--- a/gst_plugins_good/gst/mpegaudioparse/gstmpegaudioparse.c Fri Jul 09 16:26:45 2010 -0500
+++ /dev/null Thu Jan 01 00:00:00 1970 +0000
@@ -1,2199 +0,0 @@
-/* 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;
-}