 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;
+
+ /* obtain real upstream total bytes */
+ mp3parse_total_bytes (mp3parse, &upstream_total_bytes);
+
+ 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 ((total_time = mp3parse->xing_total_time) &&
+ (total_bytes = mp3parse->xing_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, new;
+ guint first;
+
+ first = data[0];
+ GST_DEBUG_OBJECT (mp3parse,
+ "Subtracting initial offset of %d bytes from Xing TOC", first);
+
+ /* xing seek table: percent time -> 1/256 bytepos */
+ for (i = 0; i < 100; i++) {
+ new = data[i] - first;
+ if (old > new) {
+ GST_WARNING_OBJECT (mp3parse, "Skipping broken Xing TOC");
+ mp3parse->xing_flags &= ~XING_TOC_FLAG;
+ goto skip_toc;
+ }
+ mp3parse->xing_seek_table[i] = old = new;
+ }
+
+ /* 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 == 99) {
+ gdouble fa, fb, fx;
+ gint a = percent, 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);
+ } 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);
+
+ /* check for truncated file */
+ if (upstream_total_bytes && mp3parse->xing_bytes &&
+ mp3parse->xing_bytes * 0.8 > upstream_total_bytes) {
+ GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
+ "invalidating Xing header duration and size");
+ mp3parse->xing_flags &= ~XING_BYTES_FLAG;
+ mp3parse->xing_flags &= ~XING_FRAMES_FLAG;
+ }
+ } 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 ((total_time = mp3parse->vbri_total_time) &&
+ (total_bytes = mp3parse->vbri_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);
+
+ 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);
+
+ /* check for truncated file */
+ if (upstream_total_bytes && mp3parse->vbri_bytes &&
+ mp3parse->vbri_bytes * 0.8 > upstream_total_bytes) {
+ GST_WARNING_OBJECT (mp3parse, "File appears to have been truncated; "
+ "invalidating VBRI header duration and size");
+ mp3parse->vbri_valid = FALSE;
+ } else {
+ mp3parse->vbri_valid = TRUE;
+ }
+ } else {
+ GST_DEBUG_OBJECT (mp3parse,
+ "Xing, LAME or VBRI header not found in first frame");
+ }
+}
+
+static void
+gst_mp3parse_check_seekability (GstMPEGAudioParse * mp3parse)
+{
+ GstQuery *query;
+ gboolean seekable = FALSE;
+ gint64 start = -1, stop = -1;
+ guint idx_interval = 0;
+
+ query = gst_query_new_seeking (GST_FORMAT_BYTES);
+ if (!gst_pad_peer_query (mp3parse->sinkpad, query)) {
+ GST_DEBUG_OBJECT (mp3parse, "seeking query failed");
+ goto done;
+ }
+
+ gst_query_parse_seeking (query, NULL, &seekable, &start, &stop);
+
+ /* try harder to query upstream size if we didn't get it the first time */
+ if (seekable && stop == -1) {
+ GstFormat fmt = GST_FORMAT_BYTES;
+
+ GST_DEBUG_OBJECT (mp3parse, "doing duration query to fix up unset stop");
+ gst_pad_query_peer_duration (mp3parse->sinkpad, &fmt, &stop);
+ }
+
+ /* if upstream doesn't know the size, it's likely that it's not seekable in
+ * practice even if it technically may be seekable */
+ if (seekable && (start != 0 || stop <= start)) {
+ GST_DEBUG_OBJECT (mp3parse, "seekable but unknown start/stop -> disable");
+ seekable = FALSE;
+ }
+
+ /* let's not put every single frame into our index */
+ if (seekable) {
+ if (stop < 10 * 1024 * 1024)
+ idx_interval = 100;
+ else if (stop < 100 * 1024 * 1024)
+ idx_interval = 500;
+ else
+ idx_interval = 1000;
+ }
+
+done:
+
+ GST_INFO_OBJECT (mp3parse, "seekable: %d (%" G_GUINT64_FORMAT " - %"
+ G_GUINT64_FORMAT ")", seekable, start, stop);
+ mp3parse->seekable = seekable;
+
+ GST_INFO_OBJECT (mp3parse, "idx_interval: %ums", idx_interval);
+ mp3parse->idx_interval = idx_interval * GST_MSECOND;
+
+ gst_query_unref (query);
+}
+
+/* Flush some number of bytes and update tracked offsets */
+static void
+gst_mp3parse_flush_bytes (GstMPEGAudioParse * mp3parse, int bytes)
+{
+ gst_adapter_flush (mp3parse->adapter, bytes);
+ if (mp3parse->cur_offset != -1)
+ mp3parse->cur_offset += bytes;
+ mp3parse->tracked_offset += bytes;
+}
+
+/* Perform extended validation to check that subsequent headers match
+ the first header given here in important characteristics, to avoid
+ false sync. We look for a minimum of MIN_RESYNC_FRAMES consecutive
+ frames to match their major characteristics.
+
+ If at_eos is set to TRUE, we just check that we don't find any invalid
+ frames in whatever data is available, rather than requiring a full
+ MIN_RESYNC_FRAMES of data.
+
+ Returns TRUE if we've seen enough data to validate or reject the frame.
+ If TRUE is returned, then *valid contains TRUE if it validated, or false
+ if we decided it was false sync.
+ */
+static gboolean
+gst_mp3parse_validate_extended (GstMPEGAudioParse * mp3parse, guint32 header,
+ int bpf, gboolean at_eos, gboolean * valid)
+{
+ guint32 next_header;
+ const guint8 *data;
+ guint available;
+ int frames_found = 1;
+ int offset = bpf;
+
+ while (frames_found < MIN_RESYNC_FRAMES) {
+ /* Check if we have enough data for all these frames, plus the next
+ frame header. */
+ available = gst_adapter_available (mp3parse->adapter);
+ if (available < offset + 4) {
+ if (at_eos) {
+ /* Running out of data at EOS is fine; just accept it */
+ *valid = TRUE;
+ return TRUE;
+ } else {
+ return FALSE;
+ }
+ }
+
+ data = gst_adapter_peek (mp3parse->adapter, offset + 4);
+ next_header = GST_READ_UINT32_BE (data + offset);
+ GST_DEBUG_OBJECT (mp3parse, "At %d: header=%08X, header2=%08X, bpf=%d",
+ offset, (unsigned int) header, (unsigned int) next_header, bpf);
+
+/* mask the bits which are allowed to differ between frames */
+#define HDRMASK ~((0xF << 12) /* bitrate */ | \
+ (0x1 << 9) /* padding */ | \
+ (0xf << 4) /* mode|mode extension */ | \
+ (0xf)) /* copyright|emphasis */
+
+ if ((next_header & HDRMASK) != (header & HDRMASK)) {
+ /* If any of the unmasked bits don't match, then it's not valid */
+ GST_DEBUG_OBJECT (mp3parse, "next header doesn't match "
+ "(header=%08X (%08X), header2=%08X (%08X), bpf=%d)",
+ (guint) header, (guint) header & HDRMASK, (guint) next_header,
+ (guint) next_header & HDRMASK, bpf);
+ *valid = FALSE;
+ return TRUE;
+ } else if ((((next_header >> 12) & 0xf) == 0) ||
+ (((next_header >> 12) & 0xf) == 0xf)) {
+ /* The essential parts were the same, but the bitrate held an
+ invalid value - also reject */
+ GST_DEBUG_OBJECT (mp3parse, "next header invalid (bitrate)");
+ *valid = FALSE;
+ return TRUE;
+ }
+
+ bpf = mp3_type_frame_length_from_header (mp3parse, next_header,
+ NULL, NULL, NULL, NULL, NULL, NULL, NULL);
+
+ offset += bpf;
+ frames_found++;
+ }
+
+ *valid = TRUE;
+ return TRUE;
+}
+
+static GstFlowReturn
+gst_mp3parse_handle_data (GstMPEGAudioParse * mp3parse, gboolean at_eos)
+{
+ GstFlowReturn flow = GST_FLOW_OK;
+ const guchar *data;
+ guint32 header;
+ int bpf;
+ guint available;
+ guint bitrate, layer, rate, channels, version, mode, crc;
+ gboolean caps_change;
+
+ /* while we still have at least 4 bytes (for the header) available */
+ while (gst_adapter_available (mp3parse->adapter) >= 4) {
+ /* Get the header bytes, check if they're potentially valid */
+ data = gst_adapter_peek (mp3parse->adapter, 4);
+ header = GST_READ_UINT32_BE (data);
+
+ if (!head_check (mp3parse, header)) {
+ /* Not a valid MP3 header; we start looking forward byte-by-byte trying to
+ find a place to resync */
+ if (!mp3parse->resyncing)
+ mp3parse->sync_offset = mp3parse->tracked_offset;
+ mp3parse->resyncing = TRUE;
+ gst_mp3parse_flush_bytes (mp3parse, 1);
+ GST_DEBUG_OBJECT (mp3parse, "wrong header, skipping byte");
+ continue;
+ }
+
+ /* We have a potentially valid header.
+ If this is just a normal 'next frame', we go ahead and output it.
+
+ However, sometimes, we do additional validation to ensure we haven't
+ got false sync (common with mp3 due to the short sync word).
+ The additional validation requires that we find several consecutive mp3
+ frames with the same major parameters, or reach EOS with a smaller
+ number of valid-looking frames.
+
+ We do this if:
+ - This is the very first frame we've processed
+ - We're resyncing after a non-accurate seek, or after losing sync
+ due to invalid data.
+ - The format of the stream changes in a major way (number of channels,
+ sample rate, layer, or mpeg version).
+ */
+ available = gst_adapter_available (mp3parse->adapter);
+
+ if (G_UNLIKELY (mp3parse->resyncing &&
+ mp3parse->tracked_offset - mp3parse->sync_offset > 2 * 1024 * 1024))
+ goto sync_failure;
+
+ bpf = mp3_type_frame_length_from_header (mp3parse, header,
+ &version, &layer, &channels, &bitrate, &rate, &mode, &crc);
+ g_assert (bpf != 0);
+
+ if (channels != mp3parse->channels ||
+ rate != mp3parse->rate || layer != mp3parse->layer ||
+ version != mp3parse->version)
+ caps_change = TRUE;
+ else
+ caps_change = FALSE;
+
+ if (mp3parse->resyncing || caps_change) {
+ gboolean valid;
+ if (!gst_mp3parse_validate_extended (mp3parse, header, bpf, at_eos,
+ &valid)) {
+ /* Not enough data to validate; wait for more */
+ break;
+ }
+
+ if (!valid) {
+ /* Extended validation failed; we probably got false sync.
+ Continue searching from the next byte in the stream */
+ if (!mp3parse->resyncing)
+ mp3parse->sync_offset = mp3parse->tracked_offset;
+ mp3parse->resyncing = TRUE;
+ gst_mp3parse_flush_bytes (mp3parse, 1);
+ 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 (caps_change) {
+ GstCaps *caps;
+
+ caps = mp3_caps_create (version, layer, channels, rate);
+ gst_pad_set_caps (mp3parse->srcpad, caps);
+ gst_caps_unref (caps);
+
+ mp3parse->channels = channels;
+ mp3parse->rate = rate;
+
+ 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 == 1) {
+ mp3parse->spf = 1152;
+ } else {
+ /* MPEG-2 or "2.5" */
+ mp3parse->spf = 576;
+ }
+
+ mp3parse->max_bitreservoir = gst_util_uint64_scale (GST_SECOND,
+ ((version == 1) ? 10 : 30) * mp3parse->spf, mp3parse->rate);
+ }
+
+ mp3parse->bit_rate = bitrate;
+
+ /* 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);
+
+ /* Check if we're seekable */
+ gst_mp3parse_check_seekability (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, mode, crc);
+ if (GST_FLOW_IS_FATAL (flow))
+ break;
+ } else {
+ GST_DEBUG_OBJECT (mp3parse, "skipping buffer of %d bytes", bpf);
+ gst_mp3parse_flush_bytes (mp3parse, bpf);
+ mp3parse->skip--;
+ }
+ }
+
+ return flow;
+
+ /* ERRORS */
+sync_failure:
+ {
+ GST_ELEMENT_ERROR (mp3parse, STREAM, DECODE,
+ ("Failed to parse stream"), (NULL));
+ return GST_FLOW_ERROR;
+ }
+}
+
+static GstFlowReturn
+gst_mp3parse_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstMPEGAudioParse *mp3parse;
+ 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);
+
+ mp3parse->discont |= GST_BUFFER_IS_DISCONT (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;
+
+ /* If we have no data pending and the next timestamp is
+ * invalid we can use the upstream timestamp for the next frame.
+ *
+ * This will give us a timestamp if we're resyncing and upstream
+ * gave us -1 as offset. */
+ if (avail == 0 && !GST_CLOCK_TIME_IS_VALID (mp3parse->next_ts))
+ mp3parse->next_ts = timestamp;
+
+ GST_LOG_OBJECT (mp3parse, "Have pending ts %" GST_TIME_FORMAT
+ " to apply in %" G_GINT64_FORMAT " bytes (@ off %" G_GINT64_FORMAT ")",
+ 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);
+
+ return gst_mp3parse_handle_data (mp3parse, FALSE);
+}
+
+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_WARNING_OBJECT (mp3parse, "invalid sync");
+ return FALSE;
+ }
+ /* if it's an invalid MPEG version */
+ if (((head >> 19) & 3) == 0x1) {
+ GST_WARNING_OBJECT (mp3parse, "invalid MPEG version: 0x%lx",
+ (head >> 19) & 3);
+ return FALSE;
+ }
+ /* if it's an invalid layer */
+ if (!((head >> 17) & 3)) {
+ GST_WARNING_OBJECT (mp3parse, "invalid layer: 0x%lx", (head >> 17) & 3);
+ return FALSE;
+ }
+ /* if it's an invalid bitrate */
+ if (((head >> 12) & 0xf) == 0x0) {
+ GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx."
+ "Free format files are not supported yet", (head >> 12) & 0xf);
+ return FALSE;
+ }
+ if (((head >> 12) & 0xf) == 0xf) {
+ GST_WARNING_OBJECT (mp3parse, "invalid bitrate: 0x%lx", (head >> 12) & 0xf);
+ return FALSE;
+ }
+ /* if it's an invalid samplerate */
+ if (((head >> 10) & 0x3) == 0x3) {
+ GST_WARNING_OBJECT (mp3parse, "invalid samplerate: 0x%lx",
+ (head >> 10) & 0x3);
+ return FALSE;
+ }
+
+ if ((head & 0x3) == 0x2) {
+ /* Ignore this as there are some files with emphasis 0x2 that can
+ * be played fine. See BGO #537235 */
+ GST_WARNING_OBJECT (mp3parse, "invalid emphasis: 0x%lx", head & 0x3);
+ }
+
+ return TRUE;
+}
+
+static void
+gst_mp3parse_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstMPEGAudioParse *src;
+
+ 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;
+
+ 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 && mp3parse->vbri_valid) {
+ *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 && mp3parse->vbri_valid) {
+ *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, TRUE))
+ 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) &&
+ (total_bytes = mp3parse->xing_bytes) &&
+ (total_time = mp3parse->xing_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 && (total_bytes = mp3parse->vbri_bytes) &&
+ (total_time = mp3parse->vbri_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, gboolean from_total_time)
+{
+ 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 (!from_total_time && (mp3parse->xing_flags & XING_TOC_FLAG) &&
+ (total_bytes = mp3parse->xing_bytes) &&
+ (total_time = mp3parse->xing_total_time)) {
+ gdouble fa, fb, fx;
+ gdouble pos;
+ gint index;
+
+ pos = CLAMP ((bytepos * 256.0) / total_bytes, 0.0, 256.0);
+ 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 (!from_total_time && mp3parse->vbri_seek_table &&
+ (total_bytes = mp3parse->vbri_bytes) &&
+ (total_time = mp3parse->vbri_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;
+ MPEGAudioPendingAccurateSeek *seek;
+ GstClockTime start;
+
+ 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;
+ }
+
+ seek = g_new0 (MPEGAudioPendingAccurateSeek, 1);
+
+ seek->segment = mp3parse->segment;
+
+ gst_segment_set_seek (&seek->segment, rate, GST_FORMAT_TIME,
+ flags, cur_type, cur, stop_type, stop, NULL);
+
+ /* 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) {
+ 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;
+ gint64 seek_ts = (cur > mp3parse->max_bitreservoir) ?
+ (cur - mp3parse->max_bitreservoir) : 0;
+
+ for (start_node = mp3parse->seek_table; start_node;
+ start_node = start_node->next) {
+ entry = start_node->data;
+
+ if (seek_ts >= 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;
+ }
+
+ }
+ event = gst_event_new_seek (rate, GST_FORMAT_BYTES, flags, cur_type,
+ byte_cur, stop_type, byte_stop);
+ g_mutex_lock (mp3parse->pending_seeks_lock);
+ 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_seeks_lock);
+ if (gst_pad_push_event (mp3parse->sinkpad, event)) {
+ mp3parse->exact_position = TRUE;
+ return TRUE;
+ } else {
+ mp3parse->exact_position = TRUE;
+ g_mutex_lock (mp3parse->pending_seeks_lock);
+ mp3parse->pending_accurate_seeks =
+ g_slist_remove (mp3parse->pending_accurate_seeks, seek);
+ g_mutex_unlock (mp3parse->pending_seeks_lock);
+ g_free (seek);
+ return FALSE;
+ }
+ }
+
+ 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);
+
+ GST_LOG_OBJECT (mp3parse, "Storing pending seek");
+ g_mutex_lock (mp3parse->pending_seeks_lock);
+ seek->upstream_start = byte_cur;
+ seek->timestamp_start = cur;
+ mp3parse->pending_nonaccurate_seeks =
+ g_slist_prepend (mp3parse->pending_nonaccurate_seeks, seek);
+ g_mutex_unlock (mp3parse->pending_seeks_lock);
+ if (gst_pad_push_event (mp3parse->sinkpad, event)) {
+ return TRUE;
+ } else {
+ g_mutex_lock (mp3parse->pending_seeks_lock);
+ mp3parse->pending_nonaccurate_seeks =
+ g_slist_remove (mp3parse->pending_nonaccurate_seeks, seek);
+ g_mutex_unlock (mp3parse->pending_seeks_lock);
+ g_free (seek);
+ return FALSE;
+ }
+
+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;
+ gboolean res = FALSE;
+
+ mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
+
+ 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;
+ gboolean res = FALSE;
+ GstPad *peer;
+
+ mp3parse = GST_MP3PARSE (gst_pad_get_parent (pad));
+
+ GST_LOG_OBJECT (pad, "%s query", GST_QUERY_TYPE_NAME (query));
+
+ 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;
+ case GST_QUERY_SEEKING:
+ gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
+
+ /* does upstream handle ? */
+ if ((peer = gst_pad_get_peer (mp3parse->sinkpad)) != NULL) {
+ res = gst_pad_query (peer, query);
+ gst_object_unref (peer);
+ }
+ /* we may be able to help if in TIME */
+ if (format == GST_FORMAT_TIME) {
+ gboolean seekable;
+
+ gst_query_parse_seeking (query, &format, &seekable, NULL, NULL);
+ /* already OK if upstream takes care */
+ if (!(res && seekable)) {
+ gint64 pos;
+
+ seekable = TRUE;
+ if (!mp3parse_total_time (mp3parse, &total) || total == -1) {
+ seekable = FALSE;
+ } else if (!mp3parse_time_to_bytepos (mp3parse, 0, &pos)) {
+ seekable = FALSE;
+ } else {
+ GstQuery *q;
+
+ q = gst_query_new_seeking (GST_FORMAT_BYTES);
+ if (!gst_pad_peer_query (mp3parse->sinkpad, q)) {
+ seekable = FALSE;
+ } else {
+ gst_query_parse_seeking (q, &format, &seekable, NULL, NULL);
+ }
+ gst_query_unref (q);
+ }
+ gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0, 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 G_GNUC_UNUSED)
+{
+ static const GstQueryType query_types[] = {
+ GST_QUERY_POSITION,
+ GST_QUERY_DURATION,
+ 0
+ };
+
+ return query_types;
+}